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| author | root <root@artemis.panaceas.org> | 2015-12-25 04:40:36 +0000 |
|---|---|---|
| committer | root <root@artemis.panaceas.org> | 2015-12-25 04:40:36 +0000 |
| commit | 849369d6c66d3054688672f97d31fceb8e8230fb (patch) | |
| tree | 6135abc790ca67dedbe07c39806591e70eda81ce /sound/oss | |
| download | linux-3.0.35-kobo-849369d6c66d3054688672f97d31fceb8e8230fb.tar.gz linux-3.0.35-kobo-849369d6c66d3054688672f97d31fceb8e8230fb.tar.bz2 linux-3.0.35-kobo-849369d6c66d3054688672f97d31fceb8e8230fb.zip | |
initial_commit
Diffstat (limited to 'sound/oss')
77 files changed, 44615 insertions, 0 deletions
diff --git a/sound/oss/.gitignore b/sound/oss/.gitignore new file mode 100644 index 00000000..7efb12b4 --- /dev/null +++ b/sound/oss/.gitignore @@ -0,0 +1,4 @@ +#Ignore generated files +maui_boot.h +pss_boot.h +trix_boot.h diff --git a/sound/oss/CHANGELOG b/sound/oss/CHANGELOG new file mode 100644 index 00000000..8706cd66 --- /dev/null +++ b/sound/oss/CHANGELOG @@ -0,0 +1,369 @@ +Note these changes relate to Hannu's code and don't include the changes +made outside of this for modularising the sound + +Changelog for version 3.8o +-------------------------- + +Since 3.8h +- Included support for OPL3-SA1 and SoftOSS + +Since 3.8 +- Fixed SNDCTL_DSP_GETOSPACE +- Compatibility fixes for Linux 2.1.47 + +Since 3.8-beta21 +- Fixed all known bugs (I think). + +Since 3.8-beta8 +- Lot of fixes to audio playback code in dmabuf.c + +Since 3.8-beta6 +- Fixed the famous Quake delay bug. + +Since 3.8-beta5 +- Fixed many bugs in audio playback. + +Since 3.8-beta4 +- Just minor changes. + +Since 3.8-beta1 +- Major rewrite of audio playback handling. +- Added AWE32 support by Takashi Iwai (in ./lowlevel/). + +Since 3.7-beta# +- Passing of ioctl() parameters between soundcard.c and other modules has been +changed so that arg always points to kernel space. +- Some bugfixes. + +Since 3.7-beta5 +- Disabled MIDI input with GUS PnP (Interwave). There seems to be constant +stream of received 0x00 bytes when the MIDI receiver is enabled. + +Since 3.5 +- Changes almost everywhere. +- Support for OPTi 82C924-based sound cards. + +Since 3.5.4-beta8 +- Fixed a bug in handling of non-fragment sized writes in 16 bit/stereo mode + with GUS. +- Limited minimum fragment size with some audio devices (GUS=512 and + SB=32). These devices require more time to "recover" from processing + of each fragment. + +Since 3.5.4-beta6/7 +- There seems to be problems in the OPTi 82C930 so cards based on this + chip don't necessarily work yet. There are problems in detecting the + MIDI interface. Also mixer volumes may be seriously wrong on some systems. + You can safely use this driver version with C930 if it looks to work. + However please don't complain if you have problems with it. C930 support + should be fixed in future releases. +- Got initialization of GUS PnP to work. With this version GUS PnP should + work in GUS compatible mode after initialization using isapnptools. +- Fixed a bug in handling of full duplex cards in write only mode. This has + been causing "audio device opening" errors with RealAudio player. + +Since 3.5.4.beta5 +- Changes to OPTi 82C930 driver. +- Major changes to the Soundscape driver. The driver requires now just one + DMA channel. The extra audio/dsp device (the "Not functional" one) used + for code download in the earlier versions has been eliminated. There is now + just one /dev/dsp# device which is used both for code download and audio. + +Since 3.5.4.beta4 +- Minor changes. + +Since 3.5.4-beta2 +- Fixed silent playback with ESS 688/1688. +- Got SB16 to work without the 16 bit DMA channel (only the 8 bit one + is required for 8 and 16 bit modes). +- Added the "lowlevel" subdirectory for additional low level drivers that + are not part of USS core. See lowlevel/README for more info. +- Included support for ACI mixer (by Markus Kuhn). ACI is a mixer used in + miroPCM sound cards. See lowlevel/aci.readme for more info. +- Support for Aztech Washington chipset (AZT2316 ASIC). + +Since 3.5.4-beta1 +- Reduced clicking with AD1848. +- Support for OPTi 82C930. Only half duplex at this time. 16 bit playback + is sometimes just white noise (occurs randomly). + +Since 3.5.2 +- Major changes to the SB/Jazz16/ESS driver (most parts rewritten). + The most noticeable new feature is support for multiple SB cards at the same + time. +- Renamed sb16_midi.c to uart401.c. Also modified it to work also with + other MPU401 UART compatible cards than SB16/ESS/Jazz. +- Some changes which reduce clicking in audio playback. +- Copying policy is now GPL. + +Since 3.5.1 +- TB Maui initialization support +Since 3.5 +- Improved handling of playback underrun situations. + +Since 3.5-beta10 +- Bug fixing + +Since 3.5-beta9 +- Fixed for compatibility with Linux 1.3.70 and later. +- Changed boot time passing of 16 bit DMA channel number to SB driver. + +Since 3.5-beta8 +- Minor changes + +Since 3.5-beta7 +- enhancements to configure program (by Jeff Tranter): + - prompts are in same format as 1.3.x Linux kernel config program + - on-line help for each question + - fixed some compile warnings detected by gcc/g++ -Wall + - minor grammatical changes to prompts + +Since 3.5-beta6 +- Fixed bugs in mmap() support. +- Minor changes to Maui driver. + +Since 3.5-beta5 +- Fixed crash after recording with ESS688. It's generally a good + idea to stop inbound DMA transfers before freeing the memory + buffer. +- Fixed handling of AD1845 codec (for example Shuttle Sound System). +- Few other fixes. + +Since 3.5-beta4 +- Fixed bug in handling of uninitialized instruments with GUS. + +Since 3.5-beta3 +- Few changes which decrease popping at end/beginning of audio playback. + +Since 3.5-beta2 +- Removed MAD16+CS4231 hack made in previous version since it didn't + help. +- Fixed the above bug in proper way and in proper place. Many thanks + to James Hightower. + +Since 3.5-beta1 +- Bug fixes. +- Full duplex audio with MAD16+CS4231 may work now. The driver configures + SB DMA of MAD16 so that it doesn't conflict with codec's DMA channels. + The side effect is that all 8 bit DMA channels (0,1,3) are populated in + duplex mode. + +Since 3.5-alpha9 +- Bug fixes (mostly in Jazz16 and ESS1688/688 supports). +- Temporarily disabled recording with ESS1688/688 since it causes crash. +- Changed audio buffer partitioning algorithm so that it selects + smaller fragment size than earlier. This improves real time capabilities + of the driver and makes recording to disk to work better. Unfortunately + this change breaks some programs which assume that fragments cannot be + shorter than 4096 bytes. + +Since 3.5-alpha8 +- Bug fixes + +Since 3.5-alpha7 +- Linux kernel compatible configuration (_EXPERIMENTAL_). Enable + using command "cd /linux/drivers/sound;make script" and then + just run kernel's make config normally. +- Minor fixes to the SB support. Hopefully the driver works with + all SB models now. +- Added support for ESS ES1688 "AudioDrive" based cards. + +Since 3.5-alpha6 +- SB Pro and SB16 supports are no longer separately selectable options. + Enabling SB enables them too. +- Changed all #ifndef EXCLUDE_xx stuff to #ifdef CONFIG_xx. Modified +configure to handle this. +- Removed initialization messages from the +modularized version. They can be enabled by using init_trace=1 in +the insmod command line (insmod sound init_trace=1). +- More AIX stuff. +- Added support for synchronizing dsp/audio devices with /dev/sequencer. +- mmap() support for dsp/audio devices. + +Since 3.5-alpha5 +- AIX port. +- Changed some xxx_PATCH macros in soundcard.h to work with + big endian machines. + +Since 3.5-alpha4 +- Removed the 'setfx' stuff from the version distributed with kernel + sources. Running 'setfx' is required again. + +Since 3.5-alpha3 +- Moved stuff from the 'setfx' program to the AudioTrix Pro driver. + +Since 3.5-alpha2 +- Modifications to makefile and configure.c. Unnecessary sources + are no longer compiled. Newly created local.h is also copied to + /etc/soundconf. "make oldconfig" reads /etc/soundconf and produces + new local.h which is compatible with current version of the driver. +- Some fixes to the SB16 support. +- Fixed random protection fault in gus_wave.c + +Since 3.5-alpha1 +- Modified to work with Linux-1.3.33 and later +- Some minor changes + +Since 3.0.2 +- Support for CS4232 based PnP cards (AcerMagic S23 etc). +- Full duplex support for some CS4231, CS4232 and AD1845 based cards +(GUS MAX, AudioTrix Pro, AcerMagic S23 and many MAD16/Mozart cards +having a codec mentioned above). +- Almost fully rewritten loadable modules support. +- Fixed some bugs. +- Huge amount of testing (more testing is still required). +- mmap() support (works with some cards). Requires much more testing. +- Sample/patch/program loading for TB Maui/Tropez. No initialization +since TB doesn't allow me to release that code. +- Using CS4231 compatible codecs as timer for /dev/music. + +Since 3.0.1 +- Added allocation of I/O ports, DMA channels and interrupts +to the initialization code. This may break modules support since +the driver may not free some resources on unload. Should be fixed soon. + +Since 3.0 +- Some important bug fixes. +- select() for /dev/dsp and /dev/audio (Linux only). +(To use select() with read, you have to call read() to start +the recording. Calling write() kills recording immediately so +use select() carefully when you are writing a half duplex app. +Full duplex mode is not implemented yet.) Select works also with +/dev/sequencer and /dev/music. Maybe with /dev/midi## too. + +Since 3.0-beta2 +- Minor fixes. +- Added Readme.cards + +Since 3.0-beta1 +- Minor fixes to the modules support. +- Eliminated call to sb_free_irq() in ad1848.c +- Rewritten MAD16&Mozart support (not tested with MAD16 Pro). +- Fix to DMA initialization of PSS cards. +- Some fixes to ad1848/cs42xx mixer support (GUS MAX, MSS, etc.) +- Fixed some bugs in the PSS driver which caused I/O errors with + the MSS mode (/dev/dsp). + +Since 3.0-950506 +- Recording with GUS MAX fixed. It works when the driver is configured + to use two DMA channels with GUS MAX (16 bit ones recommended). + +Since 3.0-94xxxx +- Too many changes + +Since 3.0-940818 +- Fixes for Linux 1.1.4x. +- Disables Disney Sound System with SG NX Pro 16 (less noise). + +Since 2.90-2 +- Fixes to soundcard.h +- Non blocking mode to /dev/sequencer +- Experimental detection code for Ensoniq Soundscape. + +Since 2.90 +- Minor and major bug fixes + +Since pre-3.0-940712 +- GUS MAX support +- Partially working MSS/WSS support (could work with some cards). +- Hardware u-Law and A-Law support with AD1848/CS4248 and CS4231 codecs + (GUS MAX, GUS16, WSS etc). Hardware ADPCM is possible with GUS16 and + GUS MAX, but it doesn't work yet. +Since pre-3.0-940426 +- AD1848/CS4248/CS4231 codec support (MSS, GUS MAX, Aztec, Orchid etc). +This codec chip is used in various sound cards. This version is developed +for the 16 bit daughtercard of GUS. It should work with other cards also +if the following requirements are met: + - The I/O, IRQ and DMA settings are jumper selectable or + the card is initialized by booting DOS before booting Linux (etc.). + - You add the IO, IRQ and DMA settings manually to the local.h. + (Just define GUS16_BASE, GUS16_IRQ and GUS16_DMA). Note that + the base address bust be the base address of the codec chip not the + card itself. For the GUS16 these are the same but most MSS compatible + cards have the codec located at card_base+4. +- Some minor changes + +Since 2.5 (******* MAJOR REWRITE ***********) + +This version is based on v2.3. I have tried to maintain two versions +together so that this one should have the same features than v2.5. +Something may still be missing. If you notice such things, please let me +know. + +The Readme.v30 contains more details. + +- /dev/midi## devices. +- /dev/sequencer2 + +Since 2.5-beta2 +- Some fine tuning to the GUS v3.7 mixer code. +- Fixed speed limits for the plain SB (1.0 to 2.0). + +Since 2.5-beta +- Fixed OPL-3 detection with SB. Caused problems with PAS16. +- GUS v3.7 mixer support. + +Since 2.4 +- Mixer support for Sound Galaxy NX Pro (define __SGNXPRO__ on your local.h). +- Fixed truncated sound on /dev/dsp when the device is closed. +- Linear volume mode for GUS +- Pitch bends larger than +/- 2 octaves. +- MIDI recording for SB and SB Pro. (Untested). +- Some other fixes. +- SB16 MIDI and DSP drivers only initialized if SB16 actually installed. +- Implemented better detection for OPL-3. This should be useful if you + have an old SB Pro (the non-OPL-3 one) or a SB 2.0 clone which has a OPL-3. +- SVR4.2 support by Ian Hartas. Initial ALPHA TEST version (untested). + +Since 2.3b +- Fixed bug which made it impossible to make long recordings to disk. + Recording was not restarted after a buffer overflow situation. +- Limited mixer support for GUS. +- Numerous improvements to the GUS driver by Andrew Robinson. Including + some click removal etc. + +Since 2.3 +- Fixed some minor bugs in the SB16 driver. + +Since 2.2b +- Full SB16 DSP support. 8/16 bit, mono/stereo +- The SCO and FreeBSD versions should be in sync now. There are some + problems with SB16 and GUS in the FreeBSD versions. + The DMA buffer allocation of the SCO version has been polished but + there could still be some problems. At least it hogs memory. + The DMA channel + configuration method used in the SCO/System is a hack. +- Support for the MPU emulation of the SB16. +- Some big arrays are now allocated boot time. This makes the BSS segment + smaller which makes it possible to use the full driver with + NetBSD. These arrays are not allocated if no suitable sound card is available. +- Fixed a bug in the compute_and_set_volume in gus_wave.c +- Fixed the too fast mono playback problem of SB Pro and PAS16. + +Since 2.2 +- Stereo recording for SB Pro. Somehow it was missing and nobody + had noticed it earlier. +- Minor polishing. +- Interpreting of boot time arguments (sound=) for Linux. +- Breakup of sb_dsp.c. Parts of the code has been moved to + sb_mixer.c and sb_midi.c + +Since 2.1 +- Preliminary support for SB16. + - The SB16 mixer is supported in its native mode. + - Digitized voice capability up to 44.1 kHz/8 bit/mono + (16 bit and stereo support coming in the next release). +- Fixed some bugs in the digitized voice driver for PAS16. +- Proper initialization of the SB emulation of latest PAS16 models. + +- Significantly improved /dev/dsp and /dev/audio support. + - Now supports half duplex mode. It's now possible to record and + playback without closing and reopening the device. + - It's possible to use smaller buffers than earlier. There is a new + ioctl(fd, SNDCTL_DSP_SUBDIVIDE, &n) where n should be 1, 2 or 4. + This call instructs the driver to use smaller buffers. The default + buffer size (0.5 to 1.0 seconds) is divided by n. Should be called + immediately after opening the device. + +Since 2.0 +Just cosmetic changes. diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig new file mode 100644 index 00000000..6c93e051 --- /dev/null +++ b/sound/oss/Kconfig @@ -0,0 +1,543 @@ +# 18 Apr 1998, Michael Elizabeth Chastain, <mailto:mec@shout.net> +# More hacking for modularisation. +# +# Prompt user for primary drivers. + +config SOUND_BCM_CS4297A + tristate "Crystal Sound CS4297a (for Swarm)" + depends on SIBYTE_SWARM + help + The BCM91250A has a Crystal CS4297a on synchronous serial + port B (in addition to the DB-9 serial port). Say Y or M + here to enable the sound chip instead of the UART. Also + note that CONFIG_KGDB should not be enabled at the same + time, since it also attempts to use this UART port. + +config SOUND_VWSND + tristate "SGI Visual Workstation Sound" + depends on X86_VISWS + help + Say Y or M if you have an SGI Visual Workstation and you want to be + able to use its on-board audio. Read + <file:Documentation/sound/oss/vwsnd> for more info on this driver's + capabilities. + +config SOUND_MSNDCLAS + tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey" + depends on (m || !STANDALONE) && ISA + help + Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or + Monterey (not for the Pinnacle or Fiji). + + See <file:Documentation/sound/oss/MultiSound> for important information + about this driver. Note that it has been discontinued, but the + Voyetra Turtle Beach knowledge base entry for it is still available + at <http://www.turtlebeach.com/site/kb_ftp/790.asp>. + +comment "Compiled-in MSND Classic support requires firmware during compilation." + depends on SOUND_PRIME && SOUND_MSNDCLAS=y + +config MSNDCLAS_HAVE_BOOT + bool + depends on SOUND_MSNDCLAS=y && !STANDALONE + default y + +config MSNDCLAS_INIT_FILE + string "Full pathname of MSNDINIT.BIN firmware file" + depends on SOUND_MSNDCLAS + default "/etc/sound/msndinit.bin" + help + The MultiSound cards have two firmware files which are required for + operation, and are not currently included. These files can be + obtained from Turtle Beach. See + <file:Documentation/sound/oss/MultiSound> for information on how to + obtain this. + +config MSNDCLAS_PERM_FILE + string "Full pathname of MSNDPERM.BIN firmware file" + depends on SOUND_MSNDCLAS + default "/etc/sound/msndperm.bin" + help + The MultiSound cards have two firmware files which are required for + operation, and are not currently included. These files can be + obtained from Turtle Beach. See + <file:Documentation/sound/oss/MultiSound> for information on how to + obtain this. + +config MSNDCLAS_IRQ + int "MSND Classic IRQ 5, 7, 9, 10, 11, 12" + depends on SOUND_MSNDCLAS=y + default "5" + help + Interrupt Request line for the MultiSound Classic and related cards. + +config MSNDCLAS_MEM + hex "MSND Classic memory B0000, C8000, D0000, D8000, E0000, E8000" + depends on SOUND_MSNDCLAS=y + default "D0000" + help + Memory-mapped I/O base address for the MultiSound Classic and + related cards. + +config MSNDCLAS_IO + hex "MSND Classic I/O 210, 220, 230, 240, 250, 260, 290, 3E0" + depends on SOUND_MSNDCLAS=y + default "290" + help + I/O port address for the MultiSound Classic and related cards. + +config SOUND_MSNDPIN + tristate "Support for Turtle Beach MultiSound Pinnacle, Fiji" + depends on (m || !STANDALONE) && ISA + help + Say M here if you have a Turtle Beach MultiSound Pinnacle or Fiji. + See <file:Documentation/sound/oss/MultiSound> for important information + about this driver. Note that it has been discontinued, but the + Voyetra Turtle Beach knowledge base entry for it is still available + at <http://www.turtlebeach.com/site/kb_ftp/600.asp>. + +comment "Compiled-in MSND Pinnacle support requires firmware during compilation." + depends on SOUND_PRIME && SOUND_MSNDPIN=y + +config MSNDPIN_HAVE_BOOT + bool + depends on SOUND_MSNDPIN=y + default y + +config MSNDPIN_INIT_FILE + string "Full pathname of PNDSPINI.BIN firmware file" + depends on SOUND_MSNDPIN + default "/etc/sound/pndspini.bin" + help + The MultiSound cards have two firmware files which are required + for operation, and are not currently included. These files can be + obtained from Turtle Beach. See + <file:Documentation/sound/oss/MultiSound> for information on how to + obtain this. + +config MSNDPIN_PERM_FILE + string "Full pathname of PNDSPERM.BIN firmware file" + depends on SOUND_MSNDPIN + default "/etc/sound/pndsperm.bin" + help + The MultiSound cards have two firmware files which are required for + operation, and are not currently included. These files can be + obtained from Turtle Beach. See + <file:Documentation/sound/oss/MultiSound> for information on how to + obtain this. + +config MSNDPIN_IRQ + int "MSND Pinnacle IRQ 5, 7, 9, 10, 11, 12" + depends on SOUND_MSNDPIN=y + default "5" + help + Interrupt request line for the primary synthesizer on MultiSound + Pinnacle and Fiji sound cards. + +config MSNDPIN_MEM + hex "MSND Pinnacle memory B0000, C8000, D0000, D8000, E0000, E8000" + depends on SOUND_MSNDPIN=y + default "D0000" + help + Memory-mapped I/O base address for the primary synthesizer on + MultiSound Pinnacle and Fiji sound cards. + +config MSNDPIN_IO + hex "MSND Pinnacle I/O 210, 220, 230, 240, 250, 260, 290, 3E0" + depends on SOUND_MSNDPIN=y + default "290" + help + Memory-mapped I/O base address for the primary synthesizer on + MultiSound Pinnacle and Fiji sound cards. + +config MSNDPIN_DIGITAL + bool "MSND Pinnacle has S/PDIF I/O" + depends on SOUND_MSNDPIN=y + help + If you have the S/PDIF daughter board for the Pinnacle or Fiji, + answer Y here; otherwise, say N. If you have this, you will be able + to play and record from the S/PDIF port (digital signal). See + <file:Documentation/sound/oss/MultiSound> for information on how to make + use of this capability. + +config MSNDPIN_NONPNP + bool "MSND Pinnacle non-PnP Mode" + depends on SOUND_MSNDPIN=y + help + The Pinnacle and Fiji card resources can be configured either with + PnP, or through a configuration port. Say Y here if your card is NOT + in PnP mode. For the Pinnacle, configuration in non-PnP mode allows + use of the IDE and joystick peripherals on the card as well; these + do not show up when the card is in PnP mode. Specifying zero for any + resource of a device will disable the device. If you are running the + card in PnP mode, you must say N here and use isapnptools to + configure the card's resources. + +comment "MSND Pinnacle DSP section will be configured to above parameters." + depends on SOUND_MSNDPIN=y && MSNDPIN_NONPNP + +config MSNDPIN_CFG + hex "MSND Pinnacle config port 250,260,270" + depends on MSNDPIN_NONPNP + default "250" + help + This is the port which the Pinnacle and Fiji uses to configure the + card's resources when not in PnP mode. If your card is in PnP mode, + then be sure to say N to the previous option, "MSND Pinnacle Non-PnP + Mode". + +comment "Pinnacle-specific Device Configuration (0 disables)" + depends on SOUND_MSNDPIN=y && MSNDPIN_NONPNP + +config MSNDPIN_MPU_IO + hex "MSND Pinnacle MPU I/O (e.g. 330)" + depends on MSNDPIN_NONPNP + default "0" + help + Memory-mapped I/O base address for the Kurzweil daughterboard + synthesizer on MultiSound Pinnacle and Fiji sound cards. + +config MSNDPIN_MPU_IRQ + int "MSND Pinnacle MPU IRQ (e.g. 9)" + depends on MSNDPIN_NONPNP + default "0" + help + Interrupt request number for the Kurzweil daughterboard + synthesizer on MultiSound Pinnacle and Fiji sound cards. + +config MSNDPIN_IDE_IO0 + hex "MSND Pinnacle IDE I/O 0 (e.g. 170)" + depends on MSNDPIN_NONPNP + default "0" + help + CD-ROM drive 0 memory-mapped I/O base address for the MultiSound + Pinnacle and Fiji sound cards. + +config MSNDPIN_IDE_IO1 + hex "MSND Pinnacle IDE I/O 1 (e.g. 376)" + depends on MSNDPIN_NONPNP + default "0" + help + CD-ROM drive 1 memory-mapped I/O base address for the MultiSound + Pinnacle and Fiji sound cards. + +config MSNDPIN_IDE_IRQ + int "MSND Pinnacle IDE IRQ (e.g. 15)" + depends on MSNDPIN_NONPNP + default "0" + help + Interrupt request number for the IDE CD-ROM interface on the + MultiSound Pinnacle and Fiji sound cards. + +config MSNDPIN_JOYSTICK_IO + hex "MSND Pinnacle joystick I/O (e.g. 200)" + depends on MSNDPIN_NONPNP + default "0" + help + Memory-mapped I/O base address for the joystick port on MultiSound + Pinnacle and Fiji sound cards. + +config MSND_FIFOSIZE + int "MSND buffer size (kB)" + depends on SOUND_MSNDPIN=y || SOUND_MSNDCLAS=y + default "128" + help + Configures the size of each audio buffer, in kilobytes, for + recording and playing in the MultiSound drivers (both the Classic + and Pinnacle). Larger values reduce the chance of data overruns at + the expense of overall latency. If unsure, use the default. + +menuconfig SOUND_OSS + tristate "OSS sound modules" + depends on ISA_DMA_API && VIRT_TO_BUS + help + OSS is the Open Sound System suite of sound card drivers. They make + sound programming easier since they provide a common API. Say Y or + M here (the module will be called sound) if you haven't found a + driver for your sound card above, then pick your driver from the + list below. + +if SOUND_OSS + +config SOUND_TRACEINIT + bool "Verbose initialisation" + help + Verbose soundcard initialization -- affects the format of autoprobe + and initialization messages at boot time. + +config SOUND_DMAP + bool "Persistent DMA buffers" + ---help--- + Linux can often have problems allocating DMA buffers for ISA sound + cards on machines with more than 16MB of RAM. This is because ISA + DMA buffers must exist below the 16MB boundary and it is quite + possible that a large enough free block in this region cannot be + found after the machine has been running for a while. If you say Y + here the DMA buffers (64Kb) will be allocated at boot time and kept + until the shutdown. This option is only useful if you said Y to + "OSS sound modules", above. If you said M to "OSS sound modules" + then you can get the persistent DMA buffer functionality by passing + the command-line argument "dmabuf=1" to the sound module. + + Say Y unless you have 16MB or more RAM or a PCI sound card. + +config SOUND_VMIDI + tristate "Loopback MIDI device support" + help + Support for MIDI loopback on port 1 or 2. + +config SOUND_TRIX + tristate "MediaTrix AudioTrix Pro support" + help + Answer Y if you have the AudioTriX Pro sound card manufactured + by MediaTrix. + +config TRIX_HAVE_BOOT + bool "Have TRXPRO.HEX firmware file" + depends on SOUND_TRIX=y && !STANDALONE + help + The MediaTrix AudioTrix Pro has an on-board microcontroller which + needs to be initialized by downloading the code from the file + TRXPRO.HEX in the DOS driver directory. If you don't have the + TRXPRO.HEX file handy you may skip this step. However, the SB and + MPU-401 modes of AudioTrix Pro will not work without this file! + +config TRIX_BOOT_FILE + string "Full pathname of TRXPRO.HEX firmware file" + depends on TRIX_HAVE_BOOT + default "/etc/sound/trxpro.hex" + help + Enter the full pathname of your TRXPRO.HEX file, starting from /. + +config SOUND_MSS + tristate "Microsoft Sound System support" + ---help--- + Again think carefully before answering Y to this question. It's + safe to answer Y if you have the original Windows Sound System card + made by Microsoft or Aztech SG 16 Pro (or NX16 Pro). Also you may + say Y in case your card is NOT among these: + + ATI Stereo F/X, AdLib, Audio Excell DSP16, Cardinal DSP16, + Ensoniq SoundScape (and compatibles made by Reveal and Spea), + Gravis Ultrasound, Gravis Ultrasound ACE, Gravis Ultrasound Max, + Gravis Ultrasound with 16 bit option, Logitech Sound Man 16, + Logitech SoundMan Games, Logitech SoundMan Wave, MAD16 Pro (OPTi + 82C929), Media Vision Jazz16, MediaTriX AudioTriX Pro, Microsoft + Windows Sound System (MSS/WSS), Mozart (OAK OTI-601), Orchid + SW32, Personal Sound System (PSS), Pro Audio Spectrum 16, Pro + Audio Studio 16, Pro Sonic 16, Roland MPU-401 MIDI interface, + Sound Blaster 1.0, Sound Blaster 16, Sound Blaster 16ASP, Sound + Blaster 2.0, Sound Blaster AWE32, Sound Blaster Pro, TI TM4000M + notebook, ThunderBoard, Turtle Beach Tropez, Yamaha FM + synthesizers (OPL2, OPL3 and OPL4), 6850 UART MIDI Interface. + + For cards having native support in VoxWare, consult the card + specific instructions in <file:Documentation/sound/oss/README.OSS>. + Some drivers have their own MSS support and saying Y to this option + will cause a conflict. + + If you compile the driver into the kernel, you have to add + "ad1848=<io>,<irq>,<dma>,<dma2>[,<type>]" to the kernel command + line. + +config SOUND_MPU401 + tristate "MPU-401 support (NOT for SB16)" + ---help--- + Be careful with this question. The MPU401 interface is supported by + all sound cards. However, some natively supported cards have their + own driver for MPU401. Enabling this MPU401 option with these cards + will cause a conflict. Also, enabling MPU401 on a system that + doesn't really have a MPU401 could cause some trouble. If your card + was in the list of supported cards, look at the card specific + instructions in the <file:Documentation/sound/oss/README.OSS> file. It + is safe to answer Y if you have a true MPU401 MIDI interface card. + + If you compile the driver into the kernel, you have to add + "mpu401=<io>,<irq>" to the kernel command line. + +config SOUND_PAS + tristate "ProAudioSpectrum 16 support" + ---help--- + Answer Y only if you have a Pro Audio Spectrum 16, ProAudio Studio + 16 or Logitech SoundMan 16 sound card. Answer N if you have some + other card made by Media Vision or Logitech since those are not + PAS16 compatible. Please read <file:Documentation/sound/oss/PAS16>. + It is not necessary to add Sound Blaster support separately; it + is included in PAS support. + + If you compile the driver into the kernel, you have to add + "pas2=<io>,<irq>,<dma>,<dma2>,<sbio>,<sbirq>,<sbdma>,<sbdma2> + to the kernel command line. + +config PAS_JOYSTICK + bool "Enable PAS16 joystick port" + depends on SOUND_PAS=y + help + Say Y here to enable the Pro Audio Spectrum 16's auxiliary joystick + port. + +config SOUND_PSS + tristate "PSS (AD1848, ADSP-2115, ESC614) support" + help + Answer Y or M if you have an Orchid SW32, Cardinal DSP16, Beethoven + ADSP-16 or some other card based on the PSS chipset (AD1848 codec + + ADSP-2115 DSP chip + Echo ESC614 ASIC CHIP). For more information on + how to compile it into the kernel or as a module see the file + <file:Documentation/sound/oss/PSS>. + + If you compile the driver into the kernel, you have to add + "pss=<io>,<mssio>,<mssirq>,<mssdma>,<mpuio>,<mpuirq>" to the kernel + command line. + +config PSS_MIXER + bool "Enable PSS mixer (Beethoven ADSP-16 and other compatible)" + depends on SOUND_PSS + help + Answer Y for Beethoven ADSP-16. You may try to say Y also for other + cards if they have master volume, bass, treble, and you can't + control it under Linux. If you answer N for Beethoven ADSP-16, you + can't control master volume, bass, treble and synth volume. + + If you said M to "PSS support" above, you may enable or disable this + PSS mixer with the module parameter pss_mixer. For more information + see the file <file:Documentation/sound/oss/PSS>. + +config PSS_HAVE_BOOT + bool "Have DSPxxx.LD firmware file" + depends on SOUND_PSS && !STANDALONE + help + If you have the DSPxxx.LD file or SYNTH.LD file for you card, say Y + to include this file. Without this file the synth device (OPL) may + not work. + +config PSS_BOOT_FILE + string "Full pathname of DSPxxx.LD firmware file" + depends on PSS_HAVE_BOOT + default "/etc/sound/dsp001.ld" + help + Enter the full pathname of your DSPxxx.LD file or SYNTH.LD file, + starting from /. + +config SOUND_SB + tristate "100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support" + ---help--- + Answer Y if you have an original Sound Blaster card made by Creative + Labs or a 100% hardware compatible clone (like the Thunderboard or + SM Games). For an unknown card you may answer Y if the card claims + to be Sound Blaster-compatible. + + Please read the file <file:Documentation/sound/oss/Soundblaster>. + + You should also say Y here for cards based on the Avance Logic + ALS-007 and ALS-1X0 chips (read <file:Documentation/sound/oss/ALS>) and + for cards based on ESS chips (read + <file:Documentation/sound/oss/ESS1868> and + <file:Documentation/sound/oss/ESS>). If you have an SB AWE 32 or SB AWE + 64, say Y here and also to "AWE32 synth" below and read + <file:Documentation/sound/oss/INSTALL.awe>. If you have an IBM Mwave + card, say Y here and read <file:Documentation/sound/oss/mwave>. + + If you compile the driver into the kernel and don't want to use + isapnp, you have to add "sb=<io>,<irq>,<dma>,<dma2>" to the kernel + command line. + + You can say M here to compile this driver as a module; the module is + called sb. + +config SOUND_YM3812 + tristate "Yamaha FM synthesizer (YM3812/OPL-3) support" + ---help--- + Answer Y if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4). + Answering Y is usually a safe and recommended choice, however some + cards may have software (TSR) FM emulation. Enabling FM support with + these cards may cause trouble (I don't currently know of any such + cards, however). Please read the file + <file:Documentation/sound/oss/OPL3> if your card has an OPL3 chip. + + If you compile the driver into the kernel, you have to add + "opl3=<io>" to the kernel command line. + + If unsure, say Y. + +config SOUND_UART6850 + tristate "6850 UART support" + help + This option enables support for MIDI interfaces based on the 6850 + UART chip. This interface is rarely found on sound cards. It's safe + to answer N to this question. + + If you compile the driver into the kernel, you have to add + "uart6850=<io>,<irq>" to the kernel command line. + +config SOUND_AEDSP16 + tristate "Gallant Audio Cards (SC-6000 and SC-6600 based)" + ---help--- + Answer Y if you have a Gallant's Audio Excel DSP 16 card. This + driver supports Audio Excel DSP 16 but not the III nor PnP versions + of this card. + + The Gallant's Audio Excel DSP 16 card can emulate either an SBPro or + a Microsoft Sound System card, so you should have said Y to either + "100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support" + or "Microsoft Sound System support", above, and you need to answer + the "MSS emulation" and "SBPro emulation" questions below + accordingly. You should say Y to one and only one of these two + questions. + + Read the <file:Documentation/sound/oss/README.OSS> file and the head of + <file:sound/oss/aedsp16.c> as well as + <file:Documentation/sound/oss/AudioExcelDSP16> to get more information + about this driver and its configuration. + +config SC6600 + bool "SC-6600 based audio cards (new Audio Excel DSP 16)" + depends on SOUND_AEDSP16 + help + The SC6600 is the new version of DSP mounted on the Audio Excel DSP + 16 cards. Find in the manual the FCC ID of your audio card and + answer Y if you have an SC6600 DSP. + +config SC6600_JOY + bool "Activate SC-6600 Joystick Interface" + depends on SC6600 + help + Say Y here in order to use the joystick interface of the Audio Excel + DSP 16 card. + +config SC6600_CDROM + int "SC-6600 CDROM Interface (4=None, 3=IDE, 1=Panasonic, 0=?Sony?)" + depends on SC6600 + default "4" + help + This is used to activate the CD-ROM interface of the Audio Excel + DSP 16 card. Enter: 0 for Sony, 1 for Panasonic, 2 for IDE, 4 for no + CD-ROM present. + +config SC6600_CDROMBASE + hex "SC-6600 CDROM Interface I/O Address" + depends on SC6600 + default "0" + help + Base I/O port address for the CD-ROM interface of the Audio Excel + DSP 16 card. + +config SOUND_VIDC + tristate "VIDC 16-bit sound" + depends on ARM && (ARCH_ACORN || ARCH_CLPS7500) + help + 16-bit support for the VIDC onboard sound hardware found on Acorn + machines. + +config SOUND_WAVEARTIST + tristate "Netwinder WaveArtist" + depends on ARM && ARCH_NETWINDER + help + Say Y here to include support for the Rockwell WaveArtist sound + system. This driver is mainly for the NetWinder. + +config SOUND_KAHLUA + tristate "XpressAudio Sound Blaster emulation" + depends on SOUND_SB + +endif # SOUND_OSS + diff --git a/sound/oss/Makefile b/sound/oss/Makefile new file mode 100644 index 00000000..77f21b68 --- /dev/null +++ b/sound/oss/Makefile @@ -0,0 +1,108 @@ +# Makefile for the Linux sound card driver +# +# 18 Apr 1998, Michael Elizabeth Chastain, <mailto:mec@shout.net> +# Rewritten to use lists instead of if-statements. + +# Each configuration option enables a list of files. + +obj-$(CONFIG_SOUND_OSS) += sound.o + +# Please leave it as is, cause the link order is significant ! + +obj-$(CONFIG_SOUND_AEDSP16) += aedsp16.o +obj-$(CONFIG_SOUND_PSS) += pss.o ad1848.o mpu401.o +obj-$(CONFIG_SOUND_TRIX) += trix.o ad1848.o sb_lib.o uart401.o +obj-$(CONFIG_SOUND_MSS) += ad1848.o +obj-$(CONFIG_SOUND_PAS) += pas2.o sb.o sb_lib.o uart401.o +obj-$(CONFIG_SOUND_SB) += sb.o sb_lib.o uart401.o +obj-$(CONFIG_SOUND_KAHLUA) += kahlua.o +obj-$(CONFIG_SOUND_MPU401) += mpu401.o +obj-$(CONFIG_SOUND_UART6850) += uart6850.o +obj-$(CONFIG_SOUND_YM3812) += opl3.o +obj-$(CONFIG_SOUND_VMIDI) += v_midi.o +obj-$(CONFIG_SOUND_VIDC) += vidc_mod.o +obj-$(CONFIG_SOUND_WAVEARTIST) += waveartist.o +obj-$(CONFIG_SOUND_MSNDCLAS) += msnd.o msnd_classic.o +obj-$(CONFIG_SOUND_MSNDPIN) += msnd.o msnd_pinnacle.o +obj-$(CONFIG_SOUND_VWSND) += vwsnd.o +obj-$(CONFIG_SOUND_BCM_CS4297A) += swarm_cs4297a.o + +obj-$(CONFIG_DMASOUND) += dmasound/ + +# Declare multi-part drivers. + +sound-objs := \ + dev_table.o soundcard.o \ + audio.o dmabuf.o \ + midi_synth.o midibuf.o \ + sequencer.o sound_timer.o sys_timer.o + +pas2-objs := pas2_card.o pas2_midi.o pas2_mixer.o pas2_pcm.o +sb-objs := sb_card.o +sb_lib-objs := sb_common.o sb_audio.o sb_midi.o sb_mixer.o sb_ess.o +vidc_mod-objs := vidc.o vidc_fill.o + +hostprogs-y := bin2hex hex2hex + +# Files generated that shall be removed upon make clean +clean-files := msndperm.c msndinit.c pndsperm.c pndspini.c \ + pss_boot.h trix_boot.h + +# Firmware files that need translation +# +# The translated files are protected by a file that keeps track +# of what name was used to build them. If the name changes, they +# will be forced to be remade. +# + +# Turtle Beach MultiSound + +ifeq ($(CONFIG_MSNDCLAS_HAVE_BOOT),y) + $(obj)/msnd_classic.o: $(obj)/msndperm.c $(obj)/msndinit.c + + $(obj)/msndperm.c: $(patsubst "%", %, $(CONFIG_MSNDCLAS_PERM_FILE)) $(obj)/bin2hex + $(obj)/bin2hex msndperm < $< > $@ + + $(obj)/msndinit.c: $(patsubst "%", %, $(CONFIG_MSNDCLAS_INIT_FILE)) $(obj)/bin2hex + $(obj)/bin2hex msndinit < $< > $@ +endif + +ifeq ($(CONFIG_MSNDPIN_HAVE_BOOT),y) + $(obj)/msnd_pinnacle.o: $(obj)/pndsperm.c $(obj)/pndspini.c + + $(obj)/pndsperm.c: $(patsubst "%", %, $(CONFIG_MSNDPIN_PERM_FILE)) $(obj)/bin2hex + $(obj)/bin2hex pndsperm < $< > $@ + + $(obj)/pndspini.c: $(patsubst "%", %, $(CONFIG_MSNDPIN_INIT_FILE)) $(obj)/bin2hex + $(obj)/bin2hex pndspini < $< > $@ +endif + +# PSS (ECHO-ADI2111) + +$(obj)/pss.o: $(obj)/pss_boot.h + +ifeq ($(CONFIG_PSS_HAVE_BOOT),y) + $(obj)/pss_boot.h: $(patsubst "%", %, $(CONFIG_PSS_BOOT_FILE)) $(obj)/bin2hex + $(obj)/bin2hex pss_synth < $< > $@ +else + $(obj)/pss_boot.h: + $(Q)( \ + echo 'static unsigned char * pss_synth = NULL;'; \ + echo 'static int pss_synthLen = 0;'; \ + ) > $@ +endif + +# MediaTrix AudioTrix Pro + +$(obj)/trix.o: $(obj)/trix_boot.h + +ifeq ($(CONFIG_TRIX_HAVE_BOOT),y) + $(obj)/trix_boot.h: $(patsubst "%", %, $(CONFIG_TRIX_BOOT_FILE)) $(obj)/hex2hex + $(obj)/hex2hex -i trix_boot < $< > $@ +else + $(obj)/trix_boot.h: + $(Q)( \ + echo 'static unsigned char * trix_boot = NULL;'; \ + echo 'static int trix_boot_len = 0;'; \ + ) > $@ +endif diff --git a/sound/oss/README.FIRST b/sound/oss/README.FIRST new file mode 100644 index 00000000..90fdcf06 --- /dev/null +++ b/sound/oss/README.FIRST @@ -0,0 +1,6 @@ +The modular sound driver patches were funded by Red Hat Software +(www.redhat.com). The sound driver here is thus a modified version of +Hannu's code. Please bear that in mind when considering the appropriate +forums for bug reporting. + +Alan Cox diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c new file mode 100644 index 00000000..4d2a6ae9 --- /dev/null +++ b/sound/oss/ad1848.c @@ -0,0 +1,3069 @@ +/* + * sound/oss/ad1848.c + * + * The low level driver for the AD1848/CS4248 codec chip which + * is used for example in the MS Sound System. + * + * The CS4231 which is used in the GUS MAX and some other cards is + * upwards compatible with AD1848 and this driver is able to drive it. + * + * CS4231A and AD1845 are upward compatible with CS4231. However + * the new features of these chips are different. + * + * CS4232 is a PnP audio chip which contains a CS4231A (and SB, MPU). + * CS4232A is an improved version of CS4232. + * + * + * + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * + * + * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed) + * general sleep/wakeup clean up. + * Alan Cox : reformatted. Fixed SMP bugs. Moved to kernel alloc/free + * of irqs. Use dev_id. + * Christoph Hellwig : adapted to module_init/module_exit + * Aki Laukkanen : added power management support + * Arnaldo C. de Melo : added missing restore_flags in ad1848_resume + * Miguel Freitas : added ISA PnP support + * Alan Cox : Added CS4236->4239 identification + * Daniel T. Cobra : Alernate config/mixer for later chips + * Alan Cox : Merged chip idents and config code + * + * TODO + * APM save restore assist code on IBM thinkpad + * + * Status: + * Tested. Believed fully functional. + */ + +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/stddef.h> +#include <linux/slab.h> +#include <linux/isapnp.h> +#include <linux/pnp.h> +#include <linux/spinlock.h> + +#define DEB(x) +#define DEB1(x) +#include "sound_config.h" + +#include "ad1848.h" +#include "ad1848_mixer.h" + +typedef struct +{ + spinlock_t lock; + int base; + int irq; + int dma1, dma2; + int dual_dma; /* 1, when two DMA channels allocated */ + int subtype; + unsigned char MCE_bit; + unsigned char saved_regs[64]; /* Includes extended register space */ + int debug_flag; + + int audio_flags; + int record_dev, playback_dev; + + int xfer_count; + int audio_mode; + int open_mode; + int intr_active; + char *chip_name, *name; + int model; +#define MD_1848 1 +#define MD_4231 2 +#define MD_4231A 3 +#define MD_1845 4 +#define MD_4232 5 +#define MD_C930 6 +#define MD_IWAVE 7 +#define MD_4235 8 /* Crystal Audio CS4235 */ +#define MD_1845_SSCAPE 9 /* Ensoniq Soundscape PNP*/ +#define MD_4236 10 /* 4236 and higher */ +#define MD_42xB 11 /* CS 42xB */ +#define MD_4239 12 /* CS4239 */ + + /* Mixer parameters */ + int recmask; + int supported_devices, orig_devices; + int supported_rec_devices, orig_rec_devices; + int *levels; + short mixer_reroute[32]; + int dev_no; + volatile unsigned long timer_ticks; + int timer_running; + int irq_ok; + mixer_ents *mix_devices; + int mixer_output_port; +} ad1848_info; + +typedef struct ad1848_port_info +{ + int open_mode; + int speed; + unsigned char speed_bits; + int channels; + int audio_format; + unsigned char format_bits; +} +ad1848_port_info; + +static struct address_info cfg; +static int nr_ad1848_devs; + +static int deskpro_xl; +static int deskpro_m; +static int soundpro; + +static volatile signed char irq2dev[17] = { + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, -1 +}; + +#ifndef EXCLUDE_TIMERS +static int timer_installed = -1; +#endif + +static int loaded; + +static int ad_format_mask[13 /*devc->model */ ] = +{ + 0, + AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW, + AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM, + AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM, + AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW, /* AD1845 */ + AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM, + AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM, + AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM, + AFMT_U8 | AFMT_S16_LE /* CS4235 */, + AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW /* Ensoniq Soundscape*/, + AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM, + AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM, + AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM +}; + +static ad1848_info adev_info[MAX_AUDIO_DEV]; + +#define io_Index_Addr(d) ((d)->base) +#define io_Indexed_Data(d) ((d)->base+1) +#define io_Status(d) ((d)->base+2) +#define io_Polled_IO(d) ((d)->base+3) + +static struct { + unsigned char flags; +#define CAP_F_TIMER 0x01 +} capabilities [10 /*devc->model */ ] = { + {0} + ,{0} /* MD_1848 */ + ,{CAP_F_TIMER} /* MD_4231 */ + ,{CAP_F_TIMER} /* MD_4231A */ + ,{CAP_F_TIMER} /* MD_1845 */ + ,{CAP_F_TIMER} /* MD_4232 */ + ,{0} /* MD_C930 */ + ,{CAP_F_TIMER} /* MD_IWAVE */ + ,{0} /* MD_4235 */ + ,{CAP_F_TIMER} /* MD_1845_SSCAPE */ +}; + +#ifdef CONFIG_PNP +static int isapnp = 1; +static int isapnpjump; +static int reverse; + +static int audio_activated; +#else +static int isapnp; +#endif + + + +static int ad1848_open(int dev, int mode); +static void ad1848_close(int dev); +static void ad1848_output_block(int dev, unsigned long buf, int count, int intrflag); +static void ad1848_start_input(int dev, unsigned long buf, int count, int intrflag); +static int ad1848_prepare_for_output(int dev, int bsize, int bcount); +static int ad1848_prepare_for_input(int dev, int bsize, int bcount); +static void ad1848_halt(int dev); +static void ad1848_halt_input(int dev); +static void ad1848_halt_output(int dev); +static void ad1848_trigger(int dev, int bits); +static irqreturn_t adintr(int irq, void *dev_id); + +#ifndef EXCLUDE_TIMERS +static int ad1848_tmr_install(int dev); +static void ad1848_tmr_reprogram(int dev); +#endif + +static int ad_read(ad1848_info * devc, int reg) +{ + int x; + int timeout = 900000; + + while (timeout > 0 && inb(devc->base) == 0x80) /*Are we initializing */ + timeout--; + + if(reg < 32) + { + outb(((unsigned char) (reg & 0xff) | devc->MCE_bit), io_Index_Addr(devc)); + x = inb(io_Indexed_Data(devc)); + } + else + { + int xreg, xra; + + xreg = (reg & 0xff) - 32; + xra = (((xreg & 0x0f) << 4) & 0xf0) | 0x08 | ((xreg & 0x10) >> 2); + outb(((unsigned char) (23 & 0xff) | devc->MCE_bit), io_Index_Addr(devc)); + outb(((unsigned char) (xra & 0xff)), io_Indexed_Data(devc)); + x = inb(io_Indexed_Data(devc)); + } + + return x; +} + +static void ad_write(ad1848_info * devc, int reg, int data) +{ + int timeout = 900000; + + while (timeout > 0 && inb(devc->base) == 0x80) /* Are we initializing */ + timeout--; + + if(reg < 32) + { + outb(((unsigned char) (reg & 0xff) | devc->MCE_bit), io_Index_Addr(devc)); + outb(((unsigned char) (data & 0xff)), io_Indexed_Data(devc)); + } + else + { + int xreg, xra; + + xreg = (reg & 0xff) - 32; + xra = (((xreg & 0x0f) << 4) & 0xf0) | 0x08 | ((xreg & 0x10) >> 2); + outb(((unsigned char) (23 & 0xff) | devc->MCE_bit), io_Index_Addr(devc)); + outb(((unsigned char) (xra & 0xff)), io_Indexed_Data(devc)); + outb((unsigned char) (data & 0xff), io_Indexed_Data(devc)); + } +} + +static void wait_for_calibration(ad1848_info * devc) +{ + int timeout = 0; + + /* + * Wait until the auto calibration process has finished. + * + * 1) Wait until the chip becomes ready (reads don't return 0x80). + * 2) Wait until the ACI bit of I11 gets on and then off. + */ + + timeout = 100000; + while (timeout > 0 && inb(devc->base) == 0x80) + timeout--; + if (inb(devc->base) & 0x80) + printk(KERN_WARNING "ad1848: Auto calibration timed out(1).\n"); + + timeout = 100; + while (timeout > 0 && !(ad_read(devc, 11) & 0x20)) + timeout--; + if (!(ad_read(devc, 11) & 0x20)) + return; + + timeout = 80000; + while (timeout > 0 && (ad_read(devc, 11) & 0x20)) + timeout--; + if (ad_read(devc, 11) & 0x20) + if ((devc->model != MD_1845) && (devc->model != MD_1845_SSCAPE)) + printk(KERN_WARNING "ad1848: Auto calibration timed out(3).\n"); +} + +static void ad_mute(ad1848_info * devc) +{ + int i; + unsigned char prev; + + /* + * Save old register settings and mute output channels + */ + + for (i = 6; i < 8; i++) + { + prev = devc->saved_regs[i] = ad_read(devc, i); + } + +} + +static void ad_unmute(ad1848_info * devc) +{ +} + +static void ad_enter_MCE(ad1848_info * devc) +{ + int timeout = 1000; + unsigned short prev; + + while (timeout > 0 && inb(devc->base) == 0x80) /*Are we initializing */ + timeout--; + + devc->MCE_bit = 0x40; + prev = inb(io_Index_Addr(devc)); + if (prev & 0x40) + { + return; + } + outb((devc->MCE_bit), io_Index_Addr(devc)); +} + +static void ad_leave_MCE(ad1848_info * devc) +{ + unsigned char prev, acal; + int timeout = 1000; + + while (timeout > 0 && inb(devc->base) == 0x80) /*Are we initializing */ + timeout--; + + acal = ad_read(devc, 9); + + devc->MCE_bit = 0x00; + prev = inb(io_Index_Addr(devc)); + outb((0x00), io_Index_Addr(devc)); /* Clear the MCE bit */ + + if ((prev & 0x40) == 0) /* Not in MCE mode */ + { + return; + } + outb((0x00), io_Index_Addr(devc)); /* Clear the MCE bit */ + if (acal & 0x08) /* Auto calibration is enabled */ + wait_for_calibration(devc); +} + +static int ad1848_set_recmask(ad1848_info * devc, int mask) +{ + unsigned char recdev; + int i, n; + unsigned long flags; + + mask &= devc->supported_rec_devices; + + /* Rename the mixer bits if necessary */ + for (i = 0; i < 32; i++) + { + if (devc->mixer_reroute[i] != i) + { + if (mask & (1 << i)) + { + mask &= ~(1 << i); + mask |= (1 << devc->mixer_reroute[i]); + } + } + } + + n = 0; + for (i = 0; i < 32; i++) /* Count selected device bits */ + if (mask & (1 << i)) + n++; + + spin_lock_irqsave(&devc->lock,flags); + if (!soundpro) { + if (n == 0) + mask = SOUND_MASK_MIC; + else if (n != 1) { /* Too many devices selected */ + mask &= ~devc->recmask; /* Filter out active settings */ + + n = 0; + for (i = 0; i < 32; i++) /* Count selected device bits */ + if (mask & (1 << i)) + n++; + + if (n != 1) + mask = SOUND_MASK_MIC; + } + switch (mask) { + case SOUND_MASK_MIC: + recdev = 2; + break; + + case SOUND_MASK_LINE: + case SOUND_MASK_LINE3: + recdev = 0; + break; + + case SOUND_MASK_CD: + case SOUND_MASK_LINE1: + recdev = 1; + break; + + case SOUND_MASK_IMIX: + recdev = 3; + break; + + default: + mask = SOUND_MASK_MIC; + recdev = 2; + } + + recdev <<= 6; + ad_write(devc, 0, (ad_read(devc, 0) & 0x3f) | recdev); + ad_write(devc, 1, (ad_read(devc, 1) & 0x3f) | recdev); + } else { /* soundpro */ + unsigned char val; + int set_rec_bit; + int j; + + for (i = 0; i < 32; i++) { /* For each bit */ + if ((devc->supported_rec_devices & (1 << i)) == 0) + continue; /* Device not supported */ + + for (j = LEFT_CHN; j <= RIGHT_CHN; j++) { + if (devc->mix_devices[i][j].nbits == 0) /* Inexistent channel */ + continue; + + /* + * This is tricky: + * set_rec_bit becomes 1 if the corresponding bit in mask is set + * then it gets flipped if the polarity is inverse + */ + set_rec_bit = ((mask & (1 << i)) != 0) ^ devc->mix_devices[i][j].recpol; + + val = ad_read(devc, devc->mix_devices[i][j].recreg); + val &= ~(1 << devc->mix_devices[i][j].recpos); + val |= (set_rec_bit << devc->mix_devices[i][j].recpos); + ad_write(devc, devc->mix_devices[i][j].recreg, val); + } + } + } + spin_unlock_irqrestore(&devc->lock,flags); + + /* Rename the mixer bits back if necessary */ + for (i = 0; i < 32; i++) + { + if (devc->mixer_reroute[i] != i) + { + if (mask & (1 << devc->mixer_reroute[i])) + { + mask &= ~(1 << devc->mixer_reroute[i]); + mask |= (1 << i); + } + } + } + devc->recmask = mask; + return mask; +} + +static void change_bits(ad1848_info * devc, unsigned char *regval, + unsigned char *muteval, int dev, int chn, int newval) +{ + unsigned char mask; + int shift; + int mute; + int mutemask; + int set_mute_bit; + + set_mute_bit = (newval == 0) ^ devc->mix_devices[dev][chn].mutepol; + + if (devc->mix_devices[dev][chn].polarity == 1) /* Reverse */ + newval = 100 - newval; + + mask = (1 << devc->mix_devices[dev][chn].nbits) - 1; + shift = devc->mix_devices[dev][chn].bitpos; + + if (devc->mix_devices[dev][chn].mutepos == 8) + { /* if there is no mute bit */ + mute = 0; /* No mute bit; do nothing special */ + mutemask = ~0; /* No mute bit; do nothing special */ + } + else + { + mute = (set_mute_bit << devc->mix_devices[dev][chn].mutepos); + mutemask = ~(1 << devc->mix_devices[dev][chn].mutepos); + } + + newval = (int) ((newval * mask) + 50) / 100; /* Scale it */ + *regval &= ~(mask << shift); /* Clear bits */ + *regval |= (newval & mask) << shift; /* Set new value */ + + *muteval &= mutemask; + *muteval |= mute; +} + +static int ad1848_mixer_get(ad1848_info * devc, int dev) +{ + if (!((1 << dev) & devc->supported_devices)) + return -EINVAL; + + dev = devc->mixer_reroute[dev]; + + return devc->levels[dev]; +} + +static void ad1848_mixer_set_channel(ad1848_info *devc, int dev, int value, int channel) +{ + int regoffs, muteregoffs; + unsigned char val, muteval; + unsigned long flags; + + regoffs = devc->mix_devices[dev][channel].regno; + muteregoffs = devc->mix_devices[dev][channel].mutereg; + val = ad_read(devc, regoffs); + + if (muteregoffs != regoffs) { + muteval = ad_read(devc, muteregoffs); + change_bits(devc, &val, &muteval, dev, channel, value); + } + else + change_bits(devc, &val, &val, dev, channel, value); + + spin_lock_irqsave(&devc->lock,flags); + ad_write(devc, regoffs, val); + devc->saved_regs[regoffs] = val; + if (muteregoffs != regoffs) { + ad_write(devc, muteregoffs, muteval); + devc->saved_regs[muteregoffs] = muteval; + } + spin_unlock_irqrestore(&devc->lock,flags); +} + +static int ad1848_mixer_set(ad1848_info * devc, int dev, int value) +{ + int left = value & 0x000000ff; + int right = (value & 0x0000ff00) >> 8; + int retvol; + + if (dev > 31) + return -EINVAL; + + if (!(devc->supported_devices & (1 << dev))) + return -EINVAL; + + dev = devc->mixer_reroute[dev]; + + if (devc->mix_devices[dev][LEFT_CHN].nbits == 0) + return -EINVAL; + + if (left > 100) + left = 100; + if (right > 100) + right = 100; + + if (devc->mix_devices[dev][RIGHT_CHN].nbits == 0) /* Mono control */ + right = left; + + retvol = left | (right << 8); + + /* Scale volumes */ + left = mix_cvt[left]; + right = mix_cvt[right]; + + devc->levels[dev] = retvol; + + /* + * Set the left channel + */ + ad1848_mixer_set_channel(devc, dev, left, LEFT_CHN); + + /* + * Set the right channel + */ + if (devc->mix_devices[dev][RIGHT_CHN].nbits == 0) + goto out; + ad1848_mixer_set_channel(devc, dev, right, RIGHT_CHN); + + out: + return retvol; +} + +static void ad1848_mixer_reset(ad1848_info * devc) +{ + int i; + char name[32]; + unsigned long flags; + + devc->mix_devices = &(ad1848_mix_devices[0]); + + sprintf(name, "%s_%d", devc->chip_name, nr_ad1848_devs); + + for (i = 0; i < 32; i++) + devc->mixer_reroute[i] = i; + + devc->supported_rec_devices = MODE1_REC_DEVICES; + + switch (devc->model) + { + case MD_4231: + case MD_4231A: + case MD_1845: + case MD_1845_SSCAPE: + devc->supported_devices = MODE2_MIXER_DEVICES; + break; + + case MD_C930: + devc->supported_devices = C930_MIXER_DEVICES; + devc->mix_devices = &(c930_mix_devices[0]); + break; + + case MD_IWAVE: + devc->supported_devices = MODE3_MIXER_DEVICES; + devc->mix_devices = &(iwave_mix_devices[0]); + break; + + case MD_42xB: + case MD_4239: + devc->mix_devices = &(cs42xb_mix_devices[0]); + devc->supported_devices = MODE3_MIXER_DEVICES; + break; + case MD_4232: + case MD_4235: + case MD_4236: + devc->supported_devices = MODE3_MIXER_DEVICES; + break; + + case MD_1848: + if (soundpro) { + devc->supported_devices = SPRO_MIXER_DEVICES; + devc->supported_rec_devices = SPRO_REC_DEVICES; + devc->mix_devices = &(spro_mix_devices[0]); + break; + } + + default: + devc->supported_devices = MODE1_MIXER_DEVICES; + } + + devc->orig_devices = devc->supported_devices; + devc->orig_rec_devices = devc->supported_rec_devices; + + devc->levels = load_mixer_volumes(name, default_mixer_levels, 1); + + for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) + { + if (devc->supported_devices & (1 << i)) + ad1848_mixer_set(devc, i, devc->levels[i]); + } + + ad1848_set_recmask(devc, SOUND_MASK_MIC); + + devc->mixer_output_port = devc->levels[31] | AUDIO_HEADPHONE | AUDIO_LINE_OUT; + + spin_lock_irqsave(&devc->lock,flags); + if (!soundpro) { + if (devc->mixer_output_port & AUDIO_SPEAKER) + ad_write(devc, 26, ad_read(devc, 26) & ~0x40); /* Unmute mono out */ + else + ad_write(devc, 26, ad_read(devc, 26) | 0x40); /* Mute mono out */ + } else { + /* + * From the "wouldn't it be nice if the mixer API had (better) + * support for custom stuff" category + */ + /* Enable surround mode and SB16 mixer */ + ad_write(devc, 16, 0x60); + } + spin_unlock_irqrestore(&devc->lock,flags); +} + +static int ad1848_mixer_ioctl(int dev, unsigned int cmd, void __user *arg) +{ + ad1848_info *devc = mixer_devs[dev]->devc; + int val; + + if (cmd == SOUND_MIXER_PRIVATE1) + { + if (get_user(val, (int __user *)arg)) + return -EFAULT; + + if (val != 0xffff) + { + unsigned long flags; + val &= (AUDIO_SPEAKER | AUDIO_HEADPHONE | AUDIO_LINE_OUT); + devc->mixer_output_port = val; + val |= AUDIO_HEADPHONE | AUDIO_LINE_OUT; /* Always on */ + devc->mixer_output_port = val; + spin_lock_irqsave(&devc->lock,flags); + if (val & AUDIO_SPEAKER) + ad_write(devc, 26, ad_read(devc, 26) & ~0x40); /* Unmute mono out */ + else + ad_write(devc, 26, ad_read(devc, 26) | 0x40); /* Mute mono out */ + spin_unlock_irqrestore(&devc->lock,flags); + } + val = devc->mixer_output_port; + return put_user(val, (int __user *)arg); + } + if (cmd == SOUND_MIXER_PRIVATE2) + { + if (get_user(val, (int __user *)arg)) + return -EFAULT; + return(ad1848_control(AD1848_MIXER_REROUTE, val)); + } + if (((cmd >> 8) & 0xff) == 'M') + { + if (_SIOC_DIR(cmd) & _SIOC_WRITE) + { + switch (cmd & 0xff) + { + case SOUND_MIXER_RECSRC: + if (get_user(val, (int __user *)arg)) + return -EFAULT; + val = ad1848_set_recmask(devc, val); + break; + + default: + if (get_user(val, (int __user *)arg)) + return -EFAULT; + val = ad1848_mixer_set(devc, cmd & 0xff, val); + break; + } + return put_user(val, (int __user *)arg); + } + else + { + switch (cmd & 0xff) + { + /* + * Return parameters + */ + + case SOUND_MIXER_RECSRC: + val = devc->recmask; + break; + + case SOUND_MIXER_DEVMASK: + val = devc->supported_devices; + break; + + case SOUND_MIXER_STEREODEVS: + val = devc->supported_devices; + if (devc->model != MD_C930) + val &= ~(SOUND_MASK_SPEAKER | SOUND_MASK_IMIX); + break; + + case SOUND_MIXER_RECMASK: + val = devc->supported_rec_devices; + break; + + case SOUND_MIXER_CAPS: + val=SOUND_CAP_EXCL_INPUT; + break; + + default: + val = ad1848_mixer_get(devc, cmd & 0xff); + break; + } + return put_user(val, (int __user *)arg); + } + } + else + return -EINVAL; +} + +static int ad1848_set_speed(int dev, int arg) +{ + ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; + ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc; + + /* + * The sampling speed is encoded in the least significant nibble of I8. The + * LSB selects the clock source (0=24.576 MHz, 1=16.9344 MHz) and other + * three bits select the divisor (indirectly): + * + * The available speeds are in the following table. Keep the speeds in + * the increasing order. + */ + typedef struct + { + int speed; + unsigned char bits; + } + speed_struct; + + static speed_struct speed_table[] = + { + {5510, (0 << 1) | 1}, + {5510, (0 << 1) | 1}, + {6620, (7 << 1) | 1}, + {8000, (0 << 1) | 0}, + {9600, (7 << 1) | 0}, + {11025, (1 << 1) | 1}, + {16000, (1 << 1) | 0}, + {18900, (2 << 1) | 1}, + {22050, (3 << 1) | 1}, + {27420, (2 << 1) | 0}, + {32000, (3 << 1) | 0}, + {33075, (6 << 1) | 1}, + {37800, (4 << 1) | 1}, + {44100, (5 << 1) | 1}, + {48000, (6 << 1) | 0} + }; + + int i, n, selected = -1; + + n = sizeof(speed_table) / sizeof(speed_struct); + + if (arg <= 0) + return portc->speed; + + if (devc->model == MD_1845 || devc->model == MD_1845_SSCAPE) /* AD1845 has different timer than others */ + { + if (arg < 4000) + arg = 4000; + if (arg > 50000) + arg = 50000; + + portc->speed = arg; + portc->speed_bits = speed_table[3].bits; + return portc->speed; + } + if (arg < speed_table[0].speed) + selected = 0; + if (arg > speed_table[n - 1].speed) + selected = n - 1; + + for (i = 1 /*really */ ; selected == -1 && i < n; i++) + { + if (speed_table[i].speed == arg) + selected = i; + else if (speed_table[i].speed > arg) + { + int diff1, diff2; + + diff1 = arg - speed_table[i - 1].speed; + diff2 = speed_table[i].speed - arg; + + if (diff1 < diff2) + selected = i - 1; + else + selected = i; + } + } + if (selected == -1) + { + printk(KERN_WARNING "ad1848: Can't find speed???\n"); + selected = 3; + } + portc->speed = speed_table[selected].speed; + portc->speed_bits = speed_table[selected].bits; + return portc->speed; +} + +static short ad1848_set_channels(int dev, short arg) +{ + ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc; + + if (arg != 1 && arg != 2) + return portc->channels; + + portc->channels = arg; + return arg; +} + +static unsigned int ad1848_set_bits(int dev, unsigned int arg) +{ + ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; + ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc; + + static struct format_tbl + { + int format; + unsigned char bits; + } + format2bits[] = + { + { + 0, 0 + } + , + { + AFMT_MU_LAW, 1 + } + , + { + AFMT_A_LAW, 3 + } + , + { + AFMT_IMA_ADPCM, 5 + } + , + { + AFMT_U8, 0 + } + , + { + AFMT_S16_LE, 2 + } + , + { + AFMT_S16_BE, 6 + } + , + { + AFMT_S8, 0 + } + , + { + AFMT_U16_LE, 0 + } + , + { + AFMT_U16_BE, 0 + } + }; + int i, n = sizeof(format2bits) / sizeof(struct format_tbl); + + if (arg == 0) + return portc->audio_format; + + if (!(arg & ad_format_mask[devc->model])) + arg = AFMT_U8; + + portc->audio_format = arg; + + for (i = 0; i < n; i++) + if (format2bits[i].format == arg) + { + if ((portc->format_bits = format2bits[i].bits) == 0) + return portc->audio_format = AFMT_U8; /* Was not supported */ + + return arg; + } + /* Still hanging here. Something must be terribly wrong */ + portc->format_bits = 0; + return portc->audio_format = AFMT_U8; +} + +static struct audio_driver ad1848_audio_driver = +{ + .owner = THIS_MODULE, + .open = ad1848_open, + .close = ad1848_close, + .output_block = ad1848_output_block, + .start_input = ad1848_start_input, + .prepare_for_input = ad1848_prepare_for_input, + .prepare_for_output = ad1848_prepare_for_output, + .halt_io = ad1848_halt, + .halt_input = ad1848_halt_input, + .halt_output = ad1848_halt_output, + .trigger = ad1848_trigger, + .set_speed = ad1848_set_speed, + .set_bits = ad1848_set_bits, + .set_channels = ad1848_set_channels +}; + +static struct mixer_operations ad1848_mixer_operations = +{ + .owner = THIS_MODULE, + .id = "SOUNDPORT", + .name = "AD1848/CS4248/CS4231", + .ioctl = ad1848_mixer_ioctl +}; + +static int ad1848_open(int dev, int mode) +{ + ad1848_info *devc; + ad1848_port_info *portc; + unsigned long flags; + + if (dev < 0 || dev >= num_audiodevs) + return -ENXIO; + + devc = (ad1848_info *) audio_devs[dev]->devc; + portc = (ad1848_port_info *) audio_devs[dev]->portc; + + /* here we don't have to protect against intr */ + spin_lock(&devc->lock); + if (portc->open_mode || (devc->open_mode & mode)) + { + spin_unlock(&devc->lock); + return -EBUSY; + } + devc->dual_dma = 0; + + if (audio_devs[dev]->flags & DMA_DUPLEX) + { + devc->dual_dma = 1; + } + devc->intr_active = 0; + devc->audio_mode = 0; + devc->open_mode |= mode; + portc->open_mode = mode; + spin_unlock(&devc->lock); + ad1848_trigger(dev, 0); + + if (mode & OPEN_READ) + devc->record_dev = dev; + if (mode & OPEN_WRITE) + devc->playback_dev = dev; +/* + * Mute output until the playback really starts. This decreases clicking (hope so). + */ + spin_lock_irqsave(&devc->lock,flags); + ad_mute(devc); + spin_unlock_irqrestore(&devc->lock,flags); + + return 0; +} + +static void ad1848_close(int dev) +{ + unsigned long flags; + ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; + ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc; + + DEB(printk("ad1848_close(void)\n")); + + devc->intr_active = 0; + ad1848_halt(dev); + + spin_lock_irqsave(&devc->lock,flags); + + devc->audio_mode = 0; + devc->open_mode &= ~portc->open_mode; + portc->open_mode = 0; + + ad_unmute(devc); + spin_unlock_irqrestore(&devc->lock,flags); +} + +static void ad1848_output_block(int dev, unsigned long buf, int count, int intrflag) +{ + unsigned long flags, cnt; + ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; + ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc; + + cnt = count; + + if (portc->audio_format == AFMT_IMA_ADPCM) + { + cnt /= 4; + } + else + { + if (portc->audio_format & (AFMT_S16_LE | AFMT_S16_BE)) /* 16 bit data */ + cnt >>= 1; + } + if (portc->channels > 1) + cnt >>= 1; + cnt--; + + if ((devc->audio_mode & PCM_ENABLE_OUTPUT) && (audio_devs[dev]->flags & DMA_AUTOMODE) && + intrflag && + cnt == devc->xfer_count) + { + devc->audio_mode |= PCM_ENABLE_OUTPUT; + devc->intr_active = 1; + return; /* + * Auto DMA mode on. No need to react + */ + } + spin_lock_irqsave(&devc->lock,flags); + + ad_write(devc, 15, (unsigned char) (cnt & 0xff)); + ad_write(devc, 14, (unsigned char) ((cnt >> 8) & 0xff)); + + devc->xfer_count = cnt; + devc->audio_mode |= PCM_ENABLE_OUTPUT; + devc->intr_active = 1; + spin_unlock_irqrestore(&devc->lock,flags); +} + +static void ad1848_start_input(int dev, unsigned long buf, int count, int intrflag) +{ + unsigned long flags, cnt; + ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; + ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc; + + cnt = count; + if (portc->audio_format == AFMT_IMA_ADPCM) + { + cnt /= 4; + } + else + { + if (portc->audio_format & (AFMT_S16_LE | AFMT_S16_BE)) /* 16 bit data */ + cnt >>= 1; + } + if (portc->channels > 1) + cnt >>= 1; + cnt--; + + if ((devc->audio_mode & PCM_ENABLE_INPUT) && (audio_devs[dev]->flags & DMA_AUTOMODE) && + intrflag && + cnt == devc->xfer_count) + { + devc->audio_mode |= PCM_ENABLE_INPUT; + devc->intr_active = 1; + return; /* + * Auto DMA mode on. No need to react + */ + } + spin_lock_irqsave(&devc->lock,flags); + + if (devc->model == MD_1848) + { + ad_write(devc, 15, (unsigned char) (cnt & 0xff)); + ad_write(devc, 14, (unsigned char) ((cnt >> 8) & 0xff)); + } + else + { + ad_write(devc, 31, (unsigned char) (cnt & 0xff)); + ad_write(devc, 30, (unsigned char) ((cnt >> 8) & 0xff)); + } + + ad_unmute(devc); + + devc->xfer_count = cnt; + devc->audio_mode |= PCM_ENABLE_INPUT; + devc->intr_active = 1; + spin_unlock_irqrestore(&devc->lock,flags); +} + +static int ad1848_prepare_for_output(int dev, int bsize, int bcount) +{ + int timeout; + unsigned char fs, old_fs, tmp = 0; + unsigned long flags; + ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; + ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc; + + ad_mute(devc); + + spin_lock_irqsave(&devc->lock,flags); + fs = portc->speed_bits | (portc->format_bits << 5); + + if (portc->channels > 1) + fs |= 0x10; + + ad_enter_MCE(devc); /* Enables changes to the format select reg */ + + if (devc->model == MD_1845 || devc->model == MD_1845_SSCAPE) /* Use alternate speed select registers */ + { + fs &= 0xf0; /* Mask off the rate select bits */ + + ad_write(devc, 22, (portc->speed >> 8) & 0xff); /* Speed MSB */ + ad_write(devc, 23, portc->speed & 0xff); /* Speed LSB */ + } + old_fs = ad_read(devc, 8); + + if (devc->model == MD_4232 || devc->model >= MD_4236) + { + tmp = ad_read(devc, 16); + ad_write(devc, 16, tmp | 0x30); + } + if (devc->model == MD_IWAVE) + ad_write(devc, 17, 0xc2); /* Disable variable frequency select */ + + ad_write(devc, 8, fs); + + /* + * Write to I8 starts resynchronization. Wait until it completes. + */ + + timeout = 0; + while (timeout < 100 && inb(devc->base) != 0x80) + timeout++; + timeout = 0; + while (timeout < 10000 && inb(devc->base) == 0x80) + timeout++; + + if (devc->model >= MD_4232) + ad_write(devc, 16, tmp & ~0x30); + + ad_leave_MCE(devc); /* + * Starts the calibration process. + */ + spin_unlock_irqrestore(&devc->lock,flags); + devc->xfer_count = 0; + +#ifndef EXCLUDE_TIMERS + if (dev == timer_installed && devc->timer_running) + if ((fs & 0x01) != (old_fs & 0x01)) + { + ad1848_tmr_reprogram(dev); + } +#endif + ad1848_halt_output(dev); + return 0; +} + +static int ad1848_prepare_for_input(int dev, int bsize, int bcount) +{ + int timeout; + unsigned char fs, old_fs, tmp = 0; + unsigned long flags; + ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; + ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc; + + if (devc->audio_mode) + return 0; + + spin_lock_irqsave(&devc->lock,flags); + fs = portc->speed_bits | (portc->format_bits << 5); + + if (portc->channels > 1) + fs |= 0x10; + + ad_enter_MCE(devc); /* Enables changes to the format select reg */ + + if ((devc->model == MD_1845) || (devc->model == MD_1845_SSCAPE)) /* Use alternate speed select registers */ + { + fs &= 0xf0; /* Mask off the rate select bits */ + + ad_write(devc, 22, (portc->speed >> 8) & 0xff); /* Speed MSB */ + ad_write(devc, 23, portc->speed & 0xff); /* Speed LSB */ + } + if (devc->model == MD_4232) + { + tmp = ad_read(devc, 16); + ad_write(devc, 16, tmp | 0x30); + } + if (devc->model == MD_IWAVE) + ad_write(devc, 17, 0xc2); /* Disable variable frequency select */ + + /* + * If mode >= 2 (CS4231), set I28. It's the capture format register. + */ + + if (devc->model != MD_1848) + { + old_fs = ad_read(devc, 28); + ad_write(devc, 28, fs); + + /* + * Write to I28 starts resynchronization. Wait until it completes. + */ + + timeout = 0; + while (timeout < 100 && inb(devc->base) != 0x80) + timeout++; + + timeout = 0; + while (timeout < 10000 && inb(devc->base) == 0x80) + timeout++; + + if (devc->model != MD_1848 && devc->model != MD_1845 && devc->model != MD_1845_SSCAPE) + { + /* + * CS4231 compatible devices don't have separate sampling rate selection + * register for recording an playback. The I8 register is shared so we have to + * set the speed encoding bits of it too. + */ + unsigned char tmp = portc->speed_bits | (ad_read(devc, 8) & 0xf0); + + ad_write(devc, 8, tmp); + /* + * Write to I8 starts resynchronization. Wait until it completes. + */ + timeout = 0; + while (timeout < 100 && inb(devc->base) != 0x80) + timeout++; + + timeout = 0; + while (timeout < 10000 && inb(devc->base) == 0x80) + timeout++; + } + } + else + { /* For AD1848 set I8. */ + + old_fs = ad_read(devc, 8); + ad_write(devc, 8, fs); + /* + * Write to I8 starts resynchronization. Wait until it completes. + */ + timeout = 0; + while (timeout < 100 && inb(devc->base) != 0x80) + timeout++; + timeout = 0; + while (timeout < 10000 && inb(devc->base) == 0x80) + timeout++; + } + + if (devc->model == MD_4232) + ad_write(devc, 16, tmp & ~0x30); + + ad_leave_MCE(devc); /* + * Starts the calibration process. + */ + spin_unlock_irqrestore(&devc->lock,flags); + devc->xfer_count = 0; + +#ifndef EXCLUDE_TIMERS + if (dev == timer_installed && devc->timer_running) + { + if ((fs & 0x01) != (old_fs & 0x01)) + { + ad1848_tmr_reprogram(dev); + } + } +#endif + ad1848_halt_input(dev); + return 0; +} + +static void ad1848_halt(int dev) +{ + ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; + ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc; + + unsigned char bits = ad_read(devc, 9); + + if (bits & 0x01 && (portc->open_mode & OPEN_WRITE)) + ad1848_halt_output(dev); + + if (bits & 0x02 && (portc->open_mode & OPEN_READ)) + ad1848_halt_input(dev); + devc->audio_mode = 0; +} + +static void ad1848_halt_input(int dev) +{ + ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; + unsigned long flags; + + if (!(ad_read(devc, 9) & 0x02)) + return; /* Capture not enabled */ + + spin_lock_irqsave(&devc->lock,flags); + + ad_mute(devc); + + { + int tmout; + + if(!isa_dma_bridge_buggy) + disable_dma(audio_devs[dev]->dmap_in->dma); + + for (tmout = 0; tmout < 100000; tmout++) + if (ad_read(devc, 11) & 0x10) + break; + ad_write(devc, 9, ad_read(devc, 9) & ~0x02); /* Stop capture */ + + if(!isa_dma_bridge_buggy) + enable_dma(audio_devs[dev]->dmap_in->dma); + devc->audio_mode &= ~PCM_ENABLE_INPUT; + } + + outb(0, io_Status(devc)); /* Clear interrupt status */ + outb(0, io_Status(devc)); /* Clear interrupt status */ + + devc->audio_mode &= ~PCM_ENABLE_INPUT; + + spin_unlock_irqrestore(&devc->lock,flags); +} + +static void ad1848_halt_output(int dev) +{ + ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; + unsigned long flags; + + if (!(ad_read(devc, 9) & 0x01)) + return; /* Playback not enabled */ + + spin_lock_irqsave(&devc->lock,flags); + + ad_mute(devc); + { + int tmout; + + if(!isa_dma_bridge_buggy) + disable_dma(audio_devs[dev]->dmap_out->dma); + + for (tmout = 0; tmout < 100000; tmout++) + if (ad_read(devc, 11) & 0x10) + break; + ad_write(devc, 9, ad_read(devc, 9) & ~0x01); /* Stop playback */ + + if(!isa_dma_bridge_buggy) + enable_dma(audio_devs[dev]->dmap_out->dma); + + devc->audio_mode &= ~PCM_ENABLE_OUTPUT; + } + + outb((0), io_Status(devc)); /* Clear interrupt status */ + outb((0), io_Status(devc)); /* Clear interrupt status */ + + devc->audio_mode &= ~PCM_ENABLE_OUTPUT; + + spin_unlock_irqrestore(&devc->lock,flags); +} + +static void ad1848_trigger(int dev, int state) +{ + ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; + ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc; + unsigned long flags; + unsigned char tmp, old; + + spin_lock_irqsave(&devc->lock,flags); + state &= devc->audio_mode; + + tmp = old = ad_read(devc, 9); + + if (portc->open_mode & OPEN_READ) + { + if (state & PCM_ENABLE_INPUT) + tmp |= 0x02; + else + tmp &= ~0x02; + } + if (portc->open_mode & OPEN_WRITE) + { + if (state & PCM_ENABLE_OUTPUT) + tmp |= 0x01; + else + tmp &= ~0x01; + } + /* ad_mute(devc); */ + if (tmp != old) + { + ad_write(devc, 9, tmp); + ad_unmute(devc); + } + spin_unlock_irqrestore(&devc->lock,flags); +} + +static void ad1848_init_hw(ad1848_info * devc) +{ + int i; + int *init_values; + + /* + * Initial values for the indirect registers of CS4248/AD1848. + */ + static int init_values_a[] = + { + 0xa8, 0xa8, 0x08, 0x08, 0x08, 0x08, 0x00, 0x00, + 0x00, 0x0c, 0x02, 0x00, 0x8a, 0x01, 0x00, 0x00, + + /* Positions 16 to 31 just for CS4231/2 and ad1845 */ + 0x80, 0x00, 0x10, 0x10, 0x00, 0x00, 0x1f, 0x40, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 + }; + + static int init_values_b[] = + { + /* + Values for the newer chips + Some of the register initialization values were changed. In + order to get rid of the click that preceded PCM playback, + calibration was disabled on the 10th byte. On that same byte, + dual DMA was enabled; on the 11th byte, ADC dithering was + enabled, since that is theoretically desirable; on the 13th + byte, Mode 3 was selected, to enable access to extended + registers. + */ + 0xa8, 0xa8, 0x08, 0x08, 0x08, 0x08, 0x00, 0x00, + 0x00, 0x00, 0x06, 0x00, 0xe0, 0x01, 0x00, 0x00, + 0x80, 0x00, 0x10, 0x10, 0x00, 0x00, 0x1f, 0x40, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 + }; + + /* + * Select initialisation data + */ + + init_values = init_values_a; + if(devc->model >= MD_4236) + init_values = init_values_b; + + for (i = 0; i < 16; i++) + ad_write(devc, i, init_values[i]); + + + ad_mute(devc); /* Initialize some variables */ + ad_unmute(devc); /* Leave it unmuted now */ + + if (devc->model > MD_1848) + { + if (devc->model == MD_1845_SSCAPE) + ad_write(devc, 12, ad_read(devc, 12) | 0x50); + else + ad_write(devc, 12, ad_read(devc, 12) | 0x40); /* Mode2 = enabled */ + + if (devc->model == MD_IWAVE) + ad_write(devc, 12, 0x6c); /* Select codec mode 3 */ + + if (devc->model != MD_1845_SSCAPE) + for (i = 16; i < 32; i++) + ad_write(devc, i, init_values[i]); + + if (devc->model == MD_IWAVE) + ad_write(devc, 16, 0x30); /* Playback and capture counters enabled */ + } + if (devc->model > MD_1848) + { + if (devc->audio_flags & DMA_DUPLEX) + ad_write(devc, 9, ad_read(devc, 9) & ~0x04); /* Dual DMA mode */ + else + ad_write(devc, 9, ad_read(devc, 9) | 0x04); /* Single DMA mode */ + + if (devc->model == MD_1845 || devc->model == MD_1845_SSCAPE) + ad_write(devc, 27, ad_read(devc, 27) | 0x08); /* Alternate freq select enabled */ + + if (devc->model == MD_IWAVE) + { /* Some magic Interwave specific initialization */ + ad_write(devc, 12, 0x6c); /* Select codec mode 3 */ + ad_write(devc, 16, 0x30); /* Playback and capture counters enabled */ + ad_write(devc, 17, 0xc2); /* Alternate feature enable */ + } + } + else + { + devc->audio_flags &= ~DMA_DUPLEX; + ad_write(devc, 9, ad_read(devc, 9) | 0x04); /* Single DMA mode */ + if (soundpro) + ad_write(devc, 12, ad_read(devc, 12) | 0x40); /* Mode2 = enabled */ + } + + outb((0), io_Status(devc)); /* Clear pending interrupts */ + + /* + * Toggle the MCE bit. It completes the initialization phase. + */ + + ad_enter_MCE(devc); /* In case the bit was off */ + ad_leave_MCE(devc); + + ad1848_mixer_reset(devc); +} + +int ad1848_detect(struct resource *ports, int *ad_flags, int *osp) +{ + unsigned char tmp; + ad1848_info *devc = &adev_info[nr_ad1848_devs]; + unsigned char tmp1 = 0xff, tmp2 = 0xff; + int optiC930 = 0; /* OPTi 82C930 flag */ + int interwave = 0; + int ad1847_flag = 0; + int cs4248_flag = 0; + int sscape_flag = 0; + int io_base = ports->start; + + int i; + + DDB(printk("ad1848_detect(%x)\n", io_base)); + + if (ad_flags) + { + if (*ad_flags == 0x12345678) + { + interwave = 1; + *ad_flags = 0; + } + + if (*ad_flags == 0x87654321) + { + sscape_flag = 1; + *ad_flags = 0; + } + + if (*ad_flags == 0x12345677) + { + cs4248_flag = 1; + *ad_flags = 0; + } + } + if (nr_ad1848_devs >= MAX_AUDIO_DEV) + { + printk(KERN_ERR "ad1848 - Too many audio devices\n"); + return 0; + } + spin_lock_init(&devc->lock); + devc->base = io_base; + devc->irq_ok = 0; + devc->timer_running = 0; + devc->MCE_bit = 0x40; + devc->irq = 0; + devc->open_mode = 0; + devc->chip_name = devc->name = "AD1848"; + devc->model = MD_1848; /* AD1848 or CS4248 */ + devc->levels = NULL; + devc->debug_flag = 0; + + /* + * Check that the I/O address is in use. + * + * The bit 0x80 of the base I/O port is known to be 0 after the + * chip has performed its power on initialization. Just assume + * this has happened before the OS is starting. + * + * If the I/O address is unused, it typically returns 0xff. + */ + + if (inb(devc->base) == 0xff) + { + DDB(printk("ad1848_detect: The base I/O address appears to be dead\n")); + } + + /* + * Wait for the device to stop initialization + */ + + DDB(printk("ad1848_detect() - step 0\n")); + + for (i = 0; i < 10000000; i++) + { + unsigned char x = inb(devc->base); + + if (x == 0xff || !(x & 0x80)) + break; + } + + DDB(printk("ad1848_detect() - step A\n")); + + if (inb(devc->base) == 0x80) /* Not ready. Let's wait */ + ad_leave_MCE(devc); + + if ((inb(devc->base) & 0x80) != 0x00) /* Not a AD1848 */ + { + DDB(printk("ad1848 detect error - step A (%02x)\n", (int) inb(devc->base))); + return 0; + } + + /* + * Test if it's possible to change contents of the indirect registers. + * Registers 0 and 1 are ADC volume registers. The bit 0x10 is read only + * so try to avoid using it. + */ + + DDB(printk("ad1848_detect() - step B\n")); + ad_write(devc, 0, 0xaa); + ad_write(devc, 1, 0x45); /* 0x55 with bit 0x10 clear */ + + if ((tmp1 = ad_read(devc, 0)) != 0xaa || (tmp2 = ad_read(devc, 1)) != 0x45) + { + if (tmp2 == 0x65) /* AD1847 has couple of bits hardcoded to 1 */ + ad1847_flag = 1; + else + { + DDB(printk("ad1848 detect error - step B (%x/%x)\n", tmp1, tmp2)); + return 0; + } + } + DDB(printk("ad1848_detect() - step C\n")); + ad_write(devc, 0, 0x45); + ad_write(devc, 1, 0xaa); + + if ((tmp1 = ad_read(devc, 0)) != 0x45 || (tmp2 = ad_read(devc, 1)) != 0xaa) + { + if (tmp2 == 0x8a) /* AD1847 has few bits hardcoded to 1 */ + ad1847_flag = 1; + else + { + DDB(printk("ad1848 detect error - step C (%x/%x)\n", tmp1, tmp2)); + return 0; + } + } + + /* + * The indirect register I12 has some read only bits. Let's + * try to change them. + */ + + DDB(printk("ad1848_detect() - step D\n")); + tmp = ad_read(devc, 12); + ad_write(devc, 12, (~tmp) & 0x0f); + + if ((tmp & 0x0f) != ((tmp1 = ad_read(devc, 12)) & 0x0f)) + { + DDB(printk("ad1848 detect error - step D (%x)\n", tmp1)); + return 0; + } + + /* + * NOTE! Last 4 bits of the reg I12 tell the chip revision. + * 0x01=RevB and 0x0A=RevC. + */ + + /* + * The original AD1848/CS4248 has just 15 indirect registers. This means + * that I0 and I16 should return the same value (etc.). + * However this doesn't work with CS4248. Actually it seems to be impossible + * to detect if the chip is a CS4231 or CS4248. + * Ensure that the Mode2 enable bit of I12 is 0. Otherwise this test fails + * with CS4231. + */ + + /* + * OPTi 82C930 has mode2 control bit in another place. This test will fail + * with it. Accept this situation as a possible indication of this chip. + */ + + DDB(printk("ad1848_detect() - step F\n")); + ad_write(devc, 12, 0); /* Mode2=disabled */ + + for (i = 0; i < 16; i++) + { + if ((tmp1 = ad_read(devc, i)) != (tmp2 = ad_read(devc, i + 16))) + { + DDB(printk("ad1848 detect step F(%d/%x/%x) - OPTi chip???\n", i, tmp1, tmp2)); + if (!ad1847_flag) + optiC930 = 1; + break; + } + } + + /* + * Try to switch the chip to mode2 (CS4231) by setting the MODE2 bit (0x40). + * The bit 0x80 is always 1 in CS4248 and CS4231. + */ + + DDB(printk("ad1848_detect() - step G\n")); + + if (ad_flags && *ad_flags == 400) + *ad_flags = 0; + else + ad_write(devc, 12, 0x40); /* Set mode2, clear 0x80 */ + + + if (ad_flags) + *ad_flags = 0; + + tmp1 = ad_read(devc, 12); + if (tmp1 & 0x80) + { + if (ad_flags) + *ad_flags |= AD_F_CS4248; + + devc->chip_name = "CS4248"; /* Our best knowledge just now */ + } + if (optiC930 || (tmp1 & 0xc0) == (0x80 | 0x40)) + { + /* + * CS4231 detected - is it? + * + * Verify that setting I0 doesn't change I16. + */ + + DDB(printk("ad1848_detect() - step H\n")); + ad_write(devc, 16, 0); /* Set I16 to known value */ + + ad_write(devc, 0, 0x45); + if ((tmp1 = ad_read(devc, 16)) != 0x45) /* No change -> CS4231? */ + { + ad_write(devc, 0, 0xaa); + if ((tmp1 = ad_read(devc, 16)) == 0xaa) /* Rotten bits? */ + { + DDB(printk("ad1848 detect error - step H(%x)\n", tmp1)); + return 0; + } + + /* + * Verify that some bits of I25 are read only. + */ + + DDB(printk("ad1848_detect() - step I\n")); + tmp1 = ad_read(devc, 25); /* Original bits */ + ad_write(devc, 25, ~tmp1); /* Invert all bits */ + if ((ad_read(devc, 25) & 0xe7) == (tmp1 & 0xe7)) + { + int id; + + /* + * It's at least CS4231 + */ + + devc->chip_name = "CS4231"; + devc->model = MD_4231; + + /* + * It could be an AD1845 or CS4231A as well. + * CS4231 and AD1845 report the same revision info in I25 + * while the CS4231A reports different. + */ + + id = ad_read(devc, 25); + if ((id & 0xe7) == 0x80) /* Device busy??? */ + id = ad_read(devc, 25); + if ((id & 0xe7) == 0x80) /* Device still busy??? */ + id = ad_read(devc, 25); + DDB(printk("ad1848_detect() - step J (%02x/%02x)\n", id, ad_read(devc, 25))); + + if ((id & 0xe7) == 0x80) { + /* + * It must be a CS4231 or AD1845. The register I23 of + * CS4231 is undefined and it appears to be read only. + * AD1845 uses I23 for setting sample rate. Assume + * the chip is AD1845 if I23 is changeable. + */ + + unsigned char tmp = ad_read(devc, 23); + ad_write(devc, 23, ~tmp); + + if (interwave) + { + devc->model = MD_IWAVE; + devc->chip_name = "IWave"; + } + else if (ad_read(devc, 23) != tmp) /* AD1845 ? */ + { + devc->chip_name = "AD1845"; + devc->model = MD_1845; + } + else if (cs4248_flag) + { + if (ad_flags) + *ad_flags |= AD_F_CS4248; + devc->chip_name = "CS4248"; + devc->model = MD_1848; + ad_write(devc, 12, ad_read(devc, 12) & ~0x40); /* Mode2 off */ + } + ad_write(devc, 23, tmp); /* Restore */ + } + else + { + switch (id & 0x1f) { + case 3: /* CS4236/CS4235/CS42xB/CS4239 */ + { + int xid; + ad_write(devc, 12, ad_read(devc, 12) | 0x60); /* switch to mode 3 */ + ad_write(devc, 23, 0x9c); /* select extended register 25 */ + xid = inb(io_Indexed_Data(devc)); + ad_write(devc, 12, ad_read(devc, 12) & ~0x60); /* back to mode 0 */ + switch (xid & 0x1f) + { + case 0x00: + devc->chip_name = "CS4237B(B)"; + devc->model = MD_42xB; + break; + case 0x08: + /* Seems to be a 4238 ?? */ + devc->chip_name = "CS4238"; + devc->model = MD_42xB; + break; + case 0x09: + devc->chip_name = "CS4238B"; + devc->model = MD_42xB; + break; + case 0x0b: + devc->chip_name = "CS4236B"; + devc->model = MD_4236; + break; + case 0x10: + devc->chip_name = "CS4237B"; + devc->model = MD_42xB; + break; + case 0x1d: + devc->chip_name = "CS4235"; + devc->model = MD_4235; + break; + case 0x1e: + devc->chip_name = "CS4239"; + devc->model = MD_4239; + break; + default: + printk("Chip ident is %X.\n", xid&0x1F); + devc->chip_name = "CS42xx"; + devc->model = MD_4232; + break; + } + } + break; + + case 2: /* CS4232/CS4232A */ + devc->chip_name = "CS4232"; + devc->model = MD_4232; + break; + + case 0: + if ((id & 0xe0) == 0xa0) + { + devc->chip_name = "CS4231A"; + devc->model = MD_4231A; + } + else + { + devc->chip_name = "CS4321"; + devc->model = MD_4231; + } + break; + + default: /* maybe */ + DDB(printk("ad1848: I25 = %02x/%02x\n", ad_read(devc, 25), ad_read(devc, 25) & 0xe7)); + if (optiC930) + { + devc->chip_name = "82C930"; + devc->model = MD_C930; + } + else + { + devc->chip_name = "CS4231"; + devc->model = MD_4231; + } + } + } + } + ad_write(devc, 25, tmp1); /* Restore bits */ + + DDB(printk("ad1848_detect() - step K\n")); + } + } else if (tmp1 == 0x0a) { + /* + * Is it perhaps a SoundPro CMI8330? + * If so, then we should be able to change indirect registers + * greater than I15 after activating MODE2, even though reading + * back I12 does not show it. + */ + + /* + * Let's try comparing register values + */ + for (i = 0; i < 16; i++) { + if ((tmp1 = ad_read(devc, i)) != (tmp2 = ad_read(devc, i + 16))) { + DDB(printk("ad1848 detect step H(%d/%x/%x) - SoundPro chip?\n", i, tmp1, tmp2)); + soundpro = 1; + devc->chip_name = "SoundPro CMI 8330"; + break; + } + } + } + + DDB(printk("ad1848_detect() - step L\n")); + if (ad_flags) + { + if (devc->model != MD_1848) + *ad_flags |= AD_F_CS4231; + } + DDB(printk("ad1848_detect() - Detected OK\n")); + + if (devc->model == MD_1848 && ad1847_flag) + devc->chip_name = "AD1847"; + + + if (sscape_flag == 1) + devc->model = MD_1845_SSCAPE; + + return 1; +} + +int ad1848_init (char *name, struct resource *ports, int irq, int dma_playback, + int dma_capture, int share_dma, int *osp, struct module *owner) +{ + /* + * NOTE! If irq < 0, there is another driver which has allocated the IRQ + * so that this driver doesn't need to allocate/deallocate it. + * The actually used IRQ is ABS(irq). + */ + + int my_dev; + char dev_name[100]; + int e; + + ad1848_info *devc = &adev_info[nr_ad1848_devs]; + + ad1848_port_info *portc = NULL; + + devc->irq = (irq > 0) ? irq : 0; + devc->open_mode = 0; + devc->timer_ticks = 0; + devc->dma1 = dma_playback; + devc->dma2 = dma_capture; + devc->subtype = cfg.card_subtype; + devc->audio_flags = DMA_AUTOMODE; + devc->playback_dev = devc->record_dev = 0; + if (name != NULL) + devc->name = name; + + if (name != NULL && name[0] != 0) + sprintf(dev_name, + "%s (%s)", name, devc->chip_name); + else + sprintf(dev_name, + "Generic audio codec (%s)", devc->chip_name); + + rename_region(ports, devc->name); + + conf_printf2(dev_name, devc->base, devc->irq, dma_playback, dma_capture); + + if (devc->model == MD_1848 || devc->model == MD_C930) + devc->audio_flags |= DMA_HARDSTOP; + + if (devc->model > MD_1848) + { + if (devc->dma1 == devc->dma2 || devc->dma2 == -1 || devc->dma1 == -1) + devc->audio_flags &= ~DMA_DUPLEX; + else + devc->audio_flags |= DMA_DUPLEX; + } + + portc = kmalloc(sizeof(ad1848_port_info), GFP_KERNEL); + if(portc==NULL) { + release_region(devc->base, 4); + return -1; + } + + if ((my_dev = sound_install_audiodrv(AUDIO_DRIVER_VERSION, + dev_name, + &ad1848_audio_driver, + sizeof(struct audio_driver), + devc->audio_flags, + ad_format_mask[devc->model], + devc, + dma_playback, + dma_capture)) < 0) + { + release_region(devc->base, 4); + kfree(portc); + return -1; + } + + audio_devs[my_dev]->portc = portc; + audio_devs[my_dev]->mixer_dev = -1; + if (owner) + audio_devs[my_dev]->d->owner = owner; + memset((char *) portc, 0, sizeof(*portc)); + + nr_ad1848_devs++; + + ad1848_init_hw(devc); + + if (irq > 0) + { + devc->dev_no = my_dev; + if (request_irq(devc->irq, adintr, 0, devc->name, + (void *)(long)my_dev) < 0) + { + printk(KERN_WARNING "ad1848: Unable to allocate IRQ\n"); + /* Don't free it either then.. */ + devc->irq = 0; + } + if (capabilities[devc->model].flags & CAP_F_TIMER) + { +#ifndef CONFIG_SMP + int x; + unsigned char tmp = ad_read(devc, 16); +#endif + + devc->timer_ticks = 0; + + ad_write(devc, 21, 0x00); /* Timer MSB */ + ad_write(devc, 20, 0x10); /* Timer LSB */ +#ifndef CONFIG_SMP + ad_write(devc, 16, tmp | 0x40); /* Enable timer */ + for (x = 0; x < 100000 && devc->timer_ticks == 0; x++); + ad_write(devc, 16, tmp & ~0x40); /* Disable timer */ + + if (devc->timer_ticks == 0) + printk(KERN_WARNING "ad1848: Interrupt test failed (IRQ%d)\n", irq); + else + { + DDB(printk("Interrupt test OK\n")); + devc->irq_ok = 1; + } +#else + devc->irq_ok = 1; +#endif + } + else + devc->irq_ok = 1; /* Couldn't test. assume it's OK */ + } else if (irq < 0) + irq2dev[-irq] = devc->dev_no = my_dev; + +#ifndef EXCLUDE_TIMERS + if ((capabilities[devc->model].flags & CAP_F_TIMER) && + devc->irq_ok) + ad1848_tmr_install(my_dev); +#endif + + if (!share_dma) + { + if (sound_alloc_dma(dma_playback, devc->name)) + printk(KERN_WARNING "ad1848.c: Can't allocate DMA%d\n", dma_playback); + + if (dma_capture != dma_playback) + if (sound_alloc_dma(dma_capture, devc->name)) + printk(KERN_WARNING "ad1848.c: Can't allocate DMA%d\n", dma_capture); + } + + if ((e = sound_install_mixer(MIXER_DRIVER_VERSION, + dev_name, + &ad1848_mixer_operations, + sizeof(struct mixer_operations), + devc)) >= 0) + { + audio_devs[my_dev]->mixer_dev = e; + if (owner) + mixer_devs[e]->owner = owner; + } + return my_dev; +} + +int ad1848_control(int cmd, int arg) +{ + ad1848_info *devc; + unsigned long flags; + + if (nr_ad1848_devs < 1) + return -ENODEV; + + devc = &adev_info[nr_ad1848_devs - 1]; + + switch (cmd) + { + case AD1848_SET_XTAL: /* Change clock frequency of AD1845 (only ) */ + if (devc->model != MD_1845 && devc->model != MD_1845_SSCAPE) + return -EINVAL; + spin_lock_irqsave(&devc->lock,flags); + ad_enter_MCE(devc); + ad_write(devc, 29, (ad_read(devc, 29) & 0x1f) | (arg << 5)); + ad_leave_MCE(devc); + spin_unlock_irqrestore(&devc->lock,flags); + break; + + case AD1848_MIXER_REROUTE: + { + int o = (arg >> 8) & 0xff; + int n = arg & 0xff; + + if (o < 0 || o >= SOUND_MIXER_NRDEVICES) + return -EINVAL; + + if (!(devc->supported_devices & (1 << o)) && + !(devc->supported_rec_devices & (1 << o))) + return -EINVAL; + + if (n == SOUND_MIXER_NONE) + { /* Just hide this control */ + ad1848_mixer_set(devc, o, 0); /* Shut up it */ + devc->supported_devices &= ~(1 << o); + devc->supported_rec_devices &= ~(1 << o); + break; + } + + /* Make the mixer control identified by o to appear as n */ + if (n < 0 || n >= SOUND_MIXER_NRDEVICES) + return -EINVAL; + + devc->mixer_reroute[n] = o; /* Rename the control */ + if (devc->supported_devices & (1 << o)) + devc->supported_devices |= (1 << n); + if (devc->supported_rec_devices & (1 << o)) + devc->supported_rec_devices |= (1 << n); + + devc->supported_devices &= ~(1 << o); + devc->supported_rec_devices &= ~(1 << o); + } + break; + } + return 0; +} + +void ad1848_unload(int io_base, int irq, int dma_playback, int dma_capture, int share_dma) +{ + int i, mixer, dev = 0; + ad1848_info *devc = NULL; + + for (i = 0; devc == NULL && i < nr_ad1848_devs; i++) + { + if (adev_info[i].base == io_base) + { + devc = &adev_info[i]; + dev = devc->dev_no; + } + } + + if (devc != NULL) + { + kfree(audio_devs[dev]->portc); + release_region(devc->base, 4); + + if (!share_dma) + { + if (devc->irq > 0) /* There is no point in freeing irq, if it wasn't allocated */ + free_irq(devc->irq, (void *)(long)devc->dev_no); + + sound_free_dma(dma_playback); + + if (dma_playback != dma_capture) + sound_free_dma(dma_capture); + + } + mixer = audio_devs[devc->dev_no]->mixer_dev; + if(mixer>=0) + sound_unload_mixerdev(mixer); + + nr_ad1848_devs--; + for ( ; i < nr_ad1848_devs ; i++) + adev_info[i] = adev_info[i+1]; + } + else + printk(KERN_ERR "ad1848: Can't find device to be unloaded. Base=%x\n", io_base); +} + +static irqreturn_t adintr(int irq, void *dev_id) +{ + unsigned char status; + ad1848_info *devc; + int dev; + int alt_stat = 0xff; + unsigned char c930_stat = 0; + int cnt = 0; + + dev = (long)dev_id; + devc = (ad1848_info *) audio_devs[dev]->devc; + +interrupt_again: /* Jump back here if int status doesn't reset */ + + status = inb(io_Status(devc)); + + if (status == 0x80) + printk(KERN_DEBUG "adintr: Why?\n"); + if (devc->model == MD_1848) + outb((0), io_Status(devc)); /* Clear interrupt status */ + + if (status & 0x01) + { + if (devc->model == MD_C930) + { /* 82C930 has interrupt status register in MAD16 register MC11 */ + + spin_lock(&devc->lock); + + /* 0xe0e is C930 address port + * 0xe0f is C930 data port + */ + outb(11, 0xe0e); + c930_stat = inb(0xe0f); + outb((~c930_stat), 0xe0f); + + spin_unlock(&devc->lock); + + alt_stat = (c930_stat << 2) & 0x30; + } + else if (devc->model != MD_1848) + { + spin_lock(&devc->lock); + alt_stat = ad_read(devc, 24); + ad_write(devc, 24, ad_read(devc, 24) & ~alt_stat); /* Selective ack */ + spin_unlock(&devc->lock); + } + + if ((devc->open_mode & OPEN_READ) && (devc->audio_mode & PCM_ENABLE_INPUT) && (alt_stat & 0x20)) + { + DMAbuf_inputintr(devc->record_dev); + } + if ((devc->open_mode & OPEN_WRITE) && (devc->audio_mode & PCM_ENABLE_OUTPUT) && + (alt_stat & 0x10)) + { + DMAbuf_outputintr(devc->playback_dev, 1); + } + if (devc->model != MD_1848 && (alt_stat & 0x40)) /* Timer interrupt */ + { + devc->timer_ticks++; +#ifndef EXCLUDE_TIMERS + if (timer_installed == dev && devc->timer_running) + sound_timer_interrupt(); +#endif + } + } +/* + * Sometimes playback or capture interrupts occur while a timer interrupt + * is being handled. The interrupt will not be retriggered if we don't + * handle it now. Check if an interrupt is still pending and restart + * the handler in this case. + */ + if (inb(io_Status(devc)) & 0x01 && cnt++ < 4) + { + goto interrupt_again; + } + return IRQ_HANDLED; +} + +/* + * Experimental initialization sequence for the integrated sound system + * of the Compaq Deskpro M. + */ + +static int init_deskpro_m(struct address_info *hw_config) +{ + unsigned char tmp; + + if ((tmp = inb(0xc44)) == 0xff) + { + DDB(printk("init_deskpro_m: Dead port 0xc44\n")); + return 0; + } + + outb(0x10, 0xc44); + outb(0x40, 0xc45); + outb(0x00, 0xc46); + outb(0xe8, 0xc47); + outb(0x14, 0xc44); + outb(0x40, 0xc45); + outb(0x00, 0xc46); + outb(0xe8, 0xc47); + outb(0x10, 0xc44); + + return 1; +} + +/* + * Experimental initialization sequence for the integrated sound system + * of Compaq Deskpro XL. + */ + +static int init_deskpro(struct address_info *hw_config) +{ + unsigned char tmp; + + if ((tmp = inb(0xc44)) == 0xff) + { + DDB(printk("init_deskpro: Dead port 0xc44\n")); + return 0; + } + outb((tmp | 0x04), 0xc44); /* Select bank 1 */ + if (inb(0xc44) != 0x04) + { + DDB(printk("init_deskpro: Invalid bank1 signature in port 0xc44\n")); + return 0; + } + /* + * OK. It looks like a Deskpro so let's proceed. + */ + + /* + * I/O port 0xc44 Audio configuration register. + * + * bits 0xc0: Audio revision bits + * 0x00 = Compaq Business Audio + * 0x40 = MS Sound System Compatible (reset default) + * 0x80 = Reserved + * 0xc0 = Reserved + * bit 0x20: No Wait State Enable + * 0x00 = Disabled (reset default, DMA mode) + * 0x20 = Enabled (programmed I/O mode) + * bit 0x10: MS Sound System Decode Enable + * 0x00 = Decoding disabled (reset default) + * 0x10 = Decoding enabled + * bit 0x08: FM Synthesis Decode Enable + * 0x00 = Decoding Disabled (reset default) + * 0x08 = Decoding enabled + * bit 0x04 Bank select + * 0x00 = Bank 0 + * 0x04 = Bank 1 + * bits 0x03 MSS Base address + * 0x00 = 0x530 (reset default) + * 0x01 = 0x604 + * 0x02 = 0xf40 + * 0x03 = 0xe80 + */ + +#ifdef DEBUGXL + /* Debug printing */ + printk("Port 0xc44 (before): "); + outb((tmp & ~0x04), 0xc44); + printk("%02x ", inb(0xc44)); + outb((tmp | 0x04), 0xc44); + printk("%02x\n", inb(0xc44)); +#endif + + /* Set bank 1 of the register */ + tmp = 0x58; /* MSS Mode, MSS&FM decode enabled */ + + switch (hw_config->io_base) + { + case 0x530: + tmp |= 0x00; + break; + case 0x604: + tmp |= 0x01; + break; + case 0xf40: + tmp |= 0x02; + break; + case 0xe80: + tmp |= 0x03; + break; + default: + DDB(printk("init_deskpro: Invalid MSS port %x\n", hw_config->io_base)); + return 0; + } + outb((tmp & ~0x04), 0xc44); /* Write to bank=0 */ + +#ifdef DEBUGXL + /* Debug printing */ + printk("Port 0xc44 (after): "); + outb((tmp & ~0x04), 0xc44); /* Select bank=0 */ + printk("%02x ", inb(0xc44)); + outb((tmp | 0x04), 0xc44); /* Select bank=1 */ + printk("%02x\n", inb(0xc44)); +#endif + + /* + * I/O port 0xc45 FM Address Decode/MSS ID Register. + * + * bank=0, bits 0xfe: FM synthesis Decode Compare bits 7:1 (default=0x88) + * bank=0, bit 0x01: SBIC Power Control Bit + * 0x00 = Powered up + * 0x01 = Powered down + * bank=1, bits 0xfc: MSS ID (default=0x40) + */ + +#ifdef DEBUGXL + /* Debug printing */ + printk("Port 0xc45 (before): "); + outb((tmp & ~0x04), 0xc44); /* Select bank=0 */ + printk("%02x ", inb(0xc45)); + outb((tmp | 0x04), 0xc44); /* Select bank=1 */ + printk("%02x\n", inb(0xc45)); +#endif + + outb((tmp & ~0x04), 0xc44); /* Select bank=0 */ + outb((0x88), 0xc45); /* FM base 7:0 = 0x88 */ + outb((tmp | 0x04), 0xc44); /* Select bank=1 */ + outb((0x10), 0xc45); /* MSS ID = 0x10 (MSS port returns 0x04) */ + +#ifdef DEBUGXL + /* Debug printing */ + printk("Port 0xc45 (after): "); + outb((tmp & ~0x04), 0xc44); /* Select bank=0 */ + printk("%02x ", inb(0xc45)); + outb((tmp | 0x04), 0xc44); /* Select bank=1 */ + printk("%02x\n", inb(0xc45)); +#endif + + + /* + * I/O port 0xc46 FM Address Decode/Address ASIC Revision Register. + * + * bank=0, bits 0xff: FM synthesis Decode Compare bits 15:8 (default=0x03) + * bank=1, bits 0xff: Audio addressing ASIC id + */ + +#ifdef DEBUGXL + /* Debug printing */ + printk("Port 0xc46 (before): "); + outb((tmp & ~0x04), 0xc44); /* Select bank=0 */ + printk("%02x ", inb(0xc46)); + outb((tmp | 0x04), 0xc44); /* Select bank=1 */ + printk("%02x\n", inb(0xc46)); +#endif + + outb((tmp & ~0x04), 0xc44); /* Select bank=0 */ + outb((0x03), 0xc46); /* FM base 15:8 = 0x03 */ + outb((tmp | 0x04), 0xc44); /* Select bank=1 */ + outb((0x11), 0xc46); /* ASIC ID = 0x11 */ + +#ifdef DEBUGXL + /* Debug printing */ + printk("Port 0xc46 (after): "); + outb((tmp & ~0x04), 0xc44); /* Select bank=0 */ + printk("%02x ", inb(0xc46)); + outb((tmp | 0x04), 0xc44); /* Select bank=1 */ + printk("%02x\n", inb(0xc46)); +#endif + + /* + * I/O port 0xc47 FM Address Decode Register. + * + * bank=0, bits 0xff: Decode enable selection for various FM address bits + * bank=1, bits 0xff: Reserved + */ + +#ifdef DEBUGXL + /* Debug printing */ + printk("Port 0xc47 (before): "); + outb((tmp & ~0x04), 0xc44); /* Select bank=0 */ + printk("%02x ", inb(0xc47)); + outb((tmp | 0x04), 0xc44); /* Select bank=1 */ + printk("%02x\n", inb(0xc47)); +#endif + + outb((tmp & ~0x04), 0xc44); /* Select bank=0 */ + outb((0x7c), 0xc47); /* FM decode enable bits = 0x7c */ + outb((tmp | 0x04), 0xc44); /* Select bank=1 */ + outb((0x00), 0xc47); /* Reserved bank1 = 0x00 */ + +#ifdef DEBUGXL + /* Debug printing */ + printk("Port 0xc47 (after): "); + outb((tmp & ~0x04), 0xc44); /* Select bank=0 */ + printk("%02x ", inb(0xc47)); + outb((tmp | 0x04), 0xc44); /* Select bank=1 */ + printk("%02x\n", inb(0xc47)); +#endif + + /* + * I/O port 0xc6f = Audio Disable Function Register + */ + +#ifdef DEBUGXL + printk("Port 0xc6f (before) = %02x\n", inb(0xc6f)); +#endif + + outb((0x80), 0xc6f); + +#ifdef DEBUGXL + printk("Port 0xc6f (after) = %02x\n", inb(0xc6f)); +#endif + + return 1; +} + +int probe_ms_sound(struct address_info *hw_config, struct resource *ports) +{ + unsigned char tmp; + + DDB(printk("Entered probe_ms_sound(%x, %d)\n", hw_config->io_base, hw_config->card_subtype)); + + if (hw_config->card_subtype == 1) /* Has no IRQ/DMA registers */ + { + /* check_opl3(0x388, hw_config); */ + return ad1848_detect(ports, NULL, hw_config->osp); + } + + if (deskpro_xl && hw_config->card_subtype == 2) /* Compaq Deskpro XL */ + { + if (!init_deskpro(hw_config)) + return 0; + } + + if (deskpro_m) /* Compaq Deskpro M */ + { + if (!init_deskpro_m(hw_config)) + return 0; + } + + /* + * Check if the IO port returns valid signature. The original MS Sound + * system returns 0x04 while some cards (AudioTrix Pro for example) + * return 0x00 or 0x0f. + */ + + if ((tmp = inb(hw_config->io_base + 3)) == 0xff) /* Bus float */ + { + int ret; + + DDB(printk("I/O address is inactive (%x)\n", tmp)); + if (!(ret = ad1848_detect(ports, NULL, hw_config->osp))) + return 0; + return 1; + } + DDB(printk("MSS signature = %x\n", tmp & 0x3f)); + if ((tmp & 0x3f) != 0x04 && + (tmp & 0x3f) != 0x0f && + (tmp & 0x3f) != 0x00) + { + int ret; + + MDB(printk(KERN_ERR "No MSS signature detected on port 0x%x (0x%x)\n", hw_config->io_base, (int) inb(hw_config->io_base + 3))); + DDB(printk("Trying to detect codec anyway but IRQ/DMA may not work\n")); + if (!(ret = ad1848_detect(ports, NULL, hw_config->osp))) + return 0; + + hw_config->card_subtype = 1; + return 1; + } + if ((hw_config->irq != 5) && + (hw_config->irq != 7) && + (hw_config->irq != 9) && + (hw_config->irq != 10) && + (hw_config->irq != 11) && + (hw_config->irq != 12)) + { + printk(KERN_ERR "MSS: Bad IRQ %d\n", hw_config->irq); + return 0; + } + if (hw_config->dma != 0 && hw_config->dma != 1 && hw_config->dma != 3) + { + printk(KERN_ERR "MSS: Bad DMA %d\n", hw_config->dma); + return 0; + } + /* + * Check that DMA0 is not in use with a 8 bit board. + */ + + if (hw_config->dma == 0 && inb(hw_config->io_base + 3) & 0x80) + { + printk(KERN_ERR "MSS: Can't use DMA0 with a 8 bit card/slot\n"); + return 0; + } + if (hw_config->irq > 7 && hw_config->irq != 9 && inb(hw_config->io_base + 3) & 0x80) + { + printk(KERN_ERR "MSS: Can't use IRQ%d with a 8 bit card/slot\n", hw_config->irq); + return 0; + } + return ad1848_detect(ports, NULL, hw_config->osp); +} + +void attach_ms_sound(struct address_info *hw_config, struct resource *ports, struct module *owner) +{ + static signed char interrupt_bits[12] = + { + -1, -1, -1, -1, -1, 0x00, -1, 0x08, -1, 0x10, 0x18, 0x20 + }; + signed char bits; + char dma2_bit = 0; + + static char dma_bits[4] = + { + 1, 2, 0, 3 + }; + + int config_port = hw_config->io_base + 0; + int version_port = hw_config->io_base + 3; + int dma = hw_config->dma; + int dma2 = hw_config->dma2; + + if (hw_config->card_subtype == 1) /* Has no IRQ/DMA registers */ + { + hw_config->slots[0] = ad1848_init("MS Sound System", ports, + hw_config->irq, + hw_config->dma, + hw_config->dma2, 0, + hw_config->osp, + owner); + return; + } + /* + * Set the IRQ and DMA addresses. + */ + + bits = interrupt_bits[hw_config->irq]; + if (bits == -1) + { + printk(KERN_ERR "MSS: Bad IRQ %d\n", hw_config->irq); + release_region(ports->start, 4); + release_region(ports->start - 4, 4); + return; + } + outb((bits | 0x40), config_port); + if ((inb(version_port) & 0x40) == 0) + printk(KERN_ERR "[MSS: IRQ Conflict?]\n"); + +/* + * Handle the capture DMA channel + */ + + if (dma2 != -1 && dma2 != dma) + { + if (!((dma == 0 && dma2 == 1) || + (dma == 1 && dma2 == 0) || + (dma == 3 && dma2 == 0))) + { /* Unsupported combination. Try to swap channels */ + int tmp = dma; + + dma = dma2; + dma2 = tmp; + } + if ((dma == 0 && dma2 == 1) || + (dma == 1 && dma2 == 0) || + (dma == 3 && dma2 == 0)) + { + dma2_bit = 0x04; /* Enable capture DMA */ + } + else + { + printk(KERN_WARNING "MSS: Invalid capture DMA\n"); + dma2 = dma; + } + } + else + { + dma2 = dma; + } + + hw_config->dma = dma; + hw_config->dma2 = dma2; + + outb((bits | dma_bits[dma] | dma2_bit), config_port); /* Write IRQ+DMA setup */ + + hw_config->slots[0] = ad1848_init("MS Sound System", ports, + hw_config->irq, + dma, dma2, 0, + hw_config->osp, + THIS_MODULE); +} + +void unload_ms_sound(struct address_info *hw_config) +{ + ad1848_unload(hw_config->io_base + 4, + hw_config->irq, + hw_config->dma, + hw_config->dma2, 0); + sound_unload_audiodev(hw_config->slots[0]); + release_region(hw_config->io_base, 4); +} + +#ifndef EXCLUDE_TIMERS + +/* + * Timer stuff (for /dev/music). + */ + +static unsigned int current_interval; + +static unsigned int ad1848_tmr_start(int dev, unsigned int usecs) +{ + unsigned long flags; + ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; + unsigned long xtal_nsecs; /* nanoseconds per xtal oscillator tick */ + unsigned long divider; + + spin_lock_irqsave(&devc->lock,flags); + + /* + * Length of the timer interval (in nanoseconds) depends on the + * selected crystal oscillator. Check this from bit 0x01 of I8. + * + * AD1845 has just one oscillator which has cycle time of 10.050 us + * (when a 24.576 MHz xtal oscillator is used). + * + * Convert requested interval to nanoseconds before computing + * the timer divider. + */ + + if (devc->model == MD_1845 || devc->model == MD_1845_SSCAPE) + xtal_nsecs = 10050; + else if (ad_read(devc, 8) & 0x01) + xtal_nsecs = 9920; + else + xtal_nsecs = 9969; + + divider = (usecs * 1000 + xtal_nsecs / 2) / xtal_nsecs; + + if (divider < 100) /* Don't allow shorter intervals than about 1ms */ + divider = 100; + + if (divider > 65535) /* Overflow check */ + divider = 65535; + + ad_write(devc, 21, (divider >> 8) & 0xff); /* Set upper bits */ + ad_write(devc, 20, divider & 0xff); /* Set lower bits */ + ad_write(devc, 16, ad_read(devc, 16) | 0x40); /* Start the timer */ + devc->timer_running = 1; + spin_unlock_irqrestore(&devc->lock,flags); + + return current_interval = (divider * xtal_nsecs + 500) / 1000; +} + +static void ad1848_tmr_reprogram(int dev) +{ + /* + * Audio driver has changed sampling rate so that a different xtal + * oscillator was selected. We have to reprogram the timer rate. + */ + + ad1848_tmr_start(dev, current_interval); + sound_timer_syncinterval(current_interval); +} + +static void ad1848_tmr_disable(int dev) +{ + unsigned long flags; + ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; + + spin_lock_irqsave(&devc->lock,flags); + ad_write(devc, 16, ad_read(devc, 16) & ~0x40); + devc->timer_running = 0; + spin_unlock_irqrestore(&devc->lock,flags); +} + +static void ad1848_tmr_restart(int dev) +{ + unsigned long flags; + ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; + + if (current_interval == 0) + return; + + spin_lock_irqsave(&devc->lock,flags); + ad_write(devc, 16, ad_read(devc, 16) | 0x40); + devc->timer_running = 1; + spin_unlock_irqrestore(&devc->lock,flags); +} + +static struct sound_lowlev_timer ad1848_tmr = +{ + 0, + 2, + ad1848_tmr_start, + ad1848_tmr_disable, + ad1848_tmr_restart +}; + +static int ad1848_tmr_install(int dev) +{ + if (timer_installed != -1) + return 0; /* Don't install another timer */ + + timer_installed = ad1848_tmr.dev = dev; + sound_timer_init(&ad1848_tmr, audio_devs[dev]->name); + + return 1; +} +#endif /* EXCLUDE_TIMERS */ + +EXPORT_SYMBOL(ad1848_detect); +EXPORT_SYMBOL(ad1848_init); +EXPORT_SYMBOL(ad1848_unload); +EXPORT_SYMBOL(ad1848_control); +EXPORT_SYMBOL(probe_ms_sound); +EXPORT_SYMBOL(attach_ms_sound); +EXPORT_SYMBOL(unload_ms_sound); + +static int __initdata io = -1; +static int __initdata irq = -1; +static int __initdata dma = -1; +static int __initdata dma2 = -1; +static int __initdata type = 0; + +module_param(io, int, 0); /* I/O for a raw AD1848 card */ +module_param(irq, int, 0); /* IRQ to use */ +module_param(dma, int, 0); /* First DMA channel */ +module_param(dma2, int, 0); /* Second DMA channel */ +module_param(type, int, 0); /* Card type */ +module_param(deskpro_xl, bool, 0); /* Special magic for Deskpro XL boxen */ +module_param(deskpro_m, bool, 0); /* Special magic for Deskpro M box */ +module_param(soundpro, bool, 0); /* More special magic for SoundPro chips */ + +#ifdef CONFIG_PNP +module_param(isapnp, int, 0); +module_param(isapnpjump, int, 0); +module_param(reverse, bool, 0); +MODULE_PARM_DESC(isapnp, "When set to 0, Plug & Play support will be disabled"); +MODULE_PARM_DESC(isapnpjump, "Jumps to a specific slot in the driver's PnP table. Use the source, Luke."); +MODULE_PARM_DESC(reverse, "When set to 1, will reverse ISAPnP search order"); + +static struct pnp_dev *ad1848_dev = NULL; + +/* Please add new entries at the end of the table */ +static struct { + char *name; + unsigned short card_vendor, card_device, + vendor, function; + short mss_io, irq, dma, dma2; /* index into isapnp table */ + int type; +} ad1848_isapnp_list[] __initdata = { + {"CMI 8330 SoundPRO", + ISAPNP_VENDOR('C','M','I'), ISAPNP_DEVICE(0x0001), + ISAPNP_VENDOR('@','@','@'), ISAPNP_FUNCTION(0x0001), + 0, 0, 0,-1, 0}, + {"CS4232 based card", + ISAPNP_ANY_ID, ISAPNP_ANY_ID, + ISAPNP_VENDOR('C','S','C'), ISAPNP_FUNCTION(0x0000), + 0, 0, 0, 1, 0}, + {"CS4232 based card", + ISAPNP_ANY_ID, ISAPNP_ANY_ID, + ISAPNP_VENDOR('C','S','C'), ISAPNP_FUNCTION(0x0100), + 0, 0, 0, 1, 0}, + {"OPL3-SA2 WSS mode", + ISAPNP_ANY_ID, ISAPNP_ANY_ID, + ISAPNP_VENDOR('Y','M','H'), ISAPNP_FUNCTION(0x0021), + 1, 0, 0, 1, 1}, + {"Advanced Gravis InterWave Audio", + ISAPNP_VENDOR('G','R','V'), ISAPNP_DEVICE(0x0001), + ISAPNP_VENDOR('G','R','V'), ISAPNP_FUNCTION(0x0000), + 0, 0, 0, 1, 0}, + {NULL} +}; + +static struct isapnp_device_id id_table[] __devinitdata = { + { ISAPNP_VENDOR('C','M','I'), ISAPNP_DEVICE(0x0001), + ISAPNP_VENDOR('@','@','@'), ISAPNP_FUNCTION(0x0001), 0 }, + { ISAPNP_ANY_ID, ISAPNP_ANY_ID, + ISAPNP_VENDOR('C','S','C'), ISAPNP_FUNCTION(0x0000), 0 }, + { ISAPNP_ANY_ID, ISAPNP_ANY_ID, + ISAPNP_VENDOR('C','S','C'), ISAPNP_FUNCTION(0x0100), 0 }, + /* The main driver for this card is opl3sa2 + { ISAPNP_ANY_ID, ISAPNP_ANY_ID, + ISAPNP_VENDOR('Y','M','H'), ISAPNP_FUNCTION(0x0021), 0 }, + */ + { ISAPNP_VENDOR('G','R','V'), ISAPNP_DEVICE(0x0001), + ISAPNP_VENDOR('G','R','V'), ISAPNP_FUNCTION(0x0000), 0 }, + {0} +}; + +MODULE_DEVICE_TABLE(isapnp, id_table); + +static struct pnp_dev *activate_dev(char *devname, char *resname, struct pnp_dev *dev) +{ + int err; + + err = pnp_device_attach(dev); + if (err < 0) + return(NULL); + + if((err = pnp_activate_dev(dev)) < 0) { + printk(KERN_ERR "ad1848: %s %s config failed (out of resources?)[%d]\n", devname, resname, err); + + pnp_device_detach(dev); + + return(NULL); + } + audio_activated = 1; + return(dev); +} + +static struct pnp_dev __init *ad1848_init_generic(struct pnp_card *bus, + struct address_info *hw_config, int slot) +{ + + /* Configure Audio device */ + if((ad1848_dev = pnp_find_dev(bus, ad1848_isapnp_list[slot].vendor, ad1848_isapnp_list[slot].function, NULL))) + { + if((ad1848_dev = activate_dev(ad1848_isapnp_list[slot].name, "ad1848", ad1848_dev))) + { + hw_config->io_base = pnp_port_start(ad1848_dev, ad1848_isapnp_list[slot].mss_io); + hw_config->irq = pnp_irq(ad1848_dev, ad1848_isapnp_list[slot].irq); + hw_config->dma = pnp_dma(ad1848_dev, ad1848_isapnp_list[slot].dma); + if(ad1848_isapnp_list[slot].dma2 != -1) + hw_config->dma2 = pnp_dma(ad1848_dev, ad1848_isapnp_list[slot].dma2); + else + hw_config->dma2 = -1; + hw_config->card_subtype = ad1848_isapnp_list[slot].type; + } else + return(NULL); + } else + return(NULL); + + return(ad1848_dev); +} + +static int __init ad1848_isapnp_init(struct address_info *hw_config, struct pnp_card *bus, int slot) +{ + char *busname = bus->name[0] ? bus->name : ad1848_isapnp_list[slot].name; + + /* Initialize this baby. */ + + if(ad1848_init_generic(bus, hw_config, slot)) { + /* We got it. */ + + printk(KERN_NOTICE "ad1848: PnP reports '%s' at i/o %#x, irq %d, dma %d, %d\n", + busname, + hw_config->io_base, hw_config->irq, hw_config->dma, + hw_config->dma2); + return 1; + } + return 0; +} + +static int __init ad1848_isapnp_probe(struct address_info *hw_config) +{ + static int first = 1; + int i; + + /* Count entries in sb_isapnp_list */ + for (i = 0; ad1848_isapnp_list[i].card_vendor != 0; i++); + i--; + + /* Check and adjust isapnpjump */ + if( isapnpjump < 0 || isapnpjump > i) { + isapnpjump = reverse ? i : 0; + printk(KERN_ERR "ad1848: Valid range for isapnpjump is 0-%d. Adjusted to %d.\n", i, isapnpjump); + } + + if(!first || !reverse) + i = isapnpjump; + first = 0; + while(ad1848_isapnp_list[i].card_vendor != 0) { + static struct pnp_card *bus = NULL; + + while ((bus = pnp_find_card( + ad1848_isapnp_list[i].card_vendor, + ad1848_isapnp_list[i].card_device, + bus))) { + + if(ad1848_isapnp_init(hw_config, bus, i)) { + isapnpjump = i; /* start next search from here */ + return 0; + } + } + i += reverse ? -1 : 1; + } + + return -ENODEV; +} +#endif + + +static int __init init_ad1848(void) +{ + printk(KERN_INFO "ad1848/cs4248 codec driver Copyright (C) by Hannu Savolainen 1993-1996\n"); + +#ifdef CONFIG_PNP + if(isapnp && (ad1848_isapnp_probe(&cfg) < 0) ) { + printk(KERN_NOTICE "ad1848: No ISAPnP cards found, trying standard ones...\n"); + isapnp = 0; + } +#endif + + if(io != -1) { + struct resource *ports; + if( isapnp == 0 ) + { + if(irq == -1 || dma == -1) { + printk(KERN_WARNING "ad1848: must give I/O , IRQ and DMA.\n"); + return -EINVAL; + } + + cfg.irq = irq; + cfg.io_base = io; + cfg.dma = dma; + cfg.dma2 = dma2; + cfg.card_subtype = type; + } + + ports = request_region(io + 4, 4, "ad1848"); + + if (!ports) + return -EBUSY; + + if (!request_region(io, 4, "WSS config")) { + release_region(io + 4, 4); + return -EBUSY; + } + + if (!probe_ms_sound(&cfg, ports)) { + release_region(io + 4, 4); + release_region(io, 4); + return -ENODEV; + } + attach_ms_sound(&cfg, ports, THIS_MODULE); + loaded = 1; + } + return 0; +} + +static void __exit cleanup_ad1848(void) +{ + if(loaded) + unload_ms_sound(&cfg); + +#ifdef CONFIG_PNP + if(ad1848_dev){ + if(audio_activated) + pnp_device_detach(ad1848_dev); + } +#endif +} + +module_init(init_ad1848); +module_exit(cleanup_ad1848); + +#ifndef MODULE +static int __init setup_ad1848(char *str) +{ + /* io, irq, dma, dma2, type */ + int ints[6]; + + str = get_options(str, ARRAY_SIZE(ints), ints); + + io = ints[1]; + irq = ints[2]; + dma = ints[3]; + dma2 = ints[4]; + type = ints[5]; + + return 1; +} + +__setup("ad1848=", setup_ad1848); +#endif +MODULE_LICENSE("GPL"); diff --git a/sound/oss/ad1848.h b/sound/oss/ad1848.h new file mode 100644 index 00000000..b95ebe28 --- /dev/null +++ b/sound/oss/ad1848.h @@ -0,0 +1,24 @@ + +#include <linux/interrupt.h> + +#define AD_F_CS4231 0x0001 /* Returned if a CS4232 (or compatible) detected */ +#define AD_F_CS4248 0x0001 /* Returned if a CS4248 (or compatible) detected */ + +#define AD1848_SET_XTAL 1 +#define AD1848_MIXER_REROUTE 2 + +#define AD1848_REROUTE(oldctl, newctl) \ + ad1848_control(AD1848_MIXER_REROUTE, ((oldctl)<<8)|(newctl)) + + +int ad1848_init(char *name, struct resource *ports, int irq, int dma_playback, + int dma_capture, int share_dma, int *osp, struct module *owner); +void ad1848_unload (int io_base, int irq, int dma_playback, int dma_capture, int share_dma); + +int ad1848_detect (struct resource *ports, int *flags, int *osp); +int ad1848_control(int cmd, int arg); + +void attach_ms_sound(struct address_info * hw_config, struct resource *ports, struct module * owner); + +int probe_ms_sound(struct address_info *hw_config, struct resource *ports); +void unload_ms_sound(struct address_info *hw_info); diff --git a/sound/oss/ad1848_mixer.h b/sound/oss/ad1848_mixer.h new file mode 100644 index 00000000..2cf719b5 --- /dev/null +++ b/sound/oss/ad1848_mixer.h @@ -0,0 +1,253 @@ +/* + * sound/oss/ad1848_mixer.h + * + * Definitions for the mixer of AD1848 and compatible codecs. + */ + +/* + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + */ + + +/* + * The AD1848 codec has generic input lines called Line, Aux1 and Aux2. + * Sound card manufacturers have connected actual inputs (CD, synth, line, + * etc) to these inputs in different order. Therefore it's difficult + * to assign mixer channels to these inputs correctly. The following + * contains two alternative mappings. The first one is for GUS MAX and + * the second is just a generic one (line1, line2 and line3). + * (Actually this is not a mapping but rather some kind of interleaving + * solution). + */ +#define MODE1_REC_DEVICES (SOUND_MASK_LINE3 | SOUND_MASK_MIC | \ + SOUND_MASK_LINE1 | SOUND_MASK_IMIX) + +#define SPRO_REC_DEVICES (SOUND_MASK_LINE | SOUND_MASK_MIC | \ + SOUND_MASK_CD | SOUND_MASK_LINE1) + +#define MODE1_MIXER_DEVICES (SOUND_MASK_LINE1 | SOUND_MASK_MIC | \ + SOUND_MASK_LINE2 | \ + SOUND_MASK_IGAIN | \ + SOUND_MASK_PCM | SOUND_MASK_IMIX) + +#define MODE2_MIXER_DEVICES (SOUND_MASK_LINE1 | SOUND_MASK_LINE2 | \ + SOUND_MASK_MIC | \ + SOUND_MASK_LINE3 | SOUND_MASK_SPEAKER | \ + SOUND_MASK_IGAIN | \ + SOUND_MASK_PCM | SOUND_MASK_IMIX) + +#define MODE3_MIXER_DEVICES (MODE2_MIXER_DEVICES | SOUND_MASK_VOLUME) + +/* OPTi 82C930 has no IMIX level control, but it can still be selected as an + * input + */ +#define C930_MIXER_DEVICES (SOUND_MASK_LINE1 | SOUND_MASK_LINE2 | \ + SOUND_MASK_MIC | SOUND_MASK_VOLUME | \ + SOUND_MASK_LINE3 | \ + SOUND_MASK_IGAIN | SOUND_MASK_PCM) + +#define SPRO_MIXER_DEVICES (SOUND_MASK_VOLUME | SOUND_MASK_PCM | \ + SOUND_MASK_LINE | SOUND_MASK_SYNTH | \ + SOUND_MASK_CD | SOUND_MASK_MIC | \ + SOUND_MASK_SPEAKER | SOUND_MASK_LINE1 | \ + SOUND_MASK_OGAIN) + +struct mixer_def { + unsigned int regno:6; /* register number for volume */ + unsigned int polarity:1; /* volume polarity: 0=normal, 1=reversed */ + unsigned int bitpos:3; /* position of bits in register for volume */ + unsigned int nbits:3; /* number of bits in register for volume */ + unsigned int mutereg:6; /* register number for mute bit */ + unsigned int mutepol:1; /* mute polarity: 0=normal, 1=reversed */ + unsigned int mutepos:4; /* position of mute bit in register */ + unsigned int recreg:6; /* register number for recording bit */ + unsigned int recpol:1; /* recording polarity: 0=normal, 1=reversed */ + unsigned int recpos:4; /* position of recording bit in register */ +}; + +static char mix_cvt[101] = { + 0, 0, 3, 7,10,13,16,19,21,23,26,28,30,32,34,35,37,39,40,42, + 43,45,46,47,49,50,51,52,53,55,56,57,58,59,60,61,62,63,64,65, + 65,66,67,68,69,70,70,71,72,73,73,74,75,75,76,77,77,78,79,79, + 80,81,81,82,82,83,84,84,85,85,86,86,87,87,88,88,89,89,90,90, + 91,91,92,92,93,93,94,94,95,95,96,96,96,97,97,98,98,98,99,99, + 100 +}; + +typedef struct mixer_def mixer_ent; +typedef mixer_ent mixer_ents[2]; + +/* + * Most of the mixer entries work in backwards. Setting the polarity field + * makes them to work correctly. + * + * The channel numbering used by individual sound cards is not fixed. Some + * cards have assigned different meanings for the AUX1, AUX2 and LINE inputs. + * The current version doesn't try to compensate this. + */ + +#define MIX_ENT(name, reg_l, pola_l, pos_l, len_l, reg_r, pola_r, pos_r, len_r, mute_bit) \ + [name] = {{reg_l, pola_l, pos_l, len_l, reg_l, 0, mute_bit, 0, 0, 8}, \ + {reg_r, pola_r, pos_r, len_r, reg_r, 0, mute_bit, 0, 0, 8}} + +#define MIX_ENT2(name, reg_l, pola_l, pos_l, len_l, mute_reg_l, mute_pola_l, mute_pos_l, \ + rec_reg_l, rec_pola_l, rec_pos_l, \ + reg_r, pola_r, pos_r, len_r, mute_reg_r, mute_pola_r, mute_pos_r, \ + rec_reg_r, rec_pola_r, rec_pos_r) \ + [name] = {{reg_l, pola_l, pos_l, len_l, mute_reg_l, mute_pola_l, mute_pos_l, \ + rec_reg_l, rec_pola_l, rec_pos_l}, \ + {reg_r, pola_r, pos_r, len_r, mute_reg_r, mute_pola_r, mute_pos_r, \ + rec_reg_r, rec_pola_r, rec_pos_r}} + +static mixer_ents ad1848_mix_devices[32] = { + MIX_ENT(SOUND_MIXER_VOLUME, 27, 1, 0, 4, 29, 1, 0, 4, 8), + MIX_ENT(SOUND_MIXER_BASS, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_TREBLE, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_SYNTH, 4, 1, 0, 5, 5, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_PCM, 6, 1, 0, 6, 7, 1, 0, 6, 7), + MIX_ENT(SOUND_MIXER_SPEAKER, 26, 1, 0, 4, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_LINE, 18, 1, 0, 5, 19, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_MIC, 0, 0, 5, 1, 1, 0, 5, 1, 8), + MIX_ENT(SOUND_MIXER_CD, 2, 1, 0, 5, 3, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_IMIX, 13, 1, 2, 6, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_ALTPCM, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_RECLEV, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_IGAIN, 0, 0, 0, 4, 1, 0, 0, 4, 8), + MIX_ENT(SOUND_MIXER_OGAIN, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_LINE1, 2, 1, 0, 5, 3, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_LINE2, 4, 1, 0, 5, 5, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_LINE3, 18, 1, 0, 5, 19, 1, 0, 5, 7) +}; + +static mixer_ents iwave_mix_devices[32] = { + MIX_ENT(SOUND_MIXER_VOLUME, 25, 1, 0, 5, 27, 1, 0, 5, 8), + MIX_ENT(SOUND_MIXER_BASS, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_TREBLE, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_SYNTH, 4, 1, 0, 5, 5, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_PCM, 6, 1, 0, 6, 7, 1, 0, 6, 7), + MIX_ENT(SOUND_MIXER_SPEAKER, 26, 1, 0, 4, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_LINE, 18, 1, 0, 5, 19, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_MIC, 0, 0, 5, 1, 1, 0, 5, 1, 8), + MIX_ENT(SOUND_MIXER_CD, 2, 1, 0, 5, 3, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_IMIX, 16, 1, 0, 5, 17, 1, 0, 5, 8), + MIX_ENT(SOUND_MIXER_ALTPCM, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_RECLEV, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_IGAIN, 0, 0, 0, 4, 1, 0, 0, 4, 8), + MIX_ENT(SOUND_MIXER_OGAIN, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_LINE1, 2, 1, 0, 5, 3, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_LINE2, 4, 1, 0, 5, 5, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_LINE3, 18, 1, 0, 5, 19, 1, 0, 5, 7) +}; + +static mixer_ents cs42xb_mix_devices[32] = { + /* Digital master volume actually has seven bits, but we only use + six to avoid the discontinuity when the analog gain kicks in. */ + MIX_ENT(SOUND_MIXER_VOLUME, 46, 1, 0, 6, 47, 1, 0, 6, 7), + MIX_ENT(SOUND_MIXER_BASS, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_TREBLE, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_SYNTH, 4, 1, 0, 5, 5, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_PCM, 6, 1, 0, 6, 7, 1, 0, 6, 7), + MIX_ENT(SOUND_MIXER_SPEAKER, 26, 1, 0, 4, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_LINE, 18, 1, 0, 5, 19, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_MIC, 34, 1, 0, 5, 35, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_CD, 2, 1, 0, 5, 3, 1, 0, 5, 7), + /* For the IMIX entry, it was not possible to use the MIX_ENT macro + because the mute bit is in different positions for the two + channels and requires reverse polarity. */ + [SOUND_MIXER_IMIX] = {{13, 1, 2, 6, 13, 1, 0, 0, 0, 8}, + {42, 1, 0, 6, 42, 1, 7, 0, 0, 8}}, + MIX_ENT(SOUND_MIXER_ALTPCM, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_RECLEV, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_IGAIN, 0, 0, 0, 4, 1, 0, 0, 4, 8), + MIX_ENT(SOUND_MIXER_OGAIN, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_LINE1, 2, 1, 0, 5, 3, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_LINE2, 4, 1, 0, 5, 5, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_LINE3, 38, 1, 0, 6, 39, 1, 0, 6, 7) +}; + +/* OPTi 82C930 has somewhat different port addresses. + * Note: VOLUME == SPEAKER, SYNTH == LINE2, LINE == LINE3, CD == LINE1 + * VOLUME, SYNTH, LINE, CD are not enabled above. + * MIC is level of mic monitoring direct to output. Same for CD, LINE, etc. + */ +static mixer_ents c930_mix_devices[32] = { + MIX_ENT(SOUND_MIXER_VOLUME, 22, 1, 1, 5, 23, 1, 1, 5, 7), + MIX_ENT(SOUND_MIXER_BASS, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_TREBLE, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_SYNTH, 4, 1, 1, 4, 5, 1, 1, 4, 7), + MIX_ENT(SOUND_MIXER_PCM, 6, 1, 0, 5, 7, 1, 0, 5, 7), + MIX_ENT(SOUND_MIXER_SPEAKER, 22, 1, 1, 5, 23, 1, 1, 5, 7), + MIX_ENT(SOUND_MIXER_LINE, 18, 1, 1, 4, 19, 1, 1, 4, 7), + MIX_ENT(SOUND_MIXER_MIC, 20, 1, 1, 4, 21, 1, 1, 4, 7), + MIX_ENT(SOUND_MIXER_CD, 2, 1, 1, 4, 3, 1, 1, 4, 7), + MIX_ENT(SOUND_MIXER_IMIX, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_ALTPCM, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_RECLEV, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_IGAIN, 0, 0, 0, 4, 1, 0, 0, 4, 8), + MIX_ENT(SOUND_MIXER_OGAIN, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT(SOUND_MIXER_LINE1, 2, 1, 1, 4, 3, 1, 1, 4, 7), + MIX_ENT(SOUND_MIXER_LINE2, 4, 1, 1, 4, 5, 1, 1, 4, 7), + MIX_ENT(SOUND_MIXER_LINE3, 18, 1, 1, 4, 19, 1, 1, 4, 7) +}; + +static mixer_ents spro_mix_devices[32] = { + MIX_ENT (SOUND_MIXER_VOLUME, 19, 0, 4, 4, 19, 0, 0, 4, 8), + MIX_ENT (SOUND_MIXER_BASS, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT (SOUND_MIXER_TREBLE, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT2(SOUND_MIXER_SYNTH, 4, 1, 1, 4, 23, 0, 3, 0, 0, 8, + 5, 1, 1, 4, 23, 0, 3, 0, 0, 8), + MIX_ENT (SOUND_MIXER_PCM, 6, 1, 1, 4, 7, 1, 1, 4, 8), + MIX_ENT (SOUND_MIXER_SPEAKER, 18, 0, 3, 2, 0, 0, 0, 0, 8), + MIX_ENT2(SOUND_MIXER_LINE, 20, 0, 4, 4, 17, 1, 4, 16, 0, 2, + 20, 0, 0, 4, 17, 1, 3, 16, 0, 1), + MIX_ENT2(SOUND_MIXER_MIC, 18, 0, 0, 3, 17, 1, 0, 16, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 0, 0), + MIX_ENT2(SOUND_MIXER_CD, 21, 0, 4, 4, 17, 1, 2, 16, 0, 4, + 21, 0, 0, 4, 17, 1, 1, 16, 0, 3), + MIX_ENT (SOUND_MIXER_IMIX, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT (SOUND_MIXER_ALTPCM, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT (SOUND_MIXER_RECLEV, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT (SOUND_MIXER_IGAIN, 0, 0, 0, 0, 0, 0, 0, 0, 8), + MIX_ENT (SOUND_MIXER_OGAIN, 17, 1, 6, 1, 0, 0, 0, 0, 8), + /* This is external wavetable */ + MIX_ENT2(SOUND_MIXER_LINE1, 22, 0, 4, 4, 23, 1, 1, 23, 0, 4, + 22, 0, 0, 4, 23, 1, 0, 23, 0, 5), +}; + +static int default_mixer_levels[32] = +{ + 0x3232, /* Master Volume */ + 0x3232, /* Bass */ + 0x3232, /* Treble */ + 0x4b4b, /* FM */ + 0x3232, /* PCM */ + 0x1515, /* PC Speaker */ + 0x2020, /* Ext Line */ + 0x1010, /* Mic */ + 0x4b4b, /* CD */ + 0x0000, /* Recording monitor */ + 0x4b4b, /* Second PCM */ + 0x4b4b, /* Recording level */ + 0x4b4b, /* Input gain */ + 0x4b4b, /* Output gain */ + 0x2020, /* Line1 */ + 0x2020, /* Line2 */ + 0x1515 /* Line3 (usually line in)*/ +}; + +#define LEFT_CHN 0 +#define RIGHT_CHN 1 + +/* + * Channel enable bits for ioctl(SOUND_MIXER_PRIVATE1) + */ + +#ifndef AUDIO_SPEAKER +#define AUDIO_SPEAKER 0x01 /* Enable mono output */ +#define AUDIO_HEADPHONE 0x02 /* Sparc only */ +#define AUDIO_LINE_OUT 0x04 /* Sparc only */ +#endif diff --git a/sound/oss/aedsp16.c b/sound/oss/aedsp16.c new file mode 100644 index 00000000..35b5912c --- /dev/null +++ b/sound/oss/aedsp16.c @@ -0,0 +1,1373 @@ +/* + sound/oss/aedsp16.c + + Audio Excel DSP 16 software configuration routines + Copyright (C) 1995,1996,1997,1998 Riccardo Facchetti (fizban@tin.it) + + This program is free software; you can redistribute it and/or modify + it under the terms of the GNU General Public License as published by + the Free Software Foundation; either version 2 of the License, or + (at your option) any later version. + + This program is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + GNU General Public License for more details. + + You should have received a copy of the GNU General Public License + along with this program; if not, write to the Free Software + Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + + */ +/* + * Include the main OSS Lite header file. It include all the os, OSS Lite, etc + * headers needed by this source. + */ +#include <linux/delay.h> +#include <linux/module.h> +#include <linux/init.h> +#include "sound_config.h" + +/* + + READ THIS + + This module started to configure the Audio Excel DSP 16 Sound Card. + Now works with the SC-6000 (old aedsp16) and new SC-6600 based cards. + + NOTE: I have NO idea about Audio Excel DSP 16 III. If someone owns this + audio card and want to see the kernel support for it, please contact me. + + Audio Excel DSP 16 is an SB pro II, Microsoft Sound System and MPU-401 + compatible card. + It is software-only configurable (no jumpers to hard-set irq/dma/mpu-irq), + so before this module, the only way to configure the DSP under linux was + boot the MS-DOS loading the sound.sys device driver (this driver soft- + configure the sound board hardware by massaging someone of its registers), + and then ctrl-alt-del to boot linux with the DSP configured by the DOS + driver. + + This module works configuring your Audio Excel DSP 16's irq, dma and + mpu-401-irq. The OSS Lite routines rely on the fact that if the + hardware is there, they can detect it. The problem with AEDSP16 is + that no hardware can be found by the probe routines if the sound card + is not configured properly. Sometimes the kernel probe routines can find + an SBPRO even when the card is not configured (this is the standard setup + of the card), but the SBPRO emulation don't work well if the card is not + properly initialized. For this reason + + aedsp16_init_board() + + routine is called before the OSS Lite probe routines try to detect the + hardware. + + NOTE (READ THE NOTE TOO, IT CONTAIN USEFUL INFORMATIONS) + + NOTE: Now it works with SC-6000 and SC-6600 based audio cards. The new cards + have no jumper switch at all. No more WSS or MPU-401 I/O port switches. They + have to be configured by software. + + NOTE: The driver is merged with the new OSS Lite sound driver. It works + as a lowlevel driver. + + The Audio Excel DSP 16 Sound Card emulates both SBPRO and MSS; + the OSS Lite sound driver can be configured for SBPRO and MSS cards + at the same time, but the aedsp16 can't be two cards!! + When we configure it, we have to choose the SBPRO or the MSS emulation + for AEDSP16. We also can install a *REAL* card of the other type (see [1]). + + NOTE: If someone can test the combination AEDSP16+MSS or AEDSP16+SBPRO + please let me know if it works. + + The MPU-401 support can be compiled in together with one of the other + two operating modes. + + NOTE: This is something like plug-and-play: we have only to plug + the AEDSP16 board in the socket, and then configure and compile + a kernel that uses the AEDSP16 software configuration capability. + No jumper setting is needed! + + For example, if you want AEDSP16 to be an SBPro, on irq 10, dma 3 + you have just to make config the OSS Lite package, configuring + the AEDSP16 sound card, then activating the SBPro emulation mode + and at last configuring IRQ and DMA. + Compile the kernel and run it. + + NOTE: This means for SC-6000 cards that you can choose irq and dma, + but not the I/O addresses. To change I/O addresses you have to set + them with jumpers. For SC-6600 cards you have no jumpers so you have + to set up your full card configuration in the make config. + + You can change the irq/dma/mirq settings WITHOUT THE NEED to open + your computer and massage the jumpers (there are no irq/dma/mirq + jumpers to be configured anyway, only I/O BASE values have to be + configured with jumpers) + + For some ununderstandable reason, the card default of irq 7, dma 1, + don't work for me. Seems to be an IRQ or DMA conflict. Under heavy + HDD work, the kernel start to erupt out a lot of messages like: + + 'Sound: DMA timed out - IRQ/DRQ config error?' + + For what I can say, I have NOT any conflict at irq 7 (under linux I'm + using the lp polling driver), and dma line 1 is unused as stated by + /proc/dma. I can suppose this is a bug of AEDSP16. I know my hardware so + I'm pretty sure I have not any conflict, but may be I'm wrong. Who knows! + Anyway a setting of irq 10, dma 3 works really fine. + + NOTE: if someone can use AEDSP16 with irq 7, dma 1, please let me know + the emulation mode, all the installed hardware and the hardware + configuration (irq and dma settings of all the hardware). + + This init module should work with SBPRO+MSS, when one of the two is + the AEDSP16 emulation and the other the real card. (see [1]) + For example: + + AEDSP16 (0x220) in SBPRO emu (0x220) + real MSS + other + AEDSP16 (0x220) in MSS emu + real SBPRO (0x240) + other + + MPU401 should work. (see [2]) + + [1] + --- + Date: Mon, 29 Jul 1997 08:35:40 +0100 + From: Mr S J Greenaway <sjg95@unixfe.rl.ac.uk> + + [...] + Just to let you know got my Audio Excel (emulating a MSS) working + with my original SB16, thanks for the driver! + [...] + --- + + [2] Not tested by me for lack of hardware. + + TODO, WISHES AND TECH + + - About I/O ports allocation - + + Request the 2x0h region (port base) in any case if we are using this card. + + NOTE: the "aedsp16 (base)" string with which we are requesting the aedsp16 + port base region (see code) does not mean necessarily that we are emulating + sbpro. Even if this region is the sbpro I/O ports region, we use this + region to access the control registers of the card, and if emulating + sbpro, I/O sbpro registers too. If we are emulating MSS, the sbpro + registers are not used, in no way, to emulate an sbpro: they are + used only for configuration purposes. + + Started Fri Mar 17 16:13:18 MET 1995 + + v0.1 (ALPHA, was a user-level program called AudioExcelDSP16.c) + - Initial code. + v0.2 (ALPHA) + - Cleanups. + - Integrated with Linux voxware v 2.90-2 kernel sound driver. + - SoundBlaster Pro mode configuration. + - Microsoft Sound System mode configuration. + - MPU-401 mode configuration. + v0.3 (ALPHA) + - Cleanups. + - Rearranged the code to let aedsp16_init_board be more general. + - Erased the REALLY_SLOW_IO. We don't need it. Erased the linux/io.h + inclusion too. We rely on os.h + - Used the to get a variable + len string (we are not sure about the len of Copyright string). + This works with any SB and compatible. + - Added the code to request_region at device init (should go in + the main body of voxware). + v0.4 (BETA) + - Better configure.c patch for aedsp16 configuration (better + logic of inclusion of AEDSP16 support) + - Modified the conditional compilation to better support more than + one sound card of the emulated type (read the NOTES above) + - Moved the sb init routine from the attach to the very first + probe in sb_card.c + - Rearrangements and cleanups + - Wiped out some unnecessary code and variables: this is kernel + code so it is better save some TEXT and DATA + - Fixed the request_region code. We must allocate the aedsp16 (sbpro) + I/O ports in any case because they are used to access the DSP + configuration registers and we can not allow anyone to get them. + v0.5 + - cleanups on comments + - prep for diffs against v3.0-proto-950402 + v0.6 + - removed the request_region()s when compiling the MODULE sound.o + because we are not allowed (by the actual voxware structure) to + release_region() + v0.7 (pre ALPHA, not distributed) + - started porting this module to kernel 1.3.84. Dummy probe/attach + routines. + v0.8 (ALPHA) + - attached all the init routines. + v0.9 (BETA) + - Integrated with linux-pre2.0.7 + - Integrated with configuration scripts. + - Cleaned up and beautyfied the code. + v0.9.9 (BETA) + - Thanks to Piercarlo Grandi: corrected the conditonal compilation code. + Now only the code configured is compiled in, with some memory saving. + v0.9.10 + - Integration into the sound/lowlevel/ section of the sound driver. + - Re-organized the code. + v0.9.11 (not distributed) + - Rewritten the init interface-routines to initialize the AEDSP16 in + one shot. + - More cosmetics. + - SC-6600 support. + - More soft/hard configuration. + v0.9.12 + - Refined the v0.9.11 code with conditional compilation to distinguish + between SC-6000 and SC-6600 code. + v1.0.0 + - Prep for merging with OSS Lite and Linux kernel 2.1.13 + - Corrected a bug in request/check/release region calls (thanks to the + new kernel exception handling). + v1.1 + - Revamped for integration with new modularized sound drivers: to enhance + the flexibility of modular version, I have removed all the conditional + compilation for SBPRO, MPU and MSS code. Now it is all managed with + the ae_config structure. + v1.2 + - Module informations added. + - Removed aedsp16_delay_10msec(), now using mdelay(10) + - All data and funcs moved to .*.init section. + v1.3 + Arnaldo Carvalho de Melo <acme@conectiva.com.br> - 2000/09/27 + - got rid of check_region + + Known Problems: + - Audio Excel DSP 16 III don't work with this driver. + + Credits: + Many thanks to Gerald Britton <gbritton@CapAccess.org>. He helped me a + lot in testing the 0.9.11 and 0.9.12 versions of this driver. + + */ + + +#define VERSION "1.3" /* Version of Audio Excel DSP 16 driver */ + +#undef AEDSP16_DEBUG /* Define this to 1 to enable debug code */ +#undef AEDSP16_DEBUG_MORE /* Define this to 1 to enable more debug */ +#undef AEDSP16_INFO /* Define this to 1 to enable info code */ + +#if defined(AEDSP16_DEBUG) +# define DBG(x) printk x +# if defined(AEDSP16_DEBUG_MORE) +# define DBG1(x) printk x +# else +# define DBG1(x) +# endif +#else +# define DBG(x) +# define DBG1(x) +#endif + +/* + * Misc definitions + */ +#define TRUE 1 +#define FALSE 0 + +/* + * Region Size for request/check/release region. + */ +#define IOBASE_REGION_SIZE 0x10 + +/* + * Hardware related defaults + */ +#define DEF_AEDSP16_IOB 0x220 /* 0x220(default) 0x240 */ +#define DEF_AEDSP16_IRQ 7 /* 5 7(default) 9 10 11 */ +#define DEF_AEDSP16_MRQ 0 /* 5 7 9 10 0(default), 0 means disable */ +#define DEF_AEDSP16_DMA 1 /* 0 1(default) 3 */ + +/* + * Commands of AEDSP16's DSP (SBPRO+special). + * Some of them are COMMAND_xx, in the future they may change. + */ +#define WRITE_MDIRQ_CFG 0x50 /* Set M&I&DRQ mask (the real config) */ +#define COMMAND_52 0x52 /* */ +#define READ_HARD_CFG 0x58 /* Read Hardware Config (I/O base etc) */ +#define COMMAND_5C 0x5c /* */ +#define COMMAND_60 0x60 /* */ +#define COMMAND_66 0x66 /* */ +#define COMMAND_6C 0x6c /* */ +#define COMMAND_6E 0x6e /* */ +#define COMMAND_88 0x88 /* */ +#define DSP_INIT_MSS 0x8c /* Enable Microsoft Sound System mode */ +#define COMMAND_C5 0xc5 /* */ +#define GET_DSP_VERSION 0xe1 /* Get DSP Version */ +#define GET_DSP_COPYRIGHT 0xe3 /* Get DSP Copyright */ + +/* + * Offsets of AEDSP16 DSP I/O ports. The offset is added to base I/O port + * to have the actual I/O port. + * Register permissions are: + * (wo) == Write Only + * (ro) == Read Only + * (w-) == Write + * (r-) == Read + */ +#define DSP_RESET 0x06 /* offset of DSP RESET (wo) */ +#define DSP_READ 0x0a /* offset of DSP READ (ro) */ +#define DSP_WRITE 0x0c /* offset of DSP WRITE (w-) */ +#define DSP_COMMAND 0x0c /* offset of DSP COMMAND (w-) */ +#define DSP_STATUS 0x0c /* offset of DSP STATUS (r-) */ +#define DSP_DATAVAIL 0x0e /* offset of DSP DATA AVAILABLE (ro) */ + + +#define RETRY 10 /* Various retry values on I/O opera- */ +#define STATUSRETRY 1000 /* tions. Sometimes we have to */ +#define HARDRETRY 500000 /* wait for previous cmd to complete */ + +/* + * Size of character arrays that store name and version of sound card + */ +#define CARDNAMELEN 15 /* Size of the card's name in chars */ +#define CARDVERLEN 10 /* Size of the card's version in chars */ +#define CARDVERDIGITS 2 /* Number of digits in the version */ + +#if defined(CONFIG_SC6600) +/* + * Bitmapped flags of hard configuration + */ +/* + * Decode macros (xl == low byte, xh = high byte) + */ +#define IOBASE(xl) ((xl & 0x01)?0x240:0x220) +#define JOY(xl) (xl & 0x02) +#define MPUADDR(xl) ( \ + (xl & 0x0C)?0x330: \ + (xl & 0x08)?0x320: \ + (xl & 0x04)?0x310: \ + 0x300) +#define WSSADDR(xl) ((xl & 0x10)?0xE80:0x530) +#define CDROM(xh) (xh & 0x20) +#define CDROMADDR(xh) (((xh & 0x1F) << 4) + 0x200) +/* + * Encode macros + */ +#define BLDIOBASE(xl, val) { \ + xl &= ~0x01; \ + if (val == 0x240) \ + xl |= 0x01; \ + } +#define BLDJOY(xl, val) { \ + xl &= ~0x02; \ + if (val == 1) \ + xl |= 0x02; \ + } +#define BLDMPUADDR(xl, val) { \ + xl &= ~0x0C; \ + switch (val) { \ + case 0x330: \ + xl |= 0x0C; \ + break; \ + case 0x320: \ + xl |= 0x08; \ + break; \ + case 0x310: \ + xl |= 0x04; \ + break; \ + case 0x300: \ + xl |= 0x00; \ + break; \ + default: \ + xl |= 0x00; \ + break; \ + } \ + } +#define BLDWSSADDR(xl, val) { \ + xl &= ~0x10; \ + if (val == 0xE80) \ + xl |= 0x10; \ + } +#define BLDCDROM(xh, val) { \ + xh &= ~0x20; \ + if (val == 1) \ + xh |= 0x20; \ + } +#define BLDCDROMADDR(xh, val) { \ + int tmp = val; \ + tmp -= 0x200; \ + tmp >>= 4; \ + tmp &= 0x1F; \ + xh |= tmp; \ + xh &= 0x7F; \ + xh |= 0x40; \ + } +#endif /* CONFIG_SC6600 */ + +/* + * Bit mapped flags for calling aedsp16_init_board(), and saving the current + * emulation mode. + */ +#define INIT_NONE (0 ) +#define INIT_SBPRO (1<<0) +#define INIT_MSS (1<<1) +#define INIT_MPU401 (1<<2) + +static int soft_cfg __initdata = 0; /* bitmapped config */ +static int soft_cfg_mss __initdata = 0; /* bitmapped mss config */ +static int ver[CARDVERDIGITS] __initdata = {0, 0}; /* DSP Ver: + hi->ver[0] lo->ver[1] */ + +#if defined(CONFIG_SC6600) +static int hard_cfg[2] /* lo<-hard_cfg[0] hi<-hard_cfg[1] */ + __initdata = { 0, 0}; +#endif /* CONFIG_SC6600 */ + +#if defined(CONFIG_SC6600) +/* Decoded hard configuration */ +struct d_hcfg { + int iobase; + int joystick; + int mpubase; + int wssbase; + int cdrom; + int cdrombase; +}; + +static struct d_hcfg decoded_hcfg __initdata = {0, }; + +#endif /* CONFIG_SC6600 */ + +/* orVals contain the values to be or'ed */ +struct orVals { + int val; /* irq|mirq|dma */ + int or; /* soft_cfg |= TheStruct.or */ +}; + +/* aedsp16_info contain the audio card configuration */ +struct aedsp16_info { + int base_io; /* base I/O address for accessing card */ + int irq; /* irq value for DSP I/O */ + int mpu_irq; /* irq for mpu401 interface I/O */ + int dma; /* dma value for DSP I/O */ + int mss_base; /* base I/O for Microsoft Sound System */ + int mpu_base; /* base I/O for MPU-401 emulation */ + int init; /* Initialization status of the card */ +}; + +/* + * Magic values that the DSP will eat when configuring irq/mirq/dma + */ +/* DSP IRQ conversion array */ +static struct orVals orIRQ[] __initdata = { + {0x05, 0x28}, + {0x07, 0x08}, + {0x09, 0x10}, + {0x0a, 0x18}, + {0x0b, 0x20}, + {0x00, 0x00} +}; + +/* MPU-401 IRQ conversion array */ +static struct orVals orMIRQ[] __initdata = { + {0x05, 0x04}, + {0x07, 0x44}, + {0x09, 0x84}, + {0x0a, 0xc4}, + {0x00, 0x00} +}; + +/* DMA Channels conversion array */ +static struct orVals orDMA[] __initdata = { + {0x00, 0x01}, + {0x01, 0x02}, + {0x03, 0x03}, + {0x00, 0x00} +}; + +static struct aedsp16_info ae_config = { + DEF_AEDSP16_IOB, + DEF_AEDSP16_IRQ, + DEF_AEDSP16_MRQ, + DEF_AEDSP16_DMA, + -1, + -1, + INIT_NONE +}; + +/* + * Buffers to store audio card informations + */ +static char DSPCopyright[CARDNAMELEN + 1] __initdata = {0, }; +static char DSPVersion[CARDVERLEN + 1] __initdata = {0, }; + +static int __init aedsp16_wait_data(int port) +{ + int loop = STATUSRETRY; + unsigned char ret = 0; + + DBG1(("aedsp16_wait_data (0x%x): ", port)); + + do { + ret = inb(port + DSP_DATAVAIL); + /* + * Wait for data available (bit 7 of ret == 1) + */ + } while (!(ret & 0x80) && loop--); + + if (ret & 0x80) { + DBG1(("success.\n")); + return TRUE; + } + + DBG1(("failure.\n")); + return FALSE; +} + +static int __init aedsp16_read(int port) +{ + int inbyte; + + DBG((" Read DSP Byte (0x%x): ", port)); + + if (aedsp16_wait_data(port) == FALSE) { + DBG(("failure.\n")); + return -1; + } + + inbyte = inb(port + DSP_READ); + + DBG(("read [0x%x]/{%c}.\n", inbyte, inbyte)); + + return inbyte; +} + +static int __init aedsp16_test_dsp(int port) +{ + return ((aedsp16_read(port) == 0xaa) ? TRUE : FALSE); +} + +static int __init aedsp16_dsp_reset(int port) +{ + /* + * Reset DSP + */ + + DBG(("Reset DSP:\n")); + + outb(1, (port + DSP_RESET)); + udelay(10); + outb(0, (port + DSP_RESET)); + udelay(10); + udelay(10); + if (aedsp16_test_dsp(port) == TRUE) { + DBG(("success.\n")); + return TRUE; + } else + DBG(("failure.\n")); + return FALSE; +} + +static int __init aedsp16_write(int port, int cmd) +{ + unsigned char ret; + int loop = HARDRETRY; + + DBG((" Write DSP Byte (0x%x) [0x%x]: ", port, cmd)); + + do { + ret = inb(port + DSP_STATUS); + /* + * DSP ready to receive data if bit 7 of ret == 0 + */ + if (!(ret & 0x80)) { + outb(cmd, port + DSP_COMMAND); + DBG(("success.\n")); + return 0; + } + } while (loop--); + + DBG(("timeout.\n")); + printk("[AEDSP16] DSP Command (0x%x) timeout.\n", cmd); + + return -1; +} + +#if defined(CONFIG_SC6600) + +#if defined(AEDSP16_INFO) || defined(AEDSP16_DEBUG) +void __init aedsp16_pinfo(void) { + DBG(("\n Base address: %x\n", decoded_hcfg.iobase)); + DBG((" Joystick : %s present\n", decoded_hcfg.joystick?"":" not")); + DBG((" WSS addr : %x\n", decoded_hcfg.wssbase)); + DBG((" MPU-401 addr: %x\n", decoded_hcfg.mpubase)); + DBG((" CDROM : %s present\n", (decoded_hcfg.cdrom!=4)?"":" not")); + DBG((" CDROMADDR : %x\n\n", decoded_hcfg.cdrombase)); +} +#endif + +static void __init aedsp16_hard_decode(void) { + + DBG((" aedsp16_hard_decode: 0x%x, 0x%x\n", hard_cfg[0], hard_cfg[1])); + +/* + * Decode Cfg Bytes. + */ + decoded_hcfg.iobase = IOBASE(hard_cfg[0]); + decoded_hcfg.joystick = JOY(hard_cfg[0]); + decoded_hcfg.wssbase = WSSADDR(hard_cfg[0]); + decoded_hcfg.mpubase = MPUADDR(hard_cfg[0]); + decoded_hcfg.cdrom = CDROM(hard_cfg[1]); + decoded_hcfg.cdrombase = CDROMADDR(hard_cfg[1]); + +#if defined(AEDSP16_INFO) || defined(AEDSP16_DEBUG) + printk(" Original sound card configuration:\n"); + aedsp16_pinfo(); +#endif + +/* + * Now set up the real kernel configuration. + */ + decoded_hcfg.iobase = ae_config.base_io; + decoded_hcfg.wssbase = ae_config.mss_base; + decoded_hcfg.mpubase = ae_config.mpu_base; + +#if defined(CONFIG_SC6600_JOY) + decoded_hcfg.joystick = CONFIG_SC6600_JOY; /* Enable */ +#endif +#if defined(CONFIG_SC6600_CDROM) + decoded_hcfg.cdrom = CONFIG_SC6600_CDROM; /* 4:N-3:I-2:G-1:P-0:S */ +#endif +#if defined(CONFIG_SC6600_CDROMBASE) + decoded_hcfg.cdrombase = CONFIG_SC6600_CDROMBASE; /* 0 Disable */ +#endif + +#if defined(AEDSP16_DEBUG) + DBG((" New Values:\n")); + aedsp16_pinfo(); +#endif + + DBG(("success.\n")); +} + +static void __init aedsp16_hard_encode(void) { + + DBG((" aedsp16_hard_encode: 0x%x, 0x%x\n", hard_cfg[0], hard_cfg[1])); + + hard_cfg[0] = 0; + hard_cfg[1] = 0; + + hard_cfg[0] |= 0x20; + + BLDIOBASE (hard_cfg[0], decoded_hcfg.iobase); + BLDWSSADDR(hard_cfg[0], decoded_hcfg.wssbase); + BLDMPUADDR(hard_cfg[0], decoded_hcfg.mpubase); + BLDJOY(hard_cfg[0], decoded_hcfg.joystick); + BLDCDROM(hard_cfg[1], decoded_hcfg.cdrom); + BLDCDROMADDR(hard_cfg[1], decoded_hcfg.cdrombase); + +#if defined(AEDSP16_DEBUG) + aedsp16_pinfo(); +#endif + + DBG((" aedsp16_hard_encode: 0x%x, 0x%x\n", hard_cfg[0], hard_cfg[1])); + DBG(("success.\n")); + +} + +static int __init aedsp16_hard_write(int port) { + + DBG(("aedsp16_hard_write:\n")); + + if (aedsp16_write(port, COMMAND_6C)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_6C); + DBG(("failure.\n")); + return FALSE; + } + if (aedsp16_write(port, COMMAND_5C)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_5C); + DBG(("failure.\n")); + return FALSE; + } + if (aedsp16_write(port, hard_cfg[0])) { + printk("[AEDSP16] DATA 0x%x: failed!\n", hard_cfg[0]); + DBG(("failure.\n")); + return FALSE; + } + if (aedsp16_write(port, hard_cfg[1])) { + printk("[AEDSP16] DATA 0x%x: failed!\n", hard_cfg[1]); + DBG(("failure.\n")); + return FALSE; + } + if (aedsp16_write(port, COMMAND_C5)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_C5); + DBG(("failure.\n")); + return FALSE; + } + + DBG(("success.\n")); + + return TRUE; +} + +static int __init aedsp16_hard_read(int port) { + + DBG(("aedsp16_hard_read:\n")); + + if (aedsp16_write(port, READ_HARD_CFG)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", READ_HARD_CFG); + DBG(("failure.\n")); + return FALSE; + } + + if ((hard_cfg[0] = aedsp16_read(port)) == -1) { + printk("[AEDSP16] aedsp16_read after CMD 0x%x: failed\n", + READ_HARD_CFG); + DBG(("failure.\n")); + return FALSE; + } + if ((hard_cfg[1] = aedsp16_read(port)) == -1) { + printk("[AEDSP16] aedsp16_read after CMD 0x%x: failed\n", + READ_HARD_CFG); + DBG(("failure.\n")); + return FALSE; + } + if (aedsp16_read(port) == -1) { + printk("[AEDSP16] aedsp16_read after CMD 0x%x: failed\n", + READ_HARD_CFG); + DBG(("failure.\n")); + return FALSE; + } + + DBG(("success.\n")); + + return TRUE; +} + +static int __init aedsp16_ext_cfg_write(int port) { + + int extcfg, val; + + if (aedsp16_write(port, COMMAND_66)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_66); + return FALSE; + } + + extcfg = 7; + if (decoded_hcfg.cdrom != 2) + extcfg = 0x0F; + if ((decoded_hcfg.cdrom == 4) || + (decoded_hcfg.cdrom == 3)) + extcfg &= ~2; + if (decoded_hcfg.cdrombase == 0) + extcfg &= ~2; + if (decoded_hcfg.mpubase == 0) + extcfg &= ~1; + + if (aedsp16_write(port, extcfg)) { + printk("[AEDSP16] Write extcfg: failed!\n"); + return FALSE; + } + if (aedsp16_write(port, 0)) { + printk("[AEDSP16] Write extcfg: failed!\n"); + return FALSE; + } + if (decoded_hcfg.cdrom == 3) { + if (aedsp16_write(port, COMMAND_52)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_52); + return FALSE; + } + if ((val = aedsp16_read(port)) == -1) { + printk("[AEDSP16] aedsp16_read after CMD 0x%x: failed\n" + , COMMAND_52); + return FALSE; + } + val &= 0x7F; + if (aedsp16_write(port, COMMAND_60)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_60); + return FALSE; + } + if (aedsp16_write(port, val)) { + printk("[AEDSP16] Write val: failed!\n"); + return FALSE; + } + } + + return TRUE; +} + +#endif /* CONFIG_SC6600 */ + +static int __init aedsp16_cfg_write(int port) { + if (aedsp16_write(port, WRITE_MDIRQ_CFG)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", WRITE_MDIRQ_CFG); + return FALSE; + } + if (aedsp16_write(port, soft_cfg)) { + printk("[AEDSP16] Initialization of (M)IRQ and DMA: failed!\n"); + return FALSE; + } + return TRUE; +} + +static int __init aedsp16_init_mss(int port) +{ + DBG(("aedsp16_init_mss:\n")); + + mdelay(10); + + if (aedsp16_write(port, DSP_INIT_MSS)) { + printk("[AEDSP16] aedsp16_init_mss [0x%x]: failed!\n", + DSP_INIT_MSS); + DBG(("failure.\n")); + return FALSE; + } + + mdelay(10); + + if (aedsp16_cfg_write(port) == FALSE) + return FALSE; + + outb(soft_cfg_mss, ae_config.mss_base); + + DBG(("success.\n")); + + return TRUE; +} + +static int __init aedsp16_setup_board(int port) { + int loop = RETRY; + +#if defined(CONFIG_SC6600) + int val = 0; + + if (aedsp16_hard_read(port) == FALSE) { + printk("[AEDSP16] aedsp16_hard_read: failed!\n"); + return FALSE; + } + + if (aedsp16_write(port, COMMAND_52)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_52); + return FALSE; + } + + if ((val = aedsp16_read(port)) == -1) { + printk("[AEDSP16] aedsp16_read after CMD 0x%x: failed\n", + COMMAND_52); + return FALSE; + } +#endif + + do { + if (aedsp16_write(port, COMMAND_88)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_88); + return FALSE; + } + mdelay(10); + } while ((aedsp16_wait_data(port) == FALSE) && loop--); + + if (aedsp16_read(port) == -1) { + printk("[AEDSP16] aedsp16_read after CMD 0x%x: failed\n", + COMMAND_88); + return FALSE; + } + +#if !defined(CONFIG_SC6600) + if (aedsp16_write(port, COMMAND_5C)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_5C); + return FALSE; + } +#endif + + if (aedsp16_cfg_write(port) == FALSE) + return FALSE; + +#if defined(CONFIG_SC6600) + if (aedsp16_write(port, COMMAND_60)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_60); + return FALSE; + } + if (aedsp16_write(port, val)) { + printk("[AEDSP16] DATA 0x%x: failed!\n", val); + return FALSE; + } + if (aedsp16_write(port, COMMAND_6E)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_6E); + return FALSE; + } + if (aedsp16_write(port, ver[0])) { + printk("[AEDSP16] DATA 0x%x: failed!\n", ver[0]); + return FALSE; + } + if (aedsp16_write(port, ver[1])) { + printk("[AEDSP16] DATA 0x%x: failed!\n", ver[1]); + return FALSE; + } + + if (aedsp16_hard_write(port) == FALSE) { + printk("[AEDSP16] aedsp16_hard_write: failed!\n"); + return FALSE; + } + + if (aedsp16_write(port, COMMAND_5C)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_5C); + return FALSE; + } + +#if defined(THIS_IS_A_THING_I_HAVE_NOT_TESTED_YET) + if (aedsp16_cfg_write(port) == FALSE) + return FALSE; +#endif + +#endif + + return TRUE; +} + +static int __init aedsp16_stdcfg(int port) { + if (aedsp16_write(port, WRITE_MDIRQ_CFG)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", WRITE_MDIRQ_CFG); + return FALSE; + } + /* + * 0x0A == (IRQ 7, DMA 1, MIRQ 0) + */ + if (aedsp16_write(port, 0x0A)) { + printk("[AEDSP16] aedsp16_stdcfg: failed!\n"); + return FALSE; + } + return TRUE; +} + +static int __init aedsp16_dsp_version(int port) +{ + int len = 0; + int ret; + + DBG(("Get DSP Version:\n")); + + if (aedsp16_write(ae_config.base_io, GET_DSP_VERSION)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", GET_DSP_VERSION); + DBG(("failed.\n")); + return FALSE; + } + + do { + if ((ret = aedsp16_read(port)) == -1) { + DBG(("failed.\n")); + return FALSE; + } + /* + * We already know how many int are stored (2), so we know when the + * string is finished. + */ + ver[len++] = ret; + } while (len < CARDVERDIGITS); + sprintf(DSPVersion, "%d.%d", ver[0], ver[1]); + + DBG(("success.\n")); + + return TRUE; +} + +static int __init aedsp16_dsp_copyright(int port) +{ + int len = 0; + int ret; + + DBG(("Get DSP Copyright:\n")); + + if (aedsp16_write(ae_config.base_io, GET_DSP_COPYRIGHT)) { + printk("[AEDSP16] CMD 0x%x: failed!\n", GET_DSP_COPYRIGHT); + DBG(("failed.\n")); + return FALSE; + } + + do { + if ((ret = aedsp16_read(port)) == -1) { + /* + * If no more data available, return to the caller, no error if len>0. + * We have no other way to know when the string is finished. + */ + if (len) + break; + else { + DBG(("failed.\n")); + return FALSE; + } + } + + DSPCopyright[len++] = ret; + + } while (len < CARDNAMELEN); + + DBG(("success.\n")); + + return TRUE; +} + +static void __init aedsp16_init_tables(void) +{ + int i = 0; + + memset(DSPCopyright, 0, CARDNAMELEN + 1); + memset(DSPVersion, 0, CARDVERLEN + 1); + + for (i = 0; orIRQ[i].or; i++) + if (orIRQ[i].val == ae_config.irq) { + soft_cfg |= orIRQ[i].or; + soft_cfg_mss |= orIRQ[i].or; + } + + for (i = 0; orMIRQ[i].or; i++) + if (orMIRQ[i].or == ae_config.mpu_irq) + soft_cfg |= orMIRQ[i].or; + + for (i = 0; orDMA[i].or; i++) + if (orDMA[i].val == ae_config.dma) { + soft_cfg |= orDMA[i].or; + soft_cfg_mss |= orDMA[i].or; + } +} + +static int __init aedsp16_init_board(void) +{ + aedsp16_init_tables(); + + if (aedsp16_dsp_reset(ae_config.base_io) == FALSE) { + printk("[AEDSP16] aedsp16_dsp_reset: failed!\n"); + return FALSE; + } + if (aedsp16_dsp_copyright(ae_config.base_io) == FALSE) { + printk("[AEDSP16] aedsp16_dsp_copyright: failed!\n"); + return FALSE; + } + + /* + * My AEDSP16 card return SC-6000 in DSPCopyright, so + * if we have something different, we have to be warned. + */ + if (strcmp("SC-6000", DSPCopyright)) + printk("[AEDSP16] Warning: non SC-6000 audio card!\n"); + + if (aedsp16_dsp_version(ae_config.base_io) == FALSE) { + printk("[AEDSP16] aedsp16_dsp_version: failed!\n"); + return FALSE; + } + + if (aedsp16_stdcfg(ae_config.base_io) == FALSE) { + printk("[AEDSP16] aedsp16_stdcfg: failed!\n"); + return FALSE; + } + +#if defined(CONFIG_SC6600) + if (aedsp16_hard_read(ae_config.base_io) == FALSE) { + printk("[AEDSP16] aedsp16_hard_read: failed!\n"); + return FALSE; + } + + aedsp16_hard_decode(); + + aedsp16_hard_encode(); + + if (aedsp16_hard_write(ae_config.base_io) == FALSE) { + printk("[AEDSP16] aedsp16_hard_write: failed!\n"); + return FALSE; + } + + if (aedsp16_ext_cfg_write(ae_config.base_io) == FALSE) { + printk("[AEDSP16] aedsp16_ext_cfg_write: failed!\n"); + return FALSE; + } +#endif /* CONFIG_SC6600 */ + + if (aedsp16_setup_board(ae_config.base_io) == FALSE) { + printk("[AEDSP16] aedsp16_setup_board: failed!\n"); + return FALSE; + } + + if (ae_config.mss_base != -1) { + if (ae_config.init & INIT_MSS) { + if (aedsp16_init_mss(ae_config.base_io) == FALSE) { + printk("[AEDSP16] Can not initialize" + "Microsoft Sound System mode.\n"); + return FALSE; + } + } + } + +#if !defined(MODULE) || defined(AEDSP16_INFO) || defined(AEDSP16_DEBUG) + + printk("Audio Excel DSP 16 init v%s (%s %s) [", + VERSION, DSPCopyright, + DSPVersion); + + if (ae_config.mpu_base != -1) { + if (ae_config.init & INIT_MPU401) { + printk("MPU401"); + if ((ae_config.init & INIT_MSS) || + (ae_config.init & INIT_SBPRO)) + printk(" "); + } + } + + if (ae_config.mss_base == -1) { + if (ae_config.init & INIT_SBPRO) { + printk("SBPro"); + if (ae_config.init & INIT_MSS) + printk(" "); + } + } + + if (ae_config.mss_base != -1) + if (ae_config.init & INIT_MSS) + printk("MSS"); + + printk("]\n"); +#endif /* MODULE || AEDSP16_INFO || AEDSP16_DEBUG */ + + mdelay(10); + + return TRUE; +} + +static int __init init_aedsp16_sb(void) +{ + DBG(("init_aedsp16_sb: ")); + +/* + * If the card is already init'ed MSS, we can not init it to SBPRO too + * because the board can not emulate simultaneously MSS and SBPRO. + */ + if (ae_config.init & INIT_MSS) + return FALSE; + if (ae_config.init & INIT_SBPRO) + return FALSE; + + ae_config.init |= INIT_SBPRO; + + DBG(("done.\n")); + + return TRUE; +} + +static void uninit_aedsp16_sb(void) +{ + DBG(("uninit_aedsp16_sb: ")); + + ae_config.init &= ~INIT_SBPRO; + + DBG(("done.\n")); +} + +static int __init init_aedsp16_mss(void) +{ + DBG(("init_aedsp16_mss: ")); + +/* + * If the card is already init'ed SBPRO, we can not init it to MSS too + * because the board can not emulate simultaneously MSS and SBPRO. + */ + if (ae_config.init & INIT_SBPRO) + return FALSE; + if (ae_config.init & INIT_MSS) + return FALSE; +/* + * We must allocate the CONFIG_AEDSP16_BASE region too because these are the + * I/O ports to access card's control registers. + */ + if (!(ae_config.init & INIT_MPU401)) { + if (!request_region(ae_config.base_io, IOBASE_REGION_SIZE, + "aedsp16 (base)")) { + printk( + "AEDSP16 BASE I/O port region is already in use.\n"); + return FALSE; + } + } + + ae_config.init |= INIT_MSS; + + DBG(("done.\n")); + + return TRUE; +} + +static void uninit_aedsp16_mss(void) +{ + DBG(("uninit_aedsp16_mss: ")); + + if ((!(ae_config.init & INIT_MPU401)) && + (ae_config.init & INIT_MSS)) { + release_region(ae_config.base_io, IOBASE_REGION_SIZE); + DBG(("AEDSP16 base region released.\n")); + } + + ae_config.init &= ~INIT_MSS; + DBG(("done.\n")); +} + +static int __init init_aedsp16_mpu(void) +{ + DBG(("init_aedsp16_mpu: ")); + + if (ae_config.init & INIT_MPU401) + return FALSE; + +/* + * We must request the CONFIG_AEDSP16_BASE region too because these are the I/O + * ports to access card's control registers. + */ + if (!(ae_config.init & (INIT_MSS | INIT_SBPRO))) { + if (!request_region(ae_config.base_io, IOBASE_REGION_SIZE, + "aedsp16 (base)")) { + printk( + "AEDSP16 BASE I/O port region is already in use.\n"); + return FALSE; + } + } + + ae_config.init |= INIT_MPU401; + + DBG(("done.\n")); + + return TRUE; +} + +static void uninit_aedsp16_mpu(void) +{ + DBG(("uninit_aedsp16_mpu: ")); + + if ((!(ae_config.init & (INIT_MSS | INIT_SBPRO))) && + (ae_config.init & INIT_MPU401)) { + release_region(ae_config.base_io, IOBASE_REGION_SIZE); + DBG(("AEDSP16 base region released.\n")); + } + + ae_config.init &= ~INIT_MPU401; + + DBG(("done.\n")); +} + +static int __init init_aedsp16(void) +{ + int initialized = FALSE; + + DBG(("Initializing BASE[0x%x] IRQ[%d] DMA[%d] MIRQ[%d]\n", + ae_config.base_io,ae_config.irq,ae_config.dma,ae_config.mpu_irq)); + + if (ae_config.mss_base == -1) { + if (init_aedsp16_sb() == FALSE) { + uninit_aedsp16_sb(); + } else { + initialized = TRUE; + } + } + + if (ae_config.mpu_base != -1) { + if (init_aedsp16_mpu() == FALSE) { + uninit_aedsp16_mpu(); + } else { + initialized = TRUE; + } + } + +/* + * In the sequence of init routines, the MSS init MUST be the last! + * This because of the special register programming the MSS mode needs. + * A board reset would disable the MSS mode restoring the default SBPRO + * mode. + */ + if (ae_config.mss_base != -1) { + if (init_aedsp16_mss() == FALSE) { + uninit_aedsp16_mss(); + } else { + initialized = TRUE; + } + } + + if (initialized) + initialized = aedsp16_init_board(); + return initialized; +} + +static void __exit uninit_aedsp16(void) +{ + if (ae_config.mss_base != -1) + uninit_aedsp16_mss(); + else + uninit_aedsp16_sb(); + if (ae_config.mpu_base != -1) + uninit_aedsp16_mpu(); +} + +static int __initdata io = -1; +static int __initdata irq = -1; +static int __initdata dma = -1; +static int __initdata mpu_irq = -1; +static int __initdata mss_base = -1; +static int __initdata mpu_base = -1; + +module_param(io, int, 0); +MODULE_PARM_DESC(io, "I/O base address (0x220 0x240)"); +module_param(irq, int, 0); +MODULE_PARM_DESC(irq, "IRQ line (5 7 9 10 11)"); +module_param(dma, int, 0); +MODULE_PARM_DESC(dma, "dma line (0 1 3)"); +module_param(mpu_irq, int, 0); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ line (5 7 9 10 0)"); +module_param(mss_base, int, 0); +MODULE_PARM_DESC(mss_base, "MSS emulation I/O base address (0x530 0xE80)"); +module_param(mpu_base, int, 0); +MODULE_PARM_DESC(mpu_base,"MPU-401 I/O base address (0x300 0x310 0x320 0x330)"); +MODULE_AUTHOR("Riccardo Facchetti <fizban@tin.it>"); +MODULE_DESCRIPTION("Audio Excel DSP 16 Driver Version " VERSION); +MODULE_LICENSE("GPL"); + +static int __init do_init_aedsp16(void) { + printk("Audio Excel DSP 16 init driver Copyright (C) Riccardo Facchetti 1995-98\n"); + if (io == -1 || dma == -1 || irq == -1) { + printk(KERN_INFO "aedsp16: I/O, IRQ and DMA are mandatory\n"); + return -EINVAL; + } + + ae_config.base_io = io; + ae_config.irq = irq; + ae_config.dma = dma; + + ae_config.mss_base = mss_base; + ae_config.mpu_base = mpu_base; + ae_config.mpu_irq = mpu_irq; + + if (init_aedsp16() == FALSE) { + printk(KERN_ERR "aedsp16: initialization failed\n"); + /* + * XXX + * What error should we return here ? + */ + return -EINVAL; + } + return 0; +} + +static void __exit cleanup_aedsp16(void) { + uninit_aedsp16(); +} + +module_init(do_init_aedsp16); +module_exit(cleanup_aedsp16); + +#ifndef MODULE +static int __init setup_aedsp16(char *str) +{ + /* io, irq, dma, mss_io, mpu_io, mpu_irq */ + int ints[7]; + + str = get_options(str, ARRAY_SIZE(ints), ints); + + io = ints[1]; + irq = ints[2]; + dma = ints[3]; + mss_base = ints[4]; + mpu_base = ints[5]; + mpu_irq = ints[6]; + return 1; +} + +__setup("aedsp16=", setup_aedsp16); +#endif diff --git a/sound/oss/audio.c b/sound/oss/audio.c new file mode 100644 index 00000000..4b958b1c --- /dev/null +++ b/sound/oss/audio.c @@ -0,0 +1,985 @@ +/* + * sound/oss/audio.c + * + * Device file manager for /dev/audio + */ + +/* + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + */ +/* + * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed) + * Thomas Sailer : moved several static variables into struct audio_operations + * (which is grossly misnamed btw.) because they have the same + * lifetime as the rest in there and dynamic allocation saves + * 12k or so + * Thomas Sailer : use more logical O_NONBLOCK semantics + * Daniel Rodriksson: reworked the use of the device specific copy_user + * still generic + * Horst von Brand: Add missing #include <linux/string.h> + * Chris Rankin : Update the module-usage counter for the coprocessor, + * and decrement the counters again if we cannot open + * the audio device. + */ + +#include <linux/stddef.h> +#include <linux/string.h> +#include <linux/kmod.h> + +#include "sound_config.h" +#include "ulaw.h" +#include "coproc.h" + +#define NEUTRAL8 0x80 +#define NEUTRAL16 0x00 + + +static int dma_ioctl(int dev, unsigned int cmd, void __user *arg); + +static int set_format(int dev, int fmt) +{ + if (fmt != AFMT_QUERY) + { + audio_devs[dev]->local_conversion = 0; + + if (!(audio_devs[dev]->format_mask & fmt)) /* Not supported */ + { + if (fmt == AFMT_MU_LAW) + { + fmt = AFMT_U8; + audio_devs[dev]->local_conversion = CNV_MU_LAW; + } + else + fmt = AFMT_U8; /* This is always supported */ + } + audio_devs[dev]->audio_format = audio_devs[dev]->d->set_bits(dev, fmt); + audio_devs[dev]->local_format = fmt; + } + else + return audio_devs[dev]->local_format; + + if (audio_devs[dev]->local_conversion) + return audio_devs[dev]->local_conversion; + else + return audio_devs[dev]->local_format; +} + +int audio_open(int dev, struct file *file) +{ + int ret; + int bits; + int dev_type = dev & 0x0f; + int mode = translate_mode(file); + const struct audio_driver *driver; + const struct coproc_operations *coprocessor; + + dev = dev >> 4; + + if (dev_type == SND_DEV_DSP16) + bits = 16; + else + bits = 8; + + if (dev < 0 || dev >= num_audiodevs) + return -ENXIO; + + driver = audio_devs[dev]->d; + + if (!try_module_get(driver->owner)) + return -ENODEV; + + if ((ret = DMAbuf_open(dev, mode)) < 0) + goto error_1; + + if ( (coprocessor = audio_devs[dev]->coproc) != NULL ) { + if (!try_module_get(coprocessor->owner)) + goto error_2; + + if ((ret = coprocessor->open(coprocessor->devc, COPR_PCM)) < 0) { + printk(KERN_WARNING "Sound: Can't access coprocessor device\n"); + goto error_3; + } + } + + audio_devs[dev]->local_conversion = 0; + + if (dev_type == SND_DEV_AUDIO) + set_format(dev, AFMT_MU_LAW); + else + set_format(dev, bits); + + audio_devs[dev]->audio_mode = AM_NONE; + + return 0; + + /* + * Clean-up stack: this is what needs (un)doing if + * we can't open the audio device ... + */ + error_3: + module_put(coprocessor->owner); + + error_2: + DMAbuf_release(dev, mode); + + error_1: + module_put(driver->owner); + + return ret; +} + +static void sync_output(int dev) +{ + int p, i; + int l; + struct dma_buffparms *dmap = audio_devs[dev]->dmap_out; + + if (dmap->fragment_size <= 0) + return; + dmap->flags |= DMA_POST; + + /* Align the write pointer with fragment boundaries */ + + if ((l = dmap->user_counter % dmap->fragment_size) > 0) + { + int len; + unsigned long offs = dmap->user_counter % dmap->bytes_in_use; + + len = dmap->fragment_size - l; + memset(dmap->raw_buf + offs, dmap->neutral_byte, len); + DMAbuf_move_wrpointer(dev, len); + } + + /* + * Clean all unused buffer fragments. + */ + + p = dmap->qtail; + dmap->flags |= DMA_POST; + + for (i = dmap->qlen + 1; i < dmap->nbufs; i++) + { + p = (p + 1) % dmap->nbufs; + if (((dmap->raw_buf + p * dmap->fragment_size) + dmap->fragment_size) > + (dmap->raw_buf + dmap->buffsize)) + printk(KERN_ERR "audio: Buffer error 2\n"); + + memset(dmap->raw_buf + p * dmap->fragment_size, + dmap->neutral_byte, + dmap->fragment_size); + } + + dmap->flags |= DMA_DIRTY; +} + +void audio_release(int dev, struct file *file) +{ + const struct coproc_operations *coprocessor; + int mode = translate_mode(file); + + dev = dev >> 4; + + /* + * We do this in DMAbuf_release(). Why are we doing it + * here? Why don't we test the file mode before setting + * both flags? DMAbuf_release() does. + * ...pester...pester...pester... + */ + audio_devs[dev]->dmap_out->closing = 1; + audio_devs[dev]->dmap_in->closing = 1; + + /* + * We need to make sure we allocated the dmap_out buffer + * before we go mucking around with it in sync_output(). + */ + if (mode & OPEN_WRITE) + sync_output(dev); + + if ( (coprocessor = audio_devs[dev]->coproc) != NULL ) { + coprocessor->close(coprocessor->devc, COPR_PCM); + module_put(coprocessor->owner); + } + DMAbuf_release(dev, mode); + + module_put(audio_devs[dev]->d->owner); +} + +static void translate_bytes(const unsigned char *table, unsigned char *buff, int n) +{ + unsigned long i; + + if (n <= 0) + return; + + for (i = 0; i < n; ++i) + buff[i] = table[buff[i]]; +} + +int audio_write(int dev, struct file *file, const char __user *buf, int count) +{ + int c, p, l, buf_size, used, returned; + int err; + char *dma_buf; + + dev = dev >> 4; + + p = 0; + c = count; + + if(count < 0) + return -EINVAL; + + if (!(audio_devs[dev]->open_mode & OPEN_WRITE)) + return -EPERM; + + if (audio_devs[dev]->flags & DMA_DUPLEX) + audio_devs[dev]->audio_mode |= AM_WRITE; + else + audio_devs[dev]->audio_mode = AM_WRITE; + + if (!count) /* Flush output */ + { + sync_output(dev); + return 0; + } + + while (c) + { + if ((err = DMAbuf_getwrbuffer(dev, &dma_buf, &buf_size, !!(file->f_flags & O_NONBLOCK))) < 0) + { + /* Handle nonblocking mode */ + if ((file->f_flags & O_NONBLOCK) && err == -EAGAIN) + return p? p : -EAGAIN; /* No more space. Return # of accepted bytes */ + return err; + } + l = c; + + if (l > buf_size) + l = buf_size; + + returned = l; + used = l; + if (!audio_devs[dev]->d->copy_user) + { + if ((dma_buf + l) > + (audio_devs[dev]->dmap_out->raw_buf + audio_devs[dev]->dmap_out->buffsize)) + { + printk(KERN_ERR "audio: Buffer error 3 (%lx,%d), (%lx, %d)\n", (long) dma_buf, l, (long) audio_devs[dev]->dmap_out->raw_buf, (int) audio_devs[dev]->dmap_out->buffsize); + return -EDOM; + } + if (dma_buf < audio_devs[dev]->dmap_out->raw_buf) + { + printk(KERN_ERR "audio: Buffer error 13 (%lx<%lx)\n", (long) dma_buf, (long) audio_devs[dev]->dmap_out->raw_buf); + return -EDOM; + } + if(copy_from_user(dma_buf, &(buf)[p], l)) + return -EFAULT; + } + else audio_devs[dev]->d->copy_user (dev, + dma_buf, 0, + buf, p, + c, buf_size, + &used, &returned, + l); + l = returned; + + if (audio_devs[dev]->local_conversion & CNV_MU_LAW) + { + translate_bytes(ulaw_dsp, (unsigned char *) dma_buf, l); + } + c -= used; + p += used; + DMAbuf_move_wrpointer(dev, l); + + } + + return count; +} + +int audio_read(int dev, struct file *file, char __user *buf, int count) +{ + int c, p, l; + char *dmabuf; + int buf_no; + + dev = dev >> 4; + p = 0; + c = count; + + if (!(audio_devs[dev]->open_mode & OPEN_READ)) + return -EPERM; + + if ((audio_devs[dev]->audio_mode & AM_WRITE) && !(audio_devs[dev]->flags & DMA_DUPLEX)) + sync_output(dev); + + if (audio_devs[dev]->flags & DMA_DUPLEX) + audio_devs[dev]->audio_mode |= AM_READ; + else + audio_devs[dev]->audio_mode = AM_READ; + + while(c) + { + if ((buf_no = DMAbuf_getrdbuffer(dev, &dmabuf, &l, !!(file->f_flags & O_NONBLOCK))) < 0) + { + /* + * Nonblocking mode handling. Return current # of bytes + */ + + if (p > 0) /* Avoid throwing away data */ + return p; /* Return it instead */ + + if ((file->f_flags & O_NONBLOCK) && buf_no == -EAGAIN) + return -EAGAIN; + + return buf_no; + } + if (l > c) + l = c; + + /* + * Insert any local processing here. + */ + + if (audio_devs[dev]->local_conversion & CNV_MU_LAW) + { + translate_bytes(dsp_ulaw, (unsigned char *) dmabuf, l); + } + + { + char *fixit = dmabuf; + + if(copy_to_user(&(buf)[p], fixit, l)) + return -EFAULT; + }; + + DMAbuf_rmchars(dev, buf_no, l); + + p += l; + c -= l; + } + + return count - c; +} + +int audio_ioctl(int dev, struct file *file, unsigned int cmd, void __user *arg) +{ + int val, count; + unsigned long flags; + struct dma_buffparms *dmap; + int __user *p = arg; + + dev = dev >> 4; + + if (_IOC_TYPE(cmd) == 'C') { + if (audio_devs[dev]->coproc) /* Coprocessor ioctl */ + return audio_devs[dev]->coproc->ioctl(audio_devs[dev]->coproc->devc, cmd, arg, 0); + /* else + printk(KERN_DEBUG"/dev/dsp%d: No coprocessor for this device\n", dev); */ + return -ENXIO; + } + else switch (cmd) + { + case SNDCTL_DSP_SYNC: + if (!(audio_devs[dev]->open_mode & OPEN_WRITE)) + return 0; + if (audio_devs[dev]->dmap_out->fragment_size == 0) + return 0; + sync_output(dev); + DMAbuf_sync(dev); + DMAbuf_reset(dev); + return 0; + + case SNDCTL_DSP_POST: + if (!(audio_devs[dev]->open_mode & OPEN_WRITE)) + return 0; + if (audio_devs[dev]->dmap_out->fragment_size == 0) + return 0; + audio_devs[dev]->dmap_out->flags |= DMA_POST | DMA_DIRTY; + sync_output(dev); + dma_ioctl(dev, SNDCTL_DSP_POST, NULL); + return 0; + + case SNDCTL_DSP_RESET: + audio_devs[dev]->audio_mode = AM_NONE; + DMAbuf_reset(dev); + return 0; + + case SNDCTL_DSP_GETFMTS: + val = audio_devs[dev]->format_mask | AFMT_MU_LAW; + break; + + case SNDCTL_DSP_SETFMT: + if (get_user(val, p)) + return -EFAULT; + val = set_format(dev, val); + break; + + case SNDCTL_DSP_GETISPACE: + if (!(audio_devs[dev]->open_mode & OPEN_READ)) + return 0; + if ((audio_devs[dev]->audio_mode & AM_WRITE) && !(audio_devs[dev]->flags & DMA_DUPLEX)) + return -EBUSY; + return dma_ioctl(dev, cmd, arg); + + case SNDCTL_DSP_GETOSPACE: + if (!(audio_devs[dev]->open_mode & OPEN_WRITE)) + return -EPERM; + if ((audio_devs[dev]->audio_mode & AM_READ) && !(audio_devs[dev]->flags & DMA_DUPLEX)) + return -EBUSY; + return dma_ioctl(dev, cmd, arg); + + case SNDCTL_DSP_NONBLOCK: + spin_lock(&file->f_lock); + file->f_flags |= O_NONBLOCK; + spin_unlock(&file->f_lock); + return 0; + + case SNDCTL_DSP_GETCAPS: + val = 1 | DSP_CAP_MMAP; /* Revision level of this ioctl() */ + if (audio_devs[dev]->flags & DMA_DUPLEX && + audio_devs[dev]->open_mode == OPEN_READWRITE) + val |= DSP_CAP_DUPLEX; + if (audio_devs[dev]->coproc) + val |= DSP_CAP_COPROC; + if (audio_devs[dev]->d->local_qlen) /* Device has hidden buffers */ + val |= DSP_CAP_BATCH; + if (audio_devs[dev]->d->trigger) /* Supports SETTRIGGER */ + val |= DSP_CAP_TRIGGER; + break; + + case SOUND_PCM_WRITE_RATE: + if (get_user(val, p)) + return -EFAULT; + val = audio_devs[dev]->d->set_speed(dev, val); + break; + + case SOUND_PCM_READ_RATE: + val = audio_devs[dev]->d->set_speed(dev, 0); + break; + + case SNDCTL_DSP_STEREO: + if (get_user(val, p)) + return -EFAULT; + if (val > 1 || val < 0) + return -EINVAL; + val = audio_devs[dev]->d->set_channels(dev, val + 1) - 1; + break; + + case SOUND_PCM_WRITE_CHANNELS: + if (get_user(val, p)) + return -EFAULT; + val = audio_devs[dev]->d->set_channels(dev, val); + break; + + case SOUND_PCM_READ_CHANNELS: + val = audio_devs[dev]->d->set_channels(dev, 0); + break; + + case SOUND_PCM_READ_BITS: + val = audio_devs[dev]->d->set_bits(dev, 0); + break; + + case SNDCTL_DSP_SETDUPLEX: + if (audio_devs[dev]->open_mode != OPEN_READWRITE) + return -EPERM; + return (audio_devs[dev]->flags & DMA_DUPLEX) ? 0 : -EIO; + + case SNDCTL_DSP_PROFILE: + if (get_user(val, p)) + return -EFAULT; + if (audio_devs[dev]->open_mode & OPEN_WRITE) + audio_devs[dev]->dmap_out->applic_profile = val; + if (audio_devs[dev]->open_mode & OPEN_READ) + audio_devs[dev]->dmap_in->applic_profile = val; + return 0; + + case SNDCTL_DSP_GETODELAY: + dmap = audio_devs[dev]->dmap_out; + if (!(audio_devs[dev]->open_mode & OPEN_WRITE)) + return -EINVAL; + if (!(dmap->flags & DMA_ALLOC_DONE)) + { + val=0; + break; + } + + spin_lock_irqsave(&dmap->lock,flags); + /* Compute number of bytes that have been played */ + count = DMAbuf_get_buffer_pointer (dev, dmap, DMODE_OUTPUT); + if (count < dmap->fragment_size && dmap->qhead != 0) + count += dmap->bytes_in_use; /* Pointer wrap not handled yet */ + count += dmap->byte_counter; + + /* Subtract current count from the number of bytes written by app */ + count = dmap->user_counter - count; + if (count < 0) + count = 0; + spin_unlock_irqrestore(&dmap->lock,flags); + val = count; + break; + + default: + return dma_ioctl(dev, cmd, arg); + } + return put_user(val, p); +} + +void audio_init_devices(void) +{ + /* + * NOTE! This routine could be called several times during boot. + */ +} + +void reorganize_buffers(int dev, struct dma_buffparms *dmap, int recording) +{ + /* + * This routine breaks the physical device buffers to logical ones. + */ + + struct audio_operations *dsp_dev = audio_devs[dev]; + + unsigned i, n; + unsigned sr, nc, sz, bsz; + + sr = dsp_dev->d->set_speed(dev, 0); + nc = dsp_dev->d->set_channels(dev, 0); + sz = dsp_dev->d->set_bits(dev, 0); + + if (sz == 8) + dmap->neutral_byte = NEUTRAL8; + else + dmap->neutral_byte = NEUTRAL16; + + if (sr < 1 || nc < 1 || sz < 1) + { +/* printk(KERN_DEBUG "Warning: Invalid PCM parameters[%d] sr=%d, nc=%d, sz=%d\n", dev, sr, nc, sz);*/ + sr = DSP_DEFAULT_SPEED; + nc = 1; + sz = 8; + } + + sz = sr * nc * sz; + + sz /= 8; /* #bits -> #bytes */ + dmap->data_rate = sz; + + if (!dmap->needs_reorg) + return; + dmap->needs_reorg = 0; + + if (dmap->fragment_size == 0) + { + /* Compute the fragment size using the default algorithm */ + + /* + * Compute a buffer size for time not exceeding 1 second. + * Usually this algorithm gives a buffer size for 0.5 to 1.0 seconds + * of sound (using the current speed, sample size and #channels). + */ + + bsz = dmap->buffsize; + while (bsz > sz) + bsz /= 2; + + if (bsz == dmap->buffsize) + bsz /= 2; /* Needs at least 2 buffers */ + + /* + * Split the computed fragment to smaller parts. After 3.5a9 + * the default subdivision is 4 which should give better + * results when recording. + */ + + if (dmap->subdivision == 0) /* Not already set */ + { + dmap->subdivision = 4; /* Init to the default value */ + + if ((bsz / dmap->subdivision) > 4096) + dmap->subdivision *= 2; + if ((bsz / dmap->subdivision) < 4096) + dmap->subdivision = 1; + } + bsz /= dmap->subdivision; + + if (bsz < 16) + bsz = 16; /* Just a sanity check */ + + dmap->fragment_size = bsz; + } + else + { + /* + * The process has specified the buffer size with SNDCTL_DSP_SETFRAGMENT or + * the buffer size computation has already been done. + */ + if (dmap->fragment_size > (dmap->buffsize / 2)) + dmap->fragment_size = (dmap->buffsize / 2); + bsz = dmap->fragment_size; + } + + if (audio_devs[dev]->min_fragment) + if (bsz < (1 << audio_devs[dev]->min_fragment)) + bsz = 1 << audio_devs[dev]->min_fragment; + if (audio_devs[dev]->max_fragment) + if (bsz > (1 << audio_devs[dev]->max_fragment)) + bsz = 1 << audio_devs[dev]->max_fragment; + bsz &= ~0x07; /* Force size which is multiple of 8 bytes */ +#ifdef OS_DMA_ALIGN_CHECK + OS_DMA_ALIGN_CHECK(bsz); +#endif + + n = dmap->buffsize / bsz; + if (n > MAX_SUB_BUFFERS) + n = MAX_SUB_BUFFERS; + if (n > dmap->max_fragments) + n = dmap->max_fragments; + + if (n < 2) + { + n = 2; + bsz /= 2; + } + dmap->nbufs = n; + dmap->bytes_in_use = n * bsz; + dmap->fragment_size = bsz; + dmap->max_byte_counter = (dmap->data_rate * 60 * 60) + + dmap->bytes_in_use; /* Approximately one hour */ + + if (dmap->raw_buf) + { + memset(dmap->raw_buf, dmap->neutral_byte, dmap->bytes_in_use); + } + + for (i = 0; i < dmap->nbufs; i++) + { + dmap->counts[i] = 0; + } + + dmap->flags |= DMA_ALLOC_DONE | DMA_EMPTY; +} + +static int dma_subdivide(int dev, struct dma_buffparms *dmap, int fact) +{ + if (fact == 0) + { + fact = dmap->subdivision; + if (fact == 0) + fact = 1; + return fact; + } + if (dmap->subdivision != 0 || dmap->fragment_size) /* Too late to change */ + return -EINVAL; + + if (fact > MAX_REALTIME_FACTOR) + return -EINVAL; + + if (fact != 1 && fact != 2 && fact != 4 && fact != 8 && fact != 16) + return -EINVAL; + + dmap->subdivision = fact; + return fact; +} + +static int dma_set_fragment(int dev, struct dma_buffparms *dmap, int fact) +{ + int bytes, count; + + if (fact == 0) + return -EIO; + + if (dmap->subdivision != 0 || + dmap->fragment_size) /* Too late to change */ + return -EINVAL; + + bytes = fact & 0xffff; + count = (fact >> 16) & 0x7fff; + + if (count == 0) + count = MAX_SUB_BUFFERS; + else if (count < MAX_SUB_BUFFERS) + count++; + + if (bytes < 4 || bytes > 17) /* <16 || > 512k */ + return -EINVAL; + + if (count < 2) + return -EINVAL; + + if (audio_devs[dev]->min_fragment > 0) + if (bytes < audio_devs[dev]->min_fragment) + bytes = audio_devs[dev]->min_fragment; + + if (audio_devs[dev]->max_fragment > 0) + if (bytes > audio_devs[dev]->max_fragment) + bytes = audio_devs[dev]->max_fragment; + +#ifdef OS_DMA_MINBITS + if (bytes < OS_DMA_MINBITS) + bytes = OS_DMA_MINBITS; +#endif + + dmap->fragment_size = (1 << bytes); + dmap->max_fragments = count; + + if (dmap->fragment_size > dmap->buffsize) + dmap->fragment_size = dmap->buffsize; + + if (dmap->fragment_size == dmap->buffsize && + audio_devs[dev]->flags & DMA_AUTOMODE) + dmap->fragment_size /= 2; /* Needs at least 2 buffers */ + + dmap->subdivision = 1; /* Disable SNDCTL_DSP_SUBDIVIDE */ + return bytes | ((count - 1) << 16); +} + +static int dma_ioctl(int dev, unsigned int cmd, void __user *arg) +{ + struct dma_buffparms *dmap_out = audio_devs[dev]->dmap_out; + struct dma_buffparms *dmap_in = audio_devs[dev]->dmap_in; + struct dma_buffparms *dmap; + audio_buf_info info; + count_info cinfo; + int fact, ret, changed, bits, count, err; + unsigned long flags; + + switch (cmd) + { + case SNDCTL_DSP_SUBDIVIDE: + ret = 0; + if (get_user(fact, (int __user *)arg)) + return -EFAULT; + if (audio_devs[dev]->open_mode & OPEN_WRITE) + ret = dma_subdivide(dev, dmap_out, fact); + if (ret < 0) + return ret; + if (audio_devs[dev]->open_mode != OPEN_WRITE || + (audio_devs[dev]->flags & DMA_DUPLEX && + audio_devs[dev]->open_mode & OPEN_READ)) + ret = dma_subdivide(dev, dmap_in, fact); + if (ret < 0) + return ret; + break; + + case SNDCTL_DSP_GETISPACE: + case SNDCTL_DSP_GETOSPACE: + dmap = dmap_out; + if (cmd == SNDCTL_DSP_GETISPACE && !(audio_devs[dev]->open_mode & OPEN_READ)) + return -EINVAL; + if (cmd == SNDCTL_DSP_GETOSPACE && !(audio_devs[dev]->open_mode & OPEN_WRITE)) + return -EINVAL; + if (cmd == SNDCTL_DSP_GETISPACE && audio_devs[dev]->flags & DMA_DUPLEX) + dmap = dmap_in; + if (dmap->mapping_flags & DMA_MAP_MAPPED) + return -EINVAL; + if (!(dmap->flags & DMA_ALLOC_DONE)) + reorganize_buffers(dev, dmap, (cmd == SNDCTL_DSP_GETISPACE)); + info.fragstotal = dmap->nbufs; + if (cmd == SNDCTL_DSP_GETISPACE) + info.fragments = dmap->qlen; + else + { + if (!DMAbuf_space_in_queue(dev)) + info.fragments = 0; + else + { + info.fragments = DMAbuf_space_in_queue(dev); + if (audio_devs[dev]->d->local_qlen) + { + int tmp = audio_devs[dev]->d->local_qlen(dev); + if (tmp && info.fragments) + tmp--; /* + * This buffer has been counted twice + */ + info.fragments -= tmp; + } + } + } + if (info.fragments < 0) + info.fragments = 0; + else if (info.fragments > dmap->nbufs) + info.fragments = dmap->nbufs; + + info.fragsize = dmap->fragment_size; + info.bytes = info.fragments * dmap->fragment_size; + + if (cmd == SNDCTL_DSP_GETISPACE && dmap->qlen) + info.bytes -= dmap->counts[dmap->qhead]; + else + { + info.fragments = info.bytes / dmap->fragment_size; + info.bytes -= dmap->user_counter % dmap->fragment_size; + } + if (copy_to_user(arg, &info, sizeof(info))) + return -EFAULT; + return 0; + + case SNDCTL_DSP_SETTRIGGER: + if (get_user(bits, (int __user *)arg)) + return -EFAULT; + bits &= audio_devs[dev]->open_mode; + if (audio_devs[dev]->d->trigger == NULL) + return -EINVAL; + if (!(audio_devs[dev]->flags & DMA_DUPLEX) && (bits & PCM_ENABLE_INPUT) && + (bits & PCM_ENABLE_OUTPUT)) + return -EINVAL; + + if (bits & PCM_ENABLE_INPUT) + { + spin_lock_irqsave(&dmap_in->lock,flags); + changed = (audio_devs[dev]->enable_bits ^ bits) & PCM_ENABLE_INPUT; + if (changed && audio_devs[dev]->go) + { + reorganize_buffers(dev, dmap_in, 1); + if ((err = audio_devs[dev]->d->prepare_for_input(dev, + dmap_in->fragment_size, dmap_in->nbufs)) < 0) { + spin_unlock_irqrestore(&dmap_in->lock,flags); + return err; + } + dmap_in->dma_mode = DMODE_INPUT; + audio_devs[dev]->enable_bits |= PCM_ENABLE_INPUT; + DMAbuf_activate_recording(dev, dmap_in); + } else + audio_devs[dev]->enable_bits &= ~PCM_ENABLE_INPUT; + spin_unlock_irqrestore(&dmap_in->lock,flags); + } + if (bits & PCM_ENABLE_OUTPUT) + { + spin_lock_irqsave(&dmap_out->lock,flags); + changed = (audio_devs[dev]->enable_bits ^ bits) & PCM_ENABLE_OUTPUT; + if (changed && + (dmap_out->mapping_flags & DMA_MAP_MAPPED || dmap_out->qlen > 0) && + audio_devs[dev]->go) + { + if (!(dmap_out->flags & DMA_ALLOC_DONE)) + reorganize_buffers(dev, dmap_out, 0); + dmap_out->dma_mode = DMODE_OUTPUT; + audio_devs[dev]->enable_bits |= PCM_ENABLE_OUTPUT; + dmap_out->counts[dmap_out->qhead] = dmap_out->fragment_size; + DMAbuf_launch_output(dev, dmap_out); + } else + audio_devs[dev]->enable_bits &= ~PCM_ENABLE_OUTPUT; + spin_unlock_irqrestore(&dmap_out->lock,flags); + } +#if 0 + if (changed && audio_devs[dev]->d->trigger) + audio_devs[dev]->d->trigger(dev, bits * audio_devs[dev]->go); +#endif + /* Falls through... */ + + case SNDCTL_DSP_GETTRIGGER: + ret = audio_devs[dev]->enable_bits; + break; + + case SNDCTL_DSP_SETSYNCRO: + if (!audio_devs[dev]->d->trigger) + return -EINVAL; + audio_devs[dev]->d->trigger(dev, 0); + audio_devs[dev]->go = 0; + return 0; + + case SNDCTL_DSP_GETIPTR: + if (!(audio_devs[dev]->open_mode & OPEN_READ)) + return -EINVAL; + spin_lock_irqsave(&dmap_in->lock,flags); + cinfo.bytes = dmap_in->byte_counter; + cinfo.ptr = DMAbuf_get_buffer_pointer(dev, dmap_in, DMODE_INPUT) & ~3; + if (cinfo.ptr < dmap_in->fragment_size && dmap_in->qtail != 0) + cinfo.bytes += dmap_in->bytes_in_use; /* Pointer wrap not handled yet */ + cinfo.blocks = dmap_in->qlen; + cinfo.bytes += cinfo.ptr; + if (dmap_in->mapping_flags & DMA_MAP_MAPPED) + dmap_in->qlen = 0; /* Reset interrupt counter */ + spin_unlock_irqrestore(&dmap_in->lock,flags); + if (copy_to_user(arg, &cinfo, sizeof(cinfo))) + return -EFAULT; + return 0; + + case SNDCTL_DSP_GETOPTR: + if (!(audio_devs[dev]->open_mode & OPEN_WRITE)) + return -EINVAL; + + spin_lock_irqsave(&dmap_out->lock,flags); + cinfo.bytes = dmap_out->byte_counter; + cinfo.ptr = DMAbuf_get_buffer_pointer(dev, dmap_out, DMODE_OUTPUT) & ~3; + if (cinfo.ptr < dmap_out->fragment_size && dmap_out->qhead != 0) + cinfo.bytes += dmap_out->bytes_in_use; /* Pointer wrap not handled yet */ + cinfo.blocks = dmap_out->qlen; + cinfo.bytes += cinfo.ptr; + if (dmap_out->mapping_flags & DMA_MAP_MAPPED) + dmap_out->qlen = 0; /* Reset interrupt counter */ + spin_unlock_irqrestore(&dmap_out->lock,flags); + if (copy_to_user(arg, &cinfo, sizeof(cinfo))) + return -EFAULT; + return 0; + + case SNDCTL_DSP_GETODELAY: + if (!(audio_devs[dev]->open_mode & OPEN_WRITE)) + return -EINVAL; + if (!(dmap_out->flags & DMA_ALLOC_DONE)) + { + ret=0; + break; + } + spin_lock_irqsave(&dmap_out->lock,flags); + /* Compute number of bytes that have been played */ + count = DMAbuf_get_buffer_pointer (dev, dmap_out, DMODE_OUTPUT); + if (count < dmap_out->fragment_size && dmap_out->qhead != 0) + count += dmap_out->bytes_in_use; /* Pointer wrap not handled yet */ + count += dmap_out->byte_counter; + /* Subtract current count from the number of bytes written by app */ + count = dmap_out->user_counter - count; + if (count < 0) + count = 0; + spin_unlock_irqrestore(&dmap_out->lock,flags); + ret = count; + break; + + case SNDCTL_DSP_POST: + if (audio_devs[dev]->dmap_out->qlen > 0) + if (!(audio_devs[dev]->dmap_out->flags & DMA_ACTIVE)) + DMAbuf_launch_output(dev, audio_devs[dev]->dmap_out); + return 0; + + case SNDCTL_DSP_GETBLKSIZE: + dmap = dmap_out; + if (audio_devs[dev]->open_mode & OPEN_WRITE) + reorganize_buffers(dev, dmap_out, (audio_devs[dev]->open_mode == OPEN_READ)); + if (audio_devs[dev]->open_mode == OPEN_READ || + (audio_devs[dev]->flags & DMA_DUPLEX && + audio_devs[dev]->open_mode & OPEN_READ)) + reorganize_buffers(dev, dmap_in, (audio_devs[dev]->open_mode == OPEN_READ)); + if (audio_devs[dev]->open_mode == OPEN_READ) + dmap = dmap_in; + ret = dmap->fragment_size; + break; + + case SNDCTL_DSP_SETFRAGMENT: + ret = 0; + if (get_user(fact, (int __user *)arg)) + return -EFAULT; + if (audio_devs[dev]->open_mode & OPEN_WRITE) + ret = dma_set_fragment(dev, dmap_out, fact); + if (ret < 0) + return ret; + if (audio_devs[dev]->open_mode == OPEN_READ || + (audio_devs[dev]->flags & DMA_DUPLEX && + audio_devs[dev]->open_mode & OPEN_READ)) + ret = dma_set_fragment(dev, dmap_in, fact); + if (ret < 0) + return ret; + if (!arg) /* don't know what this is good for, but preserve old semantics */ + return 0; + break; + + default: + if (!audio_devs[dev]->d->ioctl) + return -EINVAL; + return audio_devs[dev]->d->ioctl(dev, cmd, arg); + } + return put_user(ret, (int __user *)arg); +} diff --git a/sound/oss/bin2hex.c b/sound/oss/bin2hex.c new file mode 100644 index 00000000..b59109eb --- /dev/null +++ b/sound/oss/bin2hex.c @@ -0,0 +1,39 @@ +#include <stdio.h> +#include <string.h> +#include <stdlib.h> + +int main( int argc, const char * argv [] ) +{ + const char * varname; + int i = 0; + int c; + int id = 0; + + if(argv[1] && strcmp(argv[1],"-i")==0) + { + argv++; + argc--; + id=1; + } + + if(argc==1) + { + fprintf(stderr, "bin2hex: [-i] firmware\n"); + exit(1); + } + + varname = argv[1]; + printf( "/* automatically generated by bin2hex */\n" ); + printf( "static unsigned char %s [] %s =\n{\n", varname , id?"__initdata":""); + + while ( ( c = getchar( ) ) != EOF ) + { + if ( i != 0 && i % 10 == 0 ) + printf( "\n" ); + printf( "0x%02lx,", c & 0xFFl ); + i++; + } + + printf( "};\nstatic int %sLen = %d;\n", varname, i ); + return 0; +} diff --git a/sound/oss/coproc.h b/sound/oss/coproc.h new file mode 100644 index 00000000..7bec21bb --- /dev/null +++ b/sound/oss/coproc.h @@ -0,0 +1,12 @@ +/* + * Definitions for various on board processors on the sound cards. For + * example DSP processors. + */ + +/* + * Coprocessor access types + */ +#define COPR_CUSTOM 0x0001 /* Custom applications */ +#define COPR_MIDI 0x0002 /* MIDI (MPU-401) emulation */ +#define COPR_PCM 0x0004 /* Digitized voice applications */ +#define COPR_SYNTH 0x0008 /* Music synthesis */ diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c new file mode 100644 index 00000000..d8cf3e58 --- /dev/null +++ b/sound/oss/dev_table.c @@ -0,0 +1,256 @@ +/* + * sound/oss/dev_table.c + * + * Device call tables. + * + * + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + */ + +#include <linux/init.h> + +#include "sound_config.h" + +struct audio_operations *audio_devs[MAX_AUDIO_DEV]; +EXPORT_SYMBOL(audio_devs); + +int num_audiodevs; +EXPORT_SYMBOL(num_audiodevs); + +struct mixer_operations *mixer_devs[MAX_MIXER_DEV]; +EXPORT_SYMBOL(mixer_devs); + +int num_mixers; +EXPORT_SYMBOL(num_mixers); + +struct synth_operations *synth_devs[MAX_SYNTH_DEV+MAX_MIDI_DEV]; +EXPORT_SYMBOL(synth_devs); + +int num_synths; + +struct midi_operations *midi_devs[MAX_MIDI_DEV]; +EXPORT_SYMBOL(midi_devs); + +int num_midis; +EXPORT_SYMBOL(num_midis); + +struct sound_timer_operations *sound_timer_devs[MAX_TIMER_DEV] = { + &default_sound_timer, NULL +}; +EXPORT_SYMBOL(sound_timer_devs); + +int num_sound_timers = 1; + + +static int sound_alloc_audiodev(void); + +int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver, + int driver_size, int flags, unsigned int format_mask, + void *devc, int dma1, int dma2) +{ + struct audio_driver *d; + struct audio_operations *op; + int num; + + if (vers != AUDIO_DRIVER_VERSION || driver_size > sizeof(struct audio_driver)) { + printk(KERN_ERR "Sound: Incompatible audio driver for %s\n", name); + return -(EINVAL); + } + num = sound_alloc_audiodev(); + + if (num == -1) { + printk(KERN_ERR "sound: Too many audio drivers\n"); + return -(EBUSY); + } + d = (struct audio_driver *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_driver))); + sound_nblocks++; + if (sound_nblocks >= MAX_MEM_BLOCKS) + sound_nblocks = MAX_MEM_BLOCKS - 1; + + op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vzalloc(sizeof(struct audio_operations))); + sound_nblocks++; + if (sound_nblocks >= MAX_MEM_BLOCKS) + sound_nblocks = MAX_MEM_BLOCKS - 1; + + if (d == NULL || op == NULL) { + printk(KERN_ERR "Sound: Can't allocate driver for (%s)\n", name); + sound_unload_audiodev(num); + return -(ENOMEM); + } + init_waitqueue_head(&op->in_sleeper); + init_waitqueue_head(&op->out_sleeper); + init_waitqueue_head(&op->poll_sleeper); + if (driver_size < sizeof(struct audio_driver)) + memset((char *) d, 0, sizeof(struct audio_driver)); + + memcpy((char *) d, (char *) driver, driver_size); + + op->d = d; + strlcpy(op->name, name, sizeof(op->name)); + op->flags = flags; + op->format_mask = format_mask; + op->devc = devc; + + /* + * Hardcoded defaults + */ + audio_devs[num] = op; + + DMAbuf_init(num, dma1, dma2); + + audio_init_devices(); + return num; +} +EXPORT_SYMBOL(sound_install_audiodrv); + +int sound_install_mixer(int vers, char *name, struct mixer_operations *driver, + int driver_size, void *devc) +{ + struct mixer_operations *op; + + int n = sound_alloc_mixerdev(); + + if (n == -1) { + printk(KERN_ERR "Sound: Too many mixer drivers\n"); + return -EBUSY; + } + if (vers != MIXER_DRIVER_VERSION || + driver_size > sizeof(struct mixer_operations)) { + printk(KERN_ERR "Sound: Incompatible mixer driver for %s\n", name); + return -EINVAL; + } + + /* FIXME: This leaks a mixer_operations struct every time its called + until you unload sound! */ + + op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vzalloc(sizeof(struct mixer_operations))); + sound_nblocks++; + if (sound_nblocks >= MAX_MEM_BLOCKS) + sound_nblocks = MAX_MEM_BLOCKS - 1; + + if (op == NULL) { + printk(KERN_ERR "Sound: Can't allocate mixer driver for (%s)\n", name); + return -ENOMEM; + } + memcpy((char *) op, (char *) driver, driver_size); + + strlcpy(op->name, name, sizeof(op->name)); + op->devc = devc; + + mixer_devs[n] = op; + return n; +} +EXPORT_SYMBOL(sound_install_mixer); + +void sound_unload_audiodev(int dev) +{ + if (dev != -1) { + DMAbuf_deinit(dev); + audio_devs[dev] = NULL; + unregister_sound_dsp((dev<<4)+3); + } +} +EXPORT_SYMBOL(sound_unload_audiodev); + +static int sound_alloc_audiodev(void) +{ + int i = register_sound_dsp(&oss_sound_fops, -1); + if(i==-1) + return i; + i>>=4; + if(i>=num_audiodevs) + num_audiodevs = i + 1; + return i; +} + +int sound_alloc_mididev(void) +{ + int i = register_sound_midi(&oss_sound_fops, -1); + if(i==-1) + return i; + i>>=4; + if(i>=num_midis) + num_midis = i + 1; + return i; +} +EXPORT_SYMBOL(sound_alloc_mididev); + +int sound_alloc_synthdev(void) +{ + int i; + + for (i = 0; i < MAX_SYNTH_DEV; i++) { + if (synth_devs[i] == NULL) { + if (i >= num_synths) + num_synths++; + return i; + } + } + return -1; +} +EXPORT_SYMBOL(sound_alloc_synthdev); + +int sound_alloc_mixerdev(void) +{ + int i = register_sound_mixer(&oss_sound_fops, -1); + if(i==-1) + return -1; + i>>=4; + if(i>=num_mixers) + num_mixers = i + 1; + return i; +} +EXPORT_SYMBOL(sound_alloc_mixerdev); + +int sound_alloc_timerdev(void) +{ + int i; + + for (i = 0; i < MAX_TIMER_DEV; i++) { + if (sound_timer_devs[i] == NULL) { + if (i >= num_sound_timers) + num_sound_timers++; + return i; + } + } + return -1; +} +EXPORT_SYMBOL(sound_alloc_timerdev); + +void sound_unload_mixerdev(int dev) +{ + if (dev != -1) { + mixer_devs[dev] = NULL; + unregister_sound_mixer(dev<<4); + num_mixers--; + } +} +EXPORT_SYMBOL(sound_unload_mixerdev); + +void sound_unload_mididev(int dev) +{ + if (dev != -1) { + midi_devs[dev] = NULL; + unregister_sound_midi((dev<<4)+2); + } +} +EXPORT_SYMBOL(sound_unload_mididev); + +void sound_unload_synthdev(int dev) +{ + if (dev != -1) + synth_devs[dev] = NULL; +} +EXPORT_SYMBOL(sound_unload_synthdev); + +void sound_unload_timerdev(int dev) +{ + if (dev != -1) + sound_timer_devs[dev] = NULL; +} +EXPORT_SYMBOL(sound_unload_timerdev); + diff --git a/sound/oss/dev_table.h b/sound/oss/dev_table.h new file mode 100644 index 00000000..0199a317 --- /dev/null +++ b/sound/oss/dev_table.h @@ -0,0 +1,390 @@ +/* + * dev_table.h + * + * Global definitions for device call tables + * + * + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + */ + + +#ifndef _DEV_TABLE_H_ +#define _DEV_TABLE_H_ + +#include <linux/spinlock.h> +/* + * Sound card numbers 27 to 999. (1 to 26 are defined in soundcard.h) + * Numbers 1000 to N are reserved for driver's internal use. + */ + +#define SNDCARD_DESKPROXL 27 /* Compaq Deskpro XL */ +#define SNDCARD_VIDC 28 /* ARMs VIDC */ +#define SNDCARD_SBPNP 29 +#define SNDCARD_SOFTOSS 36 +#define SNDCARD_VMIDI 37 +#define SNDCARD_OPL3SA1 38 /* Note: clash in msnd.h */ +#define SNDCARD_OPL3SA1_SB 39 +#define SNDCARD_OPL3SA1_MPU 40 +#define SNDCARD_WAVEFRONT 41 +#define SNDCARD_OPL3SA2 42 +#define SNDCARD_OPL3SA2_MPU 43 +#define SNDCARD_WAVEARTIST 44 /* Waveartist */ +#define SNDCARD_OPL3SA2_MSS 45 /* Originally missed */ +#define SNDCARD_AD1816 88 + +/* + * NOTE! NOTE! NOTE! NOTE! + * + * If you modify this file, please check the dev_table.c also. + * + * NOTE! NOTE! NOTE! NOTE! + */ + +struct driver_info +{ + char *driver_id; + int card_subtype; /* Driver specific. Usually 0 */ + int card_type; /* From soundcard.h */ + char *name; + void (*attach) (struct address_info *hw_config); + int (*probe) (struct address_info *hw_config); + void (*unload) (struct address_info *hw_config); +}; + +struct card_info +{ + int card_type; /* Link (search key) to the driver list */ + struct address_info config; + int enabled; + void *for_driver_use; +}; + + +/* + * Device specific parameters (used only by dmabuf.c) + */ +#define MAX_SUB_BUFFERS (32*MAX_REALTIME_FACTOR) + +#define DMODE_NONE 0 +#define DMODE_OUTPUT PCM_ENABLE_OUTPUT +#define DMODE_INPUT PCM_ENABLE_INPUT + +struct dma_buffparms +{ + int dma_mode; /* DMODE_INPUT, DMODE_OUTPUT or DMODE_NONE */ + int closing; + + /* + * Pointers to raw buffers + */ + + char *raw_buf; + unsigned long raw_buf_phys; + int buffsize; + + /* + * Device state tables + */ + + unsigned long flags; +#define DMA_BUSY 0x00000001 +#define DMA_RESTART 0x00000002 +#define DMA_ACTIVE 0x00000004 +#define DMA_STARTED 0x00000008 +#define DMA_EMPTY 0x00000010 +#define DMA_ALLOC_DONE 0x00000020 +#define DMA_SYNCING 0x00000040 +#define DMA_DIRTY 0x00000080 +#define DMA_POST 0x00000100 +#define DMA_NODMA 0x00000200 +#define DMA_NOTIMEOUT 0x00000400 + + int open_mode; + + /* + * Queue parameters. + */ + int qlen; + int qhead; + int qtail; + spinlock_t lock; + + int cfrag; /* Current incomplete fragment (write) */ + + int nbufs; + int counts[MAX_SUB_BUFFERS]; + int subdivision; + + int fragment_size; + int needs_reorg; + int max_fragments; + + int bytes_in_use; + + int underrun_count; + unsigned long byte_counter; + unsigned long user_counter; + unsigned long max_byte_counter; + int data_rate; /* Bytes/second */ + + int mapping_flags; +#define DMA_MAP_MAPPED 0x00000001 + char neutral_byte; + int dma; /* DMA channel */ + + int applic_profile; /* Application profile (APF_*) */ + /* Interrupt callback stuff */ + void (*audio_callback) (int dev, int parm); + int callback_parm; + + int buf_flags[MAX_SUB_BUFFERS]; +#define BUFF_EOF 0x00000001 /* Increment eof count */ +#define BUFF_DIRTY 0x00000002 /* Buffer written */ +}; + +/* + * Structure for use with various microcontrollers and DSP processors + * in the recent sound cards. + */ +typedef struct coproc_operations +{ + char name[64]; + struct module *owner; + int (*open) (void *devc, int sub_device); + void (*close) (void *devc, int sub_device); + int (*ioctl) (void *devc, unsigned int cmd, void __user * arg, int local); + void (*reset) (void *devc); + + void *devc; /* Driver specific info */ +} coproc_operations; + +struct audio_driver +{ + struct module *owner; + int (*open) (int dev, int mode); + void (*close) (int dev); + void (*output_block) (int dev, unsigned long buf, + int count, int intrflag); + void (*start_input) (int dev, unsigned long buf, + int count, int intrflag); + int (*ioctl) (int dev, unsigned int cmd, void __user * arg); + int (*prepare_for_input) (int dev, int bufsize, int nbufs); + int (*prepare_for_output) (int dev, int bufsize, int nbufs); + void (*halt_io) (int dev); + int (*local_qlen)(int dev); + void (*copy_user) (int dev, + char *localbuf, int localoffs, + const char __user *userbuf, int useroffs, + int max_in, int max_out, + int *used, int *returned, + int len); + void (*halt_input) (int dev); + void (*halt_output) (int dev); + void (*trigger) (int dev, int bits); + int (*set_speed)(int dev, int speed); + unsigned int (*set_bits)(int dev, unsigned int bits); + short (*set_channels)(int dev, short channels); + void (*postprocess_write)(int dev); /* Device spesific postprocessing for written data */ + void (*preprocess_read)(int dev); /* Device spesific preprocessing for read data */ + void (*mmap)(int dev); +}; + +struct audio_operations +{ + char name[128]; + int flags; +#define NOTHING_SPECIAL 0x00 +#define NEEDS_RESTART 0x01 +#define DMA_AUTOMODE 0x02 +#define DMA_DUPLEX 0x04 +#define DMA_PSEUDO_AUTOMODE 0x08 +#define DMA_HARDSTOP 0x10 +#define DMA_EXACT 0x40 +#define DMA_NORESET 0x80 + int format_mask; /* Bitmask for supported audio formats */ + void *devc; /* Driver specific info */ + struct audio_driver *d; + void *portc; /* Driver specific info */ + struct dma_buffparms *dmap_in, *dmap_out; + struct coproc_operations *coproc; + int mixer_dev; + int enable_bits; + int open_mode; + int go; + int min_fragment; /* 0 == unlimited */ + int max_fragment; /* 0 == unlimited */ + int parent_dev; /* 0 -> no parent, 1 to n -> parent=parent_dev+1 */ + + /* fields formerly in dmabuf.c */ + wait_queue_head_t in_sleeper; + wait_queue_head_t out_sleeper; + wait_queue_head_t poll_sleeper; + + /* fields formerly in audio.c */ + int audio_mode; + +#define AM_NONE 0 +#define AM_WRITE OPEN_WRITE +#define AM_READ OPEN_READ + + int local_format; + int audio_format; + int local_conversion; +#define CNV_MU_LAW 0x00000001 + + /* large structures at the end to keep offsets small */ + struct dma_buffparms dmaps[2]; +}; + +int *load_mixer_volumes(char *name, int *levels, int present); + +struct mixer_operations +{ + struct module *owner; + char id[16]; + char name[64]; + int (*ioctl) (int dev, unsigned int cmd, void __user * arg); + + void *devc; + int modify_counter; +}; + +struct synth_operations +{ + struct module *owner; + char *id; /* Unique identifier (ASCII) max 29 char */ + struct synth_info *info; + int midi_dev; + int synth_type; + int synth_subtype; + + int (*open) (int dev, int mode); + void (*close) (int dev); + int (*ioctl) (int dev, unsigned int cmd, void __user * arg); + int (*kill_note) (int dev, int voice, int note, int velocity); + int (*start_note) (int dev, int voice, int note, int velocity); + int (*set_instr) (int dev, int voice, int instr); + void (*reset) (int dev); + void (*hw_control) (int dev, unsigned char *event); + int (*load_patch) (int dev, int format, const char __user *addr, + int count, int pmgr_flag); + void (*aftertouch) (int dev, int voice, int pressure); + void (*controller) (int dev, int voice, int ctrl_num, int value); + void (*panning) (int dev, int voice, int value); + void (*volume_method) (int dev, int mode); + void (*bender) (int dev, int chn, int value); + int (*alloc_voice) (int dev, int chn, int note, struct voice_alloc_info *alloc); + void (*setup_voice) (int dev, int voice, int chn); + int (*send_sysex)(int dev, unsigned char *bytes, int len); + + struct voice_alloc_info alloc; + struct channel_info chn_info[16]; + int emulation; +#define EMU_GM 1 /* General MIDI */ +#define EMU_XG 2 /* Yamaha XG */ +#define MAX_SYSEX_BUF 64 + unsigned char sysex_buf[MAX_SYSEX_BUF]; + int sysex_ptr; +}; + +struct midi_input_info +{ + /* MIDI input scanner variables */ +#define MI_MAX 10 + volatile int m_busy; + unsigned char m_buf[MI_MAX]; + unsigned char m_prev_status; /* For running status */ + int m_ptr; +#define MST_INIT 0 +#define MST_DATA 1 +#define MST_SYSEX 2 + int m_state; + int m_left; +}; + +struct midi_operations +{ + struct module *owner; + struct midi_info info; + struct synth_operations *converter; + struct midi_input_info in_info; + int (*open) (int dev, int mode, + void (*inputintr)(int dev, unsigned char data), + void (*outputintr)(int dev) + ); + void (*close) (int dev); + int (*ioctl) (int dev, unsigned int cmd, void __user * arg); + int (*outputc) (int dev, unsigned char data); + int (*start_read) (int dev); + int (*end_read) (int dev); + void (*kick)(int dev); + int (*command) (int dev, unsigned char *data); + int (*buffer_status) (int dev); + int (*prefix_cmd) (int dev, unsigned char status); + struct coproc_operations *coproc; + void *devc; +}; + +struct sound_lowlev_timer +{ + int dev; + int priority; + unsigned int (*tmr_start)(int dev, unsigned int usecs); + void (*tmr_disable)(int dev); + void (*tmr_restart)(int dev); +}; + +struct sound_timer_operations +{ + struct module *owner; + struct sound_timer_info info; + int priority; + int devlink; + int (*open)(int dev, int mode); + void (*close)(int dev); + int (*event)(int dev, unsigned char *ev); + unsigned long (*get_time)(int dev); + int (*ioctl) (int dev, unsigned int cmd, void __user * arg); + void (*arm_timer)(int dev, long time); +}; + +extern struct sound_timer_operations default_sound_timer; + +extern struct audio_operations *audio_devs[MAX_AUDIO_DEV]; +extern int num_audiodevs; +extern struct mixer_operations *mixer_devs[MAX_MIXER_DEV]; +extern int num_mixers; +extern struct synth_operations *synth_devs[MAX_SYNTH_DEV+MAX_MIDI_DEV]; +extern int num_synths; +extern struct midi_operations *midi_devs[MAX_MIDI_DEV]; +extern int num_midis; +extern struct sound_timer_operations * sound_timer_devs[MAX_TIMER_DEV]; +extern int num_sound_timers; + +extern int sound_map_buffer (int dev, struct dma_buffparms *dmap, buffmem_desc *info); +void sound_timer_init (struct sound_lowlev_timer *t, char *name); +void sound_dma_intr (int dev, struct dma_buffparms *dmap, int chan); + +#define AUDIO_DRIVER_VERSION 2 +#define MIXER_DRIVER_VERSION 2 +int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver, + int driver_size, int flags, unsigned int format_mask, + void *devc, int dma1, int dma2); +int sound_install_mixer(int vers, char *name, struct mixer_operations *driver, + int driver_size, void *devc); + +void sound_unload_audiodev(int dev); +void sound_unload_mixerdev(int dev); +void sound_unload_mididev(int dev); +void sound_unload_synthdev(int dev); +void sound_unload_timerdev(int dev); +int sound_alloc_mixerdev(void); +int sound_alloc_timerdev(void); +int sound_alloc_synthdev(void); +int sound_alloc_mididev(void); +#endif /* _DEV_TABLE_H_ */ + diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c new file mode 100644 index 00000000..bcc3e8e0 --- /dev/null +++ b/sound/oss/dmabuf.c @@ -0,0 +1,1268 @@ +/* + * sound/oss/dmabuf.c + * + * The DMA buffer manager for digitized voice applications + */ +/* + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * + * Thomas Sailer : moved several static variables into struct audio_operations + * (which is grossly misnamed btw.) because they have the same + * lifetime as the rest in there and dynamic allocation saves + * 12k or so + * Thomas Sailer : remove {in,out}_sleep_flag. It was used for the sleeper to + * determine if it was woken up by the expiring timeout or by + * an explicit wake_up. The return value from schedule_timeout + * can be used instead; if 0, the wakeup was due to the timeout. + * + * Rob Riggs Added persistent DMA buffers (1998/10/17) + */ + +#define BE_CONSERVATIVE +#define SAMPLE_ROUNDUP 0 + +#include <linux/mm.h> +#include <linux/gfp.h> +#include "sound_config.h" + +#define DMAP_FREE_ON_CLOSE 0 +#define DMAP_KEEP_ON_CLOSE 1 +extern int sound_dmap_flag; + +static void dma_reset_output(int dev); +static void dma_reset_input(int dev); +static int local_start_dma(struct audio_operations *adev, unsigned long physaddr, int count, int dma_mode); + + + +static int debugmem; /* switched off by default */ +static int dma_buffsize = DSP_BUFFSIZE; + +static long dmabuf_timeout(struct dma_buffparms *dmap) +{ + long tmout; + + tmout = (dmap->fragment_size * HZ) / dmap->data_rate; + tmout += HZ / 5; /* Some safety distance */ + if (tmout < (HZ / 2)) + tmout = HZ / 2; + if (tmout > 20 * HZ) + tmout = 20 * HZ; + return tmout; +} + +static int sound_alloc_dmap(struct dma_buffparms *dmap) +{ + char *start_addr, *end_addr; + int dma_pagesize; + int sz, size; + struct page *page; + + dmap->mapping_flags &= ~DMA_MAP_MAPPED; + + if (dmap->raw_buf != NULL) + return 0; /* Already done */ + if (dma_buffsize < 4096) + dma_buffsize = 4096; + dma_pagesize = (dmap->dma < 4) ? (64 * 1024) : (128 * 1024); + + /* + * Now check for the Cyrix problem. + */ + + if(isa_dma_bridge_buggy==2) + dma_pagesize=32768; + + dmap->raw_buf = NULL; + dmap->buffsize = dma_buffsize; + if (dmap->buffsize > dma_pagesize) + dmap->buffsize = dma_pagesize; + start_addr = NULL; + /* + * Now loop until we get a free buffer. Try to get smaller buffer if + * it fails. Don't accept smaller than 8k buffer for performance + * reasons. + */ + while (start_addr == NULL && dmap->buffsize > PAGE_SIZE) { + for (sz = 0, size = PAGE_SIZE; size < dmap->buffsize; sz++, size <<= 1); + dmap->buffsize = PAGE_SIZE * (1 << sz); + start_addr = (char *) __get_free_pages(GFP_ATOMIC|GFP_DMA|__GFP_NOWARN, sz); + if (start_addr == NULL) + dmap->buffsize /= 2; + } + + if (start_addr == NULL) { + printk(KERN_WARNING "Sound error: Couldn't allocate DMA buffer\n"); + return -ENOMEM; + } else { + /* make some checks */ + end_addr = start_addr + dmap->buffsize - 1; + + if (debugmem) + printk(KERN_DEBUG "sound: start 0x%lx, end 0x%lx\n", (long) start_addr, (long) end_addr); + + /* now check if it fits into the same dma-pagesize */ + + if (((long) start_addr & ~(dma_pagesize - 1)) != ((long) end_addr & ~(dma_pagesize - 1)) + || end_addr >= (char *) (MAX_DMA_ADDRESS)) { + printk(KERN_ERR "sound: Got invalid address 0x%lx for %db DMA-buffer\n", (long) start_addr, dmap->buffsize); + return -EFAULT; + } + } + dmap->raw_buf = start_addr; + dmap->raw_buf_phys = virt_to_bus(start_addr); + + for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) + SetPageReserved(page); + return 0; +} + +static void sound_free_dmap(struct dma_buffparms *dmap) +{ + int sz, size; + struct page *page; + unsigned long start_addr, end_addr; + + if (dmap->raw_buf == NULL) + return; + if (dmap->mapping_flags & DMA_MAP_MAPPED) + return; /* Don't free mmapped buffer. Will use it next time */ + for (sz = 0, size = PAGE_SIZE; size < dmap->buffsize; sz++, size <<= 1); + + start_addr = (unsigned long) dmap->raw_buf; + end_addr = start_addr + dmap->buffsize; + + for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) + ClearPageReserved(page); + + free_pages((unsigned long) dmap->raw_buf, sz); + dmap->raw_buf = NULL; +} + + +/* Intel version !!!!!!!!! */ + +static int sound_start_dma(struct dma_buffparms *dmap, unsigned long physaddr, int count, int dma_mode) +{ + unsigned long flags; + int chan = dmap->dma; + + /* printk( "Start DMA%d %d, %d\n", chan, (int)(physaddr-dmap->raw_buf_phys), count); */ + + flags = claim_dma_lock(); + disable_dma(chan); + clear_dma_ff(chan); + set_dma_mode(chan, dma_mode); + set_dma_addr(chan, physaddr); + set_dma_count(chan, count); + enable_dma(chan); + release_dma_lock(flags); + + return 0; +} + +static void dma_init_buffers(struct dma_buffparms *dmap) +{ + dmap->qlen = dmap->qhead = dmap->qtail = dmap->user_counter = 0; + dmap->byte_counter = 0; + dmap->max_byte_counter = 8000 * 60 * 60; + dmap->bytes_in_use = dmap->buffsize; + + dmap->dma_mode = DMODE_NONE; + dmap->mapping_flags = 0; + dmap->neutral_byte = 0x80; + dmap->data_rate = 8000; + dmap->cfrag = -1; + dmap->closing = 0; + dmap->nbufs = 1; + dmap->flags = DMA_BUSY; /* Other flags off */ +} + +static int open_dmap(struct audio_operations *adev, int mode, struct dma_buffparms *dmap) +{ + int err; + + if (dmap->flags & DMA_BUSY) + return -EBUSY; + if ((err = sound_alloc_dmap(dmap)) < 0) + return err; + + if (dmap->raw_buf == NULL) { + printk(KERN_WARNING "Sound: DMA buffers not available\n"); + return -ENOSPC; /* Memory allocation failed during boot */ + } + if (dmap->dma >= 0 && sound_open_dma(dmap->dma, adev->name)) { + printk(KERN_WARNING "Unable to grab(2) DMA%d for the audio driver\n", dmap->dma); + return -EBUSY; + } + dma_init_buffers(dmap); + spin_lock_init(&dmap->lock); + dmap->open_mode = mode; + dmap->subdivision = dmap->underrun_count = 0; + dmap->fragment_size = 0; + dmap->max_fragments = 65536; /* Just a large value */ + dmap->byte_counter = 0; + dmap->max_byte_counter = 8000 * 60 * 60; + dmap->applic_profile = APF_NORMAL; + dmap->needs_reorg = 1; + dmap->audio_callback = NULL; + dmap->callback_parm = 0; + return 0; +} + +static void close_dmap(struct audio_operations *adev, struct dma_buffparms *dmap) +{ + unsigned long flags; + + if (dmap->dma >= 0) { + sound_close_dma(dmap->dma); + flags=claim_dma_lock(); + disable_dma(dmap->dma); + release_dma_lock(flags); + } + if (dmap->flags & DMA_BUSY) + dmap->dma_mode = DMODE_NONE; + dmap->flags &= ~DMA_BUSY; + + if (sound_dmap_flag == DMAP_FREE_ON_CLOSE) + sound_free_dmap(dmap); +} + + +static unsigned int default_set_bits(int dev, unsigned int bits) +{ + mm_segment_t fs = get_fs(); + + set_fs(get_ds()); + audio_devs[dev]->d->ioctl(dev, SNDCTL_DSP_SETFMT, (void __user *)&bits); + set_fs(fs); + return bits; +} + +static int default_set_speed(int dev, int speed) +{ + mm_segment_t fs = get_fs(); + + set_fs(get_ds()); + audio_devs[dev]->d->ioctl(dev, SNDCTL_DSP_SPEED, (void __user *)&speed); + set_fs(fs); + return speed; +} + +static short default_set_channels(int dev, short channels) +{ + int c = channels; + mm_segment_t fs = get_fs(); + + set_fs(get_ds()); + audio_devs[dev]->d->ioctl(dev, SNDCTL_DSP_CHANNELS, (void __user *)&c); + set_fs(fs); + return c; +} + +static void check_driver(struct audio_driver *d) +{ + if (d->set_speed == NULL) + d->set_speed = default_set_speed; + if (d->set_bits == NULL) + d->set_bits = default_set_bits; + if (d->set_channels == NULL) + d->set_channels = default_set_channels; +} + +int DMAbuf_open(int dev, int mode) +{ + struct audio_operations *adev = audio_devs[dev]; + int retval; + struct dma_buffparms *dmap_in = NULL; + struct dma_buffparms *dmap_out = NULL; + + if (!adev) + return -ENXIO; + if (!(adev->flags & DMA_DUPLEX)) + adev->dmap_in = adev->dmap_out; + check_driver(adev->d); + + if ((retval = adev->d->open(dev, mode)) < 0) + return retval; + dmap_out = adev->dmap_out; + dmap_in = adev->dmap_in; + if (dmap_in == dmap_out) + adev->flags &= ~DMA_DUPLEX; + + if (mode & OPEN_WRITE) { + if ((retval = open_dmap(adev, mode, dmap_out)) < 0) { + adev->d->close(dev); + return retval; + } + } + adev->enable_bits = mode; + + if (mode == OPEN_READ || (mode != OPEN_WRITE && (adev->flags & DMA_DUPLEX))) { + if ((retval = open_dmap(adev, mode, dmap_in)) < 0) { + adev->d->close(dev); + if (mode & OPEN_WRITE) + close_dmap(adev, dmap_out); + return retval; + } + } + adev->open_mode = mode; + adev->go = 1; + + adev->d->set_bits(dev, 8); + adev->d->set_channels(dev, 1); + adev->d->set_speed(dev, DSP_DEFAULT_SPEED); + if (adev->dmap_out->dma_mode == DMODE_OUTPUT) + memset(adev->dmap_out->raw_buf, adev->dmap_out->neutral_byte, + adev->dmap_out->bytes_in_use); + return 0; +} +/* MUST not hold the spinlock */ +void DMAbuf_reset(int dev) +{ + if (audio_devs[dev]->open_mode & OPEN_WRITE) + dma_reset_output(dev); + + if (audio_devs[dev]->open_mode & OPEN_READ) + dma_reset_input(dev); +} + +static void dma_reset_output(int dev) +{ + struct audio_operations *adev = audio_devs[dev]; + unsigned long flags,f ; + struct dma_buffparms *dmap = adev->dmap_out; + + if (!(dmap->flags & DMA_STARTED)) /* DMA is not active */ + return; + + /* + * First wait until the current fragment has been played completely + */ + spin_lock_irqsave(&dmap->lock,flags); + adev->dmap_out->flags |= DMA_SYNCING; + + adev->dmap_out->underrun_count = 0; + if (!signal_pending(current) && adev->dmap_out->qlen && + adev->dmap_out->underrun_count == 0){ + spin_unlock_irqrestore(&dmap->lock,flags); + interruptible_sleep_on_timeout(&adev->out_sleeper, + dmabuf_timeout(dmap)); + spin_lock_irqsave(&dmap->lock,flags); + } + adev->dmap_out->flags &= ~(DMA_SYNCING | DMA_ACTIVE); + + /* + * Finally shut the device off + */ + if (!(adev->flags & DMA_DUPLEX) || !adev->d->halt_output) + adev->d->halt_io(dev); + else + adev->d->halt_output(dev); + adev->dmap_out->flags &= ~DMA_STARTED; + + f=claim_dma_lock(); + clear_dma_ff(dmap->dma); + disable_dma(dmap->dma); + release_dma_lock(f); + + dmap->byte_counter = 0; + reorganize_buffers(dev, adev->dmap_out, 0); + dmap->qlen = dmap->qhead = dmap->qtail = dmap->user_counter = 0; + spin_unlock_irqrestore(&dmap->lock,flags); +} + +static void dma_reset_input(int dev) +{ + struct audio_operations *adev = audio_devs[dev]; + unsigned long flags; + struct dma_buffparms *dmap = adev->dmap_in; + + spin_lock_irqsave(&dmap->lock,flags); + if (!(adev->flags & DMA_DUPLEX) || !adev->d->halt_input) + adev->d->halt_io(dev); + else + adev->d->halt_input(dev); + adev->dmap_in->flags &= ~DMA_STARTED; + + dmap->qlen = dmap->qhead = dmap->qtail = dmap->user_counter = 0; + dmap->byte_counter = 0; + reorganize_buffers(dev, adev->dmap_in, 1); + spin_unlock_irqrestore(&dmap->lock,flags); +} +/* MUST be called with holding the dmap->lock */ +void DMAbuf_launch_output(int dev, struct dma_buffparms *dmap) +{ + struct audio_operations *adev = audio_devs[dev]; + + if (!((adev->enable_bits * adev->go) & PCM_ENABLE_OUTPUT)) + return; /* Don't start DMA yet */ + dmap->dma_mode = DMODE_OUTPUT; + + if (!(dmap->flags & DMA_ACTIVE) || !(adev->flags & DMA_AUTOMODE) || (dmap->flags & DMA_NODMA)) { + if (!(dmap->flags & DMA_STARTED)) { + reorganize_buffers(dev, dmap, 0); + if (adev->d->prepare_for_output(dev, dmap->fragment_size, dmap->nbufs)) + return; + if (!(dmap->flags & DMA_NODMA)) + local_start_dma(adev, dmap->raw_buf_phys, dmap->bytes_in_use,DMA_MODE_WRITE); + dmap->flags |= DMA_STARTED; + } + if (dmap->counts[dmap->qhead] == 0) + dmap->counts[dmap->qhead] = dmap->fragment_size; + dmap->dma_mode = DMODE_OUTPUT; + adev->d->output_block(dev, dmap->raw_buf_phys + dmap->qhead * dmap->fragment_size, + dmap->counts[dmap->qhead], 1); + if (adev->d->trigger) + adev->d->trigger(dev,adev->enable_bits * adev->go); + } + dmap->flags |= DMA_ACTIVE; +} + +int DMAbuf_sync(int dev) +{ + struct audio_operations *adev = audio_devs[dev]; + unsigned long flags; + int n = 0; + struct dma_buffparms *dmap; + + if (!adev->go && !(adev->enable_bits & PCM_ENABLE_OUTPUT)) + return 0; + + if (adev->dmap_out->dma_mode == DMODE_OUTPUT) { + dmap = adev->dmap_out; + spin_lock_irqsave(&dmap->lock,flags); + if (dmap->qlen > 0 && !(dmap->flags & DMA_ACTIVE)) + DMAbuf_launch_output(dev, dmap); + adev->dmap_out->flags |= DMA_SYNCING; + adev->dmap_out->underrun_count = 0; + while (!signal_pending(current) && n++ < adev->dmap_out->nbufs && + adev->dmap_out->qlen && adev->dmap_out->underrun_count == 0) { + long t = dmabuf_timeout(dmap); + spin_unlock_irqrestore(&dmap->lock,flags); + /* FIXME: not safe may miss events */ + t = interruptible_sleep_on_timeout(&adev->out_sleeper, t); + spin_lock_irqsave(&dmap->lock,flags); + if (!t) { + adev->dmap_out->flags &= ~DMA_SYNCING; + spin_unlock_irqrestore(&dmap->lock,flags); + return adev->dmap_out->qlen; + } + } + adev->dmap_out->flags &= ~(DMA_SYNCING | DMA_ACTIVE); + + /* + * Some devices such as GUS have huge amount of on board RAM for the + * audio data. We have to wait until the device has finished playing. + */ + + /* still holding the lock */ + if (adev->d->local_qlen) { /* Device has hidden buffers */ + while (!signal_pending(current) && + adev->d->local_qlen(dev)){ + spin_unlock_irqrestore(&dmap->lock,flags); + interruptible_sleep_on_timeout(&adev->out_sleeper, + dmabuf_timeout(dmap)); + spin_lock_irqsave(&dmap->lock,flags); + } + } + spin_unlock_irqrestore(&dmap->lock,flags); + } + adev->dmap_out->dma_mode = DMODE_NONE; + return adev->dmap_out->qlen; +} + +int DMAbuf_release(int dev, int mode) +{ + struct audio_operations *adev = audio_devs[dev]; + struct dma_buffparms *dmap; + unsigned long flags; + + dmap = adev->dmap_out; + if (adev->open_mode & OPEN_WRITE) + adev->dmap_out->closing = 1; + + if (adev->open_mode & OPEN_READ){ + adev->dmap_in->closing = 1; + dmap = adev->dmap_in; + } + if (adev->open_mode & OPEN_WRITE) + if (!(adev->dmap_out->mapping_flags & DMA_MAP_MAPPED)) + if (!signal_pending(current) && (adev->dmap_out->dma_mode == DMODE_OUTPUT)) + DMAbuf_sync(dev); + if (adev->dmap_out->dma_mode == DMODE_OUTPUT) + memset(adev->dmap_out->raw_buf, adev->dmap_out->neutral_byte, adev->dmap_out->bytes_in_use); + + DMAbuf_reset(dev); + spin_lock_irqsave(&dmap->lock,flags); + adev->d->close(dev); + + if (adev->open_mode & OPEN_WRITE) + close_dmap(adev, adev->dmap_out); + + if (adev->open_mode == OPEN_READ || + (adev->open_mode != OPEN_WRITE && + (adev->flags & DMA_DUPLEX))) + close_dmap(adev, adev->dmap_in); + adev->open_mode = 0; + spin_unlock_irqrestore(&dmap->lock,flags); + return 0; +} +/* called with dmap->lock dold */ +int DMAbuf_activate_recording(int dev, struct dma_buffparms *dmap) +{ + struct audio_operations *adev = audio_devs[dev]; + int err; + + if (!(adev->open_mode & OPEN_READ)) + return 0; + if (!(adev->enable_bits & PCM_ENABLE_INPUT)) + return 0; + if (dmap->dma_mode == DMODE_OUTPUT) { /* Direction change */ + /* release lock - it's not recursive */ + spin_unlock_irq(&dmap->lock); + DMAbuf_sync(dev); + DMAbuf_reset(dev); + spin_lock_irq(&dmap->lock); + dmap->dma_mode = DMODE_NONE; + } + if (!dmap->dma_mode) { + reorganize_buffers(dev, dmap, 1); + if ((err = adev->d->prepare_for_input(dev, + dmap->fragment_size, dmap->nbufs)) < 0) + return err; + dmap->dma_mode = DMODE_INPUT; + } + if (!(dmap->flags & DMA_ACTIVE)) { + if (dmap->needs_reorg) + reorganize_buffers(dev, dmap, 0); + local_start_dma(adev, dmap->raw_buf_phys, dmap->bytes_in_use, DMA_MODE_READ); + adev->d->start_input(dev, dmap->raw_buf_phys + dmap->qtail * dmap->fragment_size, + dmap->fragment_size, 0); + dmap->flags |= DMA_ACTIVE; + if (adev->d->trigger) + adev->d->trigger(dev, adev->enable_bits * adev->go); + } + return 0; +} +/* acquires lock */ +int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock) +{ + struct audio_operations *adev = audio_devs[dev]; + unsigned long flags; + int err = 0, n = 0; + struct dma_buffparms *dmap = adev->dmap_in; + int go; + + if (!(adev->open_mode & OPEN_READ)) + return -EIO; + spin_lock_irqsave(&dmap->lock,flags); + if (dmap->needs_reorg) + reorganize_buffers(dev, dmap, 0); + if (adev->dmap_in->mapping_flags & DMA_MAP_MAPPED) { +/* printk(KERN_WARNING "Sound: Can't read from mmapped device (1)\n");*/ + spin_unlock_irqrestore(&dmap->lock,flags); + return -EINVAL; + } else while (dmap->qlen <= 0 && n++ < 10) { + long timeout = MAX_SCHEDULE_TIMEOUT; + if (!(adev->enable_bits & PCM_ENABLE_INPUT) || !adev->go) { + spin_unlock_irqrestore(&dmap->lock,flags); + return -EAGAIN; + } + if ((err = DMAbuf_activate_recording(dev, dmap)) < 0) { + spin_unlock_irqrestore(&dmap->lock,flags); + return err; + } + /* Wait for the next block */ + + if (dontblock) { + spin_unlock_irqrestore(&dmap->lock,flags); + return -EAGAIN; + } + if ((go = adev->go)) + timeout = dmabuf_timeout(dmap); + + spin_unlock_irqrestore(&dmap->lock,flags); + timeout = interruptible_sleep_on_timeout(&adev->in_sleeper, + timeout); + if (!timeout) { + /* FIXME: include device name */ + err = -EIO; + printk(KERN_WARNING "Sound: DMA (input) timed out - IRQ/DRQ config error?\n"); + dma_reset_input(dev); + } else + err = -EINTR; + spin_lock_irqsave(&dmap->lock,flags); + } + spin_unlock_irqrestore(&dmap->lock,flags); + + if (dmap->qlen <= 0) + return err ? err : -EINTR; + *buf = &dmap->raw_buf[dmap->qhead * dmap->fragment_size + dmap->counts[dmap->qhead]]; + *len = dmap->fragment_size - dmap->counts[dmap->qhead]; + + return dmap->qhead; +} + +int DMAbuf_rmchars(int dev, int buff_no, int c) +{ + struct audio_operations *adev = audio_devs[dev]; + struct dma_buffparms *dmap = adev->dmap_in; + int p = dmap->counts[dmap->qhead] + c; + + if (dmap->mapping_flags & DMA_MAP_MAPPED) + { +/* printk("Sound: Can't read from mmapped device (2)\n");*/ + return -EINVAL; + } + else if (dmap->qlen <= 0) + return -EIO; + else if (p >= dmap->fragment_size) { /* This buffer is completely empty */ + dmap->counts[dmap->qhead] = 0; + dmap->qlen--; + dmap->qhead = (dmap->qhead + 1) % dmap->nbufs; + } + else dmap->counts[dmap->qhead] = p; + + return 0; +} +/* MUST be called with dmap->lock hold */ +int DMAbuf_get_buffer_pointer(int dev, struct dma_buffparms *dmap, int direction) +{ + /* + * Try to approximate the active byte position of the DMA pointer within the + * buffer area as well as possible. + */ + + int pos; + unsigned long f; + + if (!(dmap->flags & DMA_ACTIVE)) + pos = 0; + else { + int chan = dmap->dma; + + f=claim_dma_lock(); + clear_dma_ff(chan); + + if(!isa_dma_bridge_buggy) + disable_dma(dmap->dma); + + pos = get_dma_residue(chan); + + pos = dmap->bytes_in_use - pos; + + if (!(dmap->mapping_flags & DMA_MAP_MAPPED)) { + if (direction == DMODE_OUTPUT) { + if (dmap->qhead == 0) + if (pos > dmap->fragment_size) + pos = 0; + } else { + if (dmap->qtail == 0) + if (pos > dmap->fragment_size) + pos = 0; + } + } + if (pos < 0) + pos = 0; + if (pos >= dmap->bytes_in_use) + pos = 0; + + if(!isa_dma_bridge_buggy) + enable_dma(dmap->dma); + + release_dma_lock(f); + } + /* printk( "%04x ", pos); */ + + return pos; +} + +/* + * DMAbuf_start_devices() is called by the /dev/music driver to start + * one or more audio devices at desired moment. + */ + +void DMAbuf_start_devices(unsigned int devmask) +{ + struct audio_operations *adev; + int dev; + + for (dev = 0; dev < num_audiodevs; dev++) { + if (!(devmask & (1 << dev))) + continue; + if (!(adev = audio_devs[dev])) + continue; + if (adev->open_mode == 0) + continue; + if (adev->go) + continue; + /* OK to start the device */ + adev->go = 1; + if (adev->d->trigger) + adev->d->trigger(dev,adev->enable_bits * adev->go); + } +} +/* via poll called without a lock ?*/ +int DMAbuf_space_in_queue(int dev) +{ + struct audio_operations *adev = audio_devs[dev]; + int len, max, tmp; + struct dma_buffparms *dmap = adev->dmap_out; + int lim = dmap->nbufs; + + if (lim < 2) + lim = 2; + + if (dmap->qlen >= lim) /* No space at all */ + return 0; + + /* + * Verify that there are no more pending buffers than the limit + * defined by the process. + */ + + max = dmap->max_fragments; + if (max > lim) + max = lim; + len = dmap->qlen; + + if (adev->d->local_qlen) { + tmp = adev->d->local_qlen(dev); + if (tmp && len) + tmp--; /* This buffer has been counted twice */ + len += tmp; + } + if (dmap->byte_counter % dmap->fragment_size) /* There is a partial fragment */ + len = len + 1; + + if (len >= max) + return 0; + return max - len; +} +/* MUST not hold the spinlock - this function may sleep */ +static int output_sleep(int dev, int dontblock) +{ + struct audio_operations *adev = audio_devs[dev]; + int err = 0; + struct dma_buffparms *dmap = adev->dmap_out; + long timeout; + long timeout_value; + + if (dontblock) + return -EAGAIN; + if (!(adev->enable_bits & PCM_ENABLE_OUTPUT)) + return -EAGAIN; + + /* + * Wait for free space + */ + if (signal_pending(current)) + return -EINTR; + timeout = (adev->go && !(dmap->flags & DMA_NOTIMEOUT)); + if (timeout) + timeout_value = dmabuf_timeout(dmap); + else + timeout_value = MAX_SCHEDULE_TIMEOUT; + timeout_value = interruptible_sleep_on_timeout(&adev->out_sleeper, + timeout_value); + if (timeout != MAX_SCHEDULE_TIMEOUT && !timeout_value) { + printk(KERN_WARNING "Sound: DMA (output) timed out - IRQ/DRQ config error?\n"); + dma_reset_output(dev); + } else { + if (signal_pending(current)) + err = -EINTR; + } + return err; +} +/* called with the lock held */ +static int find_output_space(int dev, char **buf, int *size) +{ + struct audio_operations *adev = audio_devs[dev]; + struct dma_buffparms *dmap = adev->dmap_out; + unsigned long active_offs; + long len, offs; + int maxfrags; + int occupied_bytes = (dmap->user_counter % dmap->fragment_size); + + *buf = dmap->raw_buf; + if (!(maxfrags = DMAbuf_space_in_queue(dev)) && !occupied_bytes) + return 0; + +#ifdef BE_CONSERVATIVE + active_offs = dmap->byte_counter + dmap->qhead * dmap->fragment_size; +#else + active_offs = max(DMAbuf_get_buffer_pointer(dev, dmap, DMODE_OUTPUT), 0); + /* Check for pointer wrapping situation */ + if (active_offs >= dmap->bytes_in_use) + active_offs = 0; + active_offs += dmap->byte_counter; +#endif + + offs = (dmap->user_counter % dmap->bytes_in_use) & ~SAMPLE_ROUNDUP; + if (offs < 0 || offs >= dmap->bytes_in_use) { + printk(KERN_ERR "Sound: Got unexpected offs %ld. Giving up.\n", offs); + printk("Counter = %ld, bytes=%d\n", dmap->user_counter, dmap->bytes_in_use); + return 0; + } + *buf = dmap->raw_buf + offs; + + len = active_offs + dmap->bytes_in_use - dmap->user_counter; /* Number of unused bytes in buffer */ + + if ((offs + len) > dmap->bytes_in_use) + len = dmap->bytes_in_use - offs; + if (len < 0) { + return 0; + } + if (len > ((maxfrags * dmap->fragment_size) - occupied_bytes)) + len = (maxfrags * dmap->fragment_size) - occupied_bytes; + *size = len & ~SAMPLE_ROUNDUP; + return (*size > 0); +} +/* acquires lock */ +int DMAbuf_getwrbuffer(int dev, char **buf, int *size, int dontblock) +{ + struct audio_operations *adev = audio_devs[dev]; + unsigned long flags; + int err = -EIO; + struct dma_buffparms *dmap = adev->dmap_out; + + if (dmap->mapping_flags & DMA_MAP_MAPPED) { +/* printk(KERN_DEBUG "Sound: Can't write to mmapped device (3)\n");*/ + return -EINVAL; + } + spin_lock_irqsave(&dmap->lock,flags); + if (dmap->needs_reorg) + reorganize_buffers(dev, dmap, 0); + + if (dmap->dma_mode == DMODE_INPUT) { /* Direction change */ + spin_unlock_irqrestore(&dmap->lock,flags); + DMAbuf_reset(dev); + spin_lock_irqsave(&dmap->lock,flags); + } + dmap->dma_mode = DMODE_OUTPUT; + + while (find_output_space(dev, buf, size) <= 0) { + spin_unlock_irqrestore(&dmap->lock,flags); + if ((err = output_sleep(dev, dontblock)) < 0) { + return err; + } + spin_lock_irqsave(&dmap->lock,flags); + } + + spin_unlock_irqrestore(&dmap->lock,flags); + return 0; +} +/* has to acquire dmap->lock */ +int DMAbuf_move_wrpointer(int dev, int l) +{ + struct audio_operations *adev = audio_devs[dev]; + struct dma_buffparms *dmap = adev->dmap_out; + unsigned long ptr; + unsigned long end_ptr, p; + int post; + unsigned long flags; + + spin_lock_irqsave(&dmap->lock,flags); + post= (dmap->flags & DMA_POST); + ptr = (dmap->user_counter / dmap->fragment_size) * dmap->fragment_size; + + dmap->flags &= ~DMA_POST; + dmap->cfrag = -1; + dmap->user_counter += l; + dmap->flags |= DMA_DIRTY; + + if (dmap->byte_counter >= dmap->max_byte_counter) { + /* Wrap the byte counters */ + long decr = dmap->byte_counter; + dmap->byte_counter = (dmap->byte_counter % dmap->bytes_in_use); + decr -= dmap->byte_counter; + dmap->user_counter -= decr; + } + end_ptr = (dmap->user_counter / dmap->fragment_size) * dmap->fragment_size; + + p = (dmap->user_counter - 1) % dmap->bytes_in_use; + dmap->neutral_byte = dmap->raw_buf[p]; + + /* Update the fragment based bookkeeping too */ + while (ptr < end_ptr) { + dmap->counts[dmap->qtail] = dmap->fragment_size; + dmap->qtail = (dmap->qtail + 1) % dmap->nbufs; + dmap->qlen++; + ptr += dmap->fragment_size; + } + + dmap->counts[dmap->qtail] = dmap->user_counter - ptr; + + /* + * Let the low level driver perform some postprocessing to + * the written data. + */ + if (adev->d->postprocess_write) + adev->d->postprocess_write(dev); + + if (!(dmap->flags & DMA_ACTIVE)) + if (dmap->qlen > 1 || (dmap->qlen > 0 && (post || dmap->qlen >= dmap->nbufs - 1))) + DMAbuf_launch_output(dev, dmap); + + spin_unlock_irqrestore(&dmap->lock,flags); + return 0; +} + +int DMAbuf_start_dma(int dev, unsigned long physaddr, int count, int dma_mode) +{ + struct audio_operations *adev = audio_devs[dev]; + struct dma_buffparms *dmap = (dma_mode == DMA_MODE_WRITE) ? adev->dmap_out : adev->dmap_in; + + if (dmap->raw_buf == NULL) { + printk(KERN_ERR "sound: DMA buffer(1) == NULL\n"); + printk("Device %d, chn=%s\n", dev, (dmap == adev->dmap_out) ? "out" : "in"); + return 0; + } + if (dmap->dma < 0) + return 0; + sound_start_dma(dmap, physaddr, count, dma_mode); + return count; +} +EXPORT_SYMBOL(DMAbuf_start_dma); + +static int local_start_dma(struct audio_operations *adev, unsigned long physaddr, int count, int dma_mode) +{ + struct dma_buffparms *dmap = (dma_mode == DMA_MODE_WRITE) ? adev->dmap_out : adev->dmap_in; + + if (dmap->raw_buf == NULL) { + printk(KERN_ERR "sound: DMA buffer(2) == NULL\n"); + printk(KERN_ERR "Device %s, chn=%s\n", adev->name, (dmap == adev->dmap_out) ? "out" : "in"); + return 0; + } + if (dmap->flags & DMA_NODMA) + return 1; + if (dmap->dma < 0) + return 0; + sound_start_dma(dmap, dmap->raw_buf_phys, dmap->bytes_in_use, dma_mode | DMA_AUTOINIT); + dmap->flags |= DMA_STARTED; + return count; +} + +static void finish_output_interrupt(int dev, struct dma_buffparms *dmap) +{ + struct audio_operations *adev = audio_devs[dev]; + + if (dmap->audio_callback != NULL) + dmap->audio_callback(dev, dmap->callback_parm); + wake_up(&adev->out_sleeper); + wake_up(&adev->poll_sleeper); +} +/* called with dmap->lock held in irq context*/ +static void do_outputintr(int dev, int dummy) +{ + struct audio_operations *adev = audio_devs[dev]; + struct dma_buffparms *dmap = adev->dmap_out; + int this_fragment; + + if (dmap->raw_buf == NULL) { + printk(KERN_ERR "Sound: Error. Audio interrupt (%d) after freeing buffers.\n", dev); + return; + } + if (dmap->mapping_flags & DMA_MAP_MAPPED) { /* Virtual memory mapped access */ + /* mmapped access */ + dmap->qhead = (dmap->qhead + 1) % dmap->nbufs; + if (dmap->qhead == 0) { /* Wrapped */ + dmap->byte_counter += dmap->bytes_in_use; + if (dmap->byte_counter >= dmap->max_byte_counter) { /* Overflow */ + long decr = dmap->byte_counter; + dmap->byte_counter = (dmap->byte_counter % dmap->bytes_in_use); + decr -= dmap->byte_counter; + dmap->user_counter -= decr; + } + } + dmap->qlen++; /* Yes increment it (don't decrement) */ + if (!(adev->flags & DMA_AUTOMODE)) + dmap->flags &= ~DMA_ACTIVE; + dmap->counts[dmap->qhead] = dmap->fragment_size; + DMAbuf_launch_output(dev, dmap); + finish_output_interrupt(dev, dmap); + return; + } + + dmap->qlen--; + this_fragment = dmap->qhead; + dmap->qhead = (dmap->qhead + 1) % dmap->nbufs; + + if (dmap->qhead == 0) { /* Wrapped */ + dmap->byte_counter += dmap->bytes_in_use; + if (dmap->byte_counter >= dmap->max_byte_counter) { /* Overflow */ + long decr = dmap->byte_counter; + dmap->byte_counter = (dmap->byte_counter % dmap->bytes_in_use); + decr -= dmap->byte_counter; + dmap->user_counter -= decr; + } + } + if (!(adev->flags & DMA_AUTOMODE)) + dmap->flags &= ~DMA_ACTIVE; + + /* + * This is dmap->qlen <= 0 except when closing when + * dmap->qlen < 0 + */ + + while (dmap->qlen <= -dmap->closing) { + dmap->underrun_count++; + dmap->qlen++; + if ((dmap->flags & DMA_DIRTY) && dmap->applic_profile != APF_CPUINTENS) { + dmap->flags &= ~DMA_DIRTY; + memset(adev->dmap_out->raw_buf, adev->dmap_out->neutral_byte, + adev->dmap_out->buffsize); + } + dmap->user_counter += dmap->fragment_size; + dmap->qtail = (dmap->qtail + 1) % dmap->nbufs; + } + if (dmap->qlen > 0) + DMAbuf_launch_output(dev, dmap); + finish_output_interrupt(dev, dmap); +} +/* called in irq context */ +void DMAbuf_outputintr(int dev, int notify_only) +{ + struct audio_operations *adev = audio_devs[dev]; + unsigned long flags; + struct dma_buffparms *dmap = adev->dmap_out; + + spin_lock_irqsave(&dmap->lock,flags); + if (!(dmap->flags & DMA_NODMA)) { + int chan = dmap->dma, pos, n; + unsigned long f; + + f=claim_dma_lock(); + + if(!isa_dma_bridge_buggy) + disable_dma(dmap->dma); + clear_dma_ff(chan); + pos = dmap->bytes_in_use - get_dma_residue(chan); + if(!isa_dma_bridge_buggy) + enable_dma(dmap->dma); + release_dma_lock(f); + + pos = pos / dmap->fragment_size; /* Actual qhead */ + if (pos < 0 || pos >= dmap->nbufs) + pos = 0; + n = 0; + while (dmap->qhead != pos && n++ < dmap->nbufs) + do_outputintr(dev, notify_only); + } + else + do_outputintr(dev, notify_only); + spin_unlock_irqrestore(&dmap->lock,flags); +} +EXPORT_SYMBOL(DMAbuf_outputintr); + +/* called with dmap->lock held in irq context */ +static void do_inputintr(int dev) +{ + struct audio_operations *adev = audio_devs[dev]; + struct dma_buffparms *dmap = adev->dmap_in; + + if (dmap->raw_buf == NULL) { + printk(KERN_ERR "Sound: Fatal error. Audio interrupt after freeing buffers.\n"); + return; + } + if (dmap->mapping_flags & DMA_MAP_MAPPED) { + dmap->qtail = (dmap->qtail + 1) % dmap->nbufs; + if (dmap->qtail == 0) { /* Wrapped */ + dmap->byte_counter += dmap->bytes_in_use; + if (dmap->byte_counter >= dmap->max_byte_counter) { /* Overflow */ + long decr = dmap->byte_counter; + dmap->byte_counter = (dmap->byte_counter % dmap->bytes_in_use) + dmap->bytes_in_use; + decr -= dmap->byte_counter; + dmap->user_counter -= decr; + } + } + dmap->qlen++; + + if (!(adev->flags & DMA_AUTOMODE)) { + if (dmap->needs_reorg) + reorganize_buffers(dev, dmap, 0); + local_start_dma(adev, dmap->raw_buf_phys, dmap->bytes_in_use,DMA_MODE_READ); + adev->d->start_input(dev, dmap->raw_buf_phys + dmap->qtail * dmap->fragment_size, + dmap->fragment_size, 1); + if (adev->d->trigger) + adev->d->trigger(dev, adev->enable_bits * adev->go); + } + dmap->flags |= DMA_ACTIVE; + } else if (dmap->qlen >= (dmap->nbufs - 1)) { + printk(KERN_WARNING "Sound: Recording overrun\n"); + dmap->underrun_count++; + + /* Just throw away the oldest fragment but keep the engine running */ + dmap->qhead = (dmap->qhead + 1) % dmap->nbufs; + dmap->qtail = (dmap->qtail + 1) % dmap->nbufs; + } else if (dmap->qlen >= 0 && dmap->qlen < dmap->nbufs) { + dmap->qlen++; + dmap->qtail = (dmap->qtail + 1) % dmap->nbufs; + if (dmap->qtail == 0) { /* Wrapped */ + dmap->byte_counter += dmap->bytes_in_use; + if (dmap->byte_counter >= dmap->max_byte_counter) { /* Overflow */ + long decr = dmap->byte_counter; + dmap->byte_counter = (dmap->byte_counter % dmap->bytes_in_use) + dmap->bytes_in_use; + decr -= dmap->byte_counter; + dmap->user_counter -= decr; + } + } + } + if (!(adev->flags & DMA_AUTOMODE) || (dmap->flags & DMA_NODMA)) { + local_start_dma(adev, dmap->raw_buf_phys, dmap->bytes_in_use, DMA_MODE_READ); + adev->d->start_input(dev, dmap->raw_buf_phys + dmap->qtail * dmap->fragment_size, dmap->fragment_size, 1); + if (adev->d->trigger) + adev->d->trigger(dev,adev->enable_bits * adev->go); + } + dmap->flags |= DMA_ACTIVE; + if (dmap->qlen > 0) + { + wake_up(&adev->in_sleeper); + wake_up(&adev->poll_sleeper); + } +} +/* called in irq context */ +void DMAbuf_inputintr(int dev) +{ + struct audio_operations *adev = audio_devs[dev]; + struct dma_buffparms *dmap = adev->dmap_in; + unsigned long flags; + + spin_lock_irqsave(&dmap->lock,flags); + + if (!(dmap->flags & DMA_NODMA)) { + int chan = dmap->dma, pos, n; + unsigned long f; + + f=claim_dma_lock(); + if(!isa_dma_bridge_buggy) + disable_dma(dmap->dma); + clear_dma_ff(chan); + pos = dmap->bytes_in_use - get_dma_residue(chan); + if(!isa_dma_bridge_buggy) + enable_dma(dmap->dma); + release_dma_lock(f); + + pos = pos / dmap->fragment_size; /* Actual qhead */ + if (pos < 0 || pos >= dmap->nbufs) + pos = 0; + + n = 0; + while (dmap->qtail != pos && ++n < dmap->nbufs) + do_inputintr(dev); + } else + do_inputintr(dev); + spin_unlock_irqrestore(&dmap->lock,flags); +} +EXPORT_SYMBOL(DMAbuf_inputintr); + +void DMAbuf_init(int dev, int dma1, int dma2) +{ + struct audio_operations *adev = audio_devs[dev]; + /* + * NOTE! This routine could be called several times. + */ + + if (adev && adev->dmap_out == NULL) { + if (adev->d == NULL) + panic("OSS: audio_devs[%d]->d == NULL\n", dev); + + if (adev->parent_dev) { /* Use DMA map of the parent dev */ + int parent = adev->parent_dev - 1; + adev->dmap_out = audio_devs[parent]->dmap_out; + adev->dmap_in = audio_devs[parent]->dmap_in; + } else { + adev->dmap_out = adev->dmap_in = &adev->dmaps[0]; + adev->dmap_out->dma = dma1; + if (adev->flags & DMA_DUPLEX) { + adev->dmap_in = &adev->dmaps[1]; + adev->dmap_in->dma = dma2; + } + } + /* Persistent DMA buffers allocated here */ + if (sound_dmap_flag == DMAP_KEEP_ON_CLOSE) { + if (adev->dmap_in->raw_buf == NULL) + sound_alloc_dmap(adev->dmap_in); + if (adev->dmap_out->raw_buf == NULL) + sound_alloc_dmap(adev->dmap_out); + } + } +} + +/* No kernel lock - DMAbuf_activate_recording protected by global cli/sti */ +static unsigned int poll_input(struct file * file, int dev, poll_table *wait) +{ + struct audio_operations *adev = audio_devs[dev]; + struct dma_buffparms *dmap = adev->dmap_in; + + if (!(adev->open_mode & OPEN_READ)) + return 0; + if (dmap->mapping_flags & DMA_MAP_MAPPED) { + if (dmap->qlen) + return POLLIN | POLLRDNORM; + return 0; + } + if (dmap->dma_mode != DMODE_INPUT) { + if (dmap->dma_mode == DMODE_NONE && + adev->enable_bits & PCM_ENABLE_INPUT && + !dmap->qlen && adev->go) { + unsigned long flags; + + spin_lock_irqsave(&dmap->lock,flags); + DMAbuf_activate_recording(dev, dmap); + spin_unlock_irqrestore(&dmap->lock,flags); + } + return 0; + } + if (!dmap->qlen) + return 0; + return POLLIN | POLLRDNORM; +} + +static unsigned int poll_output(struct file * file, int dev, poll_table *wait) +{ + struct audio_operations *adev = audio_devs[dev]; + struct dma_buffparms *dmap = adev->dmap_out; + + if (!(adev->open_mode & OPEN_WRITE)) + return 0; + if (dmap->mapping_flags & DMA_MAP_MAPPED) { + if (dmap->qlen) + return POLLOUT | POLLWRNORM; + return 0; + } + if (dmap->dma_mode == DMODE_INPUT) + return 0; + if (dmap->dma_mode == DMODE_NONE) + return POLLOUT | POLLWRNORM; + if (!DMAbuf_space_in_queue(dev)) + return 0; + return POLLOUT | POLLWRNORM; +} + +unsigned int DMAbuf_poll(struct file * file, int dev, poll_table *wait) +{ + struct audio_operations *adev = audio_devs[dev]; + poll_wait(file, &adev->poll_sleeper, wait); + return poll_input(file, dev, wait) | poll_output(file, dev, wait); +} + +void DMAbuf_deinit(int dev) +{ + struct audio_operations *adev = audio_devs[dev]; + /* This routine is called when driver is being unloaded */ + if (!adev) + return; + + /* Persistent DMA buffers deallocated here */ + if (sound_dmap_flag == DMAP_KEEP_ON_CLOSE) { + sound_free_dmap(adev->dmap_out); + if (adev->flags & DMA_DUPLEX) + sound_free_dmap(adev->dmap_in); + } +} diff --git a/sound/oss/dmasound/Kconfig b/sound/oss/dmasound/Kconfig new file mode 100644 index 00000000..f456574a --- /dev/null +++ b/sound/oss/dmasound/Kconfig @@ -0,0 +1,45 @@ +config DMASOUND_ATARI + tristate "Atari DMA sound support" + depends on ATARI && SOUND + select DMASOUND + help + If you want to use the internal audio of your Atari in Linux, answer + Y to this question. This will provide a Sun-like /dev/audio, + compatible with the Linux/i386 sound system. Otherwise, say N. + + This driver is also available as a module ( = code which can be + inserted in and removed from the running kernel whenever you + want). If you want to compile it as a module, say M here and read + <file:Documentation/kbuild/modules.txt>. + +config DMASOUND_PAULA + tristate "Amiga DMA sound support" + depends on AMIGA && SOUND + select DMASOUND + help + If you want to use the internal audio of your Amiga in Linux, answer + Y to this question. This will provide a Sun-like /dev/audio, + compatible with the Linux/i386 sound system. Otherwise, say N. + + This driver is also available as a module ( = code which can be + inserted in and removed from the running kernel whenever you + want). If you want to compile it as a module, say M here and read + <file:Documentation/kbuild/modules.txt>. + +config DMASOUND_Q40 + tristate "Q40 sound support" + depends on Q40 && SOUND + select DMASOUND + help + If you want to use the internal audio of your Q40 in Linux, answer + Y to this question. This will provide a Sun-like /dev/audio, + compatible with the Linux/i386 sound system. Otherwise, say N. + + This driver is also available as a module ( = code which can be + inserted in and removed from the running kernel whenever you + want). If you want to compile it as a module, say M here and read + <file:Documentation/kbuild/modules.txt>. + +config DMASOUND + tristate + select SOUND_OSS_CORE diff --git a/sound/oss/dmasound/Makefile b/sound/oss/dmasound/Makefile new file mode 100644 index 00000000..3c153165 --- /dev/null +++ b/sound/oss/dmasound/Makefile @@ -0,0 +1,7 @@ +# +# Makefile for the DMA sound driver +# + +obj-$(CONFIG_DMASOUND_ATARI) += dmasound_core.o dmasound_atari.o +obj-$(CONFIG_DMASOUND_PAULA) += dmasound_core.o dmasound_paula.o +obj-$(CONFIG_DMASOUND_Q40) += dmasound_core.o dmasound_q40.o diff --git a/sound/oss/dmasound/dmasound.h b/sound/oss/dmasound/dmasound.h new file mode 100644 index 00000000..1308d8d3 --- /dev/null +++ b/sound/oss/dmasound/dmasound.h @@ -0,0 +1,262 @@ +#ifndef _dmasound_h_ +/* + * linux/sound/oss/dmasound/dmasound.h + * + * + * Minor numbers for the sound driver. + * + * Unfortunately Creative called the codec chip of SB as a DSP. For this + * reason the /dev/dsp is reserved for digitized audio use. There is a + * device for true DSP processors but it will be called something else. + * In v3.0 it's /dev/sndproc but this could be a temporary solution. + */ +#define _dmasound_h_ + +#include <linux/types.h> + +#define SND_NDEVS 256 /* Number of supported devices */ +#define SND_DEV_CTL 0 /* Control port /dev/mixer */ +#define SND_DEV_SEQ 1 /* Sequencer output /dev/sequencer (FM + synthesizer and MIDI output) */ +#define SND_DEV_MIDIN 2 /* Raw midi access */ +#define SND_DEV_DSP 3 /* Digitized voice /dev/dsp */ +#define SND_DEV_AUDIO 4 /* Sparc compatible /dev/audio */ +#define SND_DEV_DSP16 5 /* Like /dev/dsp but 16 bits/sample */ +#define SND_DEV_STATUS 6 /* /dev/sndstat */ +/* #7 not in use now. Was in 2.4. Free for use after v3.0. */ +#define SND_DEV_SEQ2 8 /* /dev/sequencer, level 2 interface */ +#define SND_DEV_SNDPROC 9 /* /dev/sndproc for programmable devices */ +#define SND_DEV_PSS SND_DEV_SNDPROC + +/* switch on various prinks */ +#define DEBUG_DMASOUND 1 + +#define MAX_AUDIO_DEV 5 +#define MAX_MIXER_DEV 4 +#define MAX_SYNTH_DEV 3 +#define MAX_MIDI_DEV 6 +#define MAX_TIMER_DEV 3 + +#define MAX_CATCH_RADIUS 10 + +#define le2be16(x) (((x)<<8 & 0xff00) | ((x)>>8 & 0x00ff)) +#define le2be16dbl(x) (((x)<<8 & 0xff00ff00) | ((x)>>8 & 0x00ff00ff)) + +#define IOCTL_IN(arg, ret) \ + do { int error = get_user(ret, (int __user *)(arg)); \ + if (error) return error; \ + } while (0) +#define IOCTL_OUT(arg, ret) ioctl_return((int __user *)(arg), ret) + +static inline int ioctl_return(int __user *addr, int value) +{ + return value < 0 ? value : put_user(value, addr); +} + + + /* + * Configuration + */ + +#undef HAS_8BIT_TABLES + +#if defined(CONFIG_DMASOUND_ATARI) || defined(CONFIG_DMASOUND_ATARI_MODULE) ||\ + defined(CONFIG_DMASOUND_PAULA) || defined(CONFIG_DMASOUND_PAULA_MODULE) ||\ + defined(CONFIG_DMASOUND_Q40) || defined(CONFIG_DMASOUND_Q40_MODULE) +#define HAS_8BIT_TABLES +#define MIN_BUFFERS 4 +#define MIN_BUFSIZE (1<<12) /* in bytes (- where does this come from ?) */ +#define MIN_FRAG_SIZE 8 /* not 100% sure about this */ +#define MAX_BUFSIZE (1<<17) /* Limit for Amiga is 128 kb */ +#define MAX_FRAG_SIZE 15 /* allow *4 for mono-8 => stereo-16 (for multi) */ + +#else /* is pmac and multi is off */ + +#define MIN_BUFFERS 2 +#define MIN_BUFSIZE (1<<8) /* in bytes */ +#define MIN_FRAG_SIZE 8 +#define MAX_BUFSIZE (1<<18) /* this is somewhat arbitrary for pmac */ +#define MAX_FRAG_SIZE 16 /* need to allow *4 for mono-8 => stereo-16 */ +#endif + +#define DEFAULT_N_BUFFERS 4 +#define DEFAULT_BUFF_SIZE (1<<15) + + /* + * Initialization + */ + +extern int dmasound_init(void); +#ifdef MODULE +extern void dmasound_deinit(void); +#else +#define dmasound_deinit() do { } while (0) +#endif + +/* description of the set-up applies to either hard or soft settings */ + +typedef struct { + int format; /* AFMT_* */ + int stereo; /* 0 = mono, 1 = stereo */ + int size; /* 8/16 bit*/ + int speed; /* speed */ +} SETTINGS; + + /* + * Machine definitions + */ + +typedef struct { + const char *name; + const char *name2; + struct module *owner; + void *(*dma_alloc)(unsigned int, gfp_t); + void (*dma_free)(void *, unsigned int); + int (*irqinit)(void); +#ifdef MODULE + void (*irqcleanup)(void); +#endif + void (*init)(void); + void (*silence)(void); + int (*setFormat)(int); + int (*setVolume)(int); + int (*setBass)(int); + int (*setTreble)(int); + int (*setGain)(int); + void (*play)(void); + void (*record)(void); /* optional */ + void (*mixer_init)(void); /* optional */ + int (*mixer_ioctl)(u_int, u_long); /* optional */ + int (*write_sq_setup)(void); /* optional */ + int (*read_sq_setup)(void); /* optional */ + int (*sq_open)(fmode_t); /* optional */ + int (*state_info)(char *, size_t); /* optional */ + void (*abort_read)(void); /* optional */ + int min_dsp_speed; + int max_dsp_speed; + int version ; + int hardware_afmts ; /* OSS says we only return h'ware info */ + /* when queried via SNDCTL_DSP_GETFMTS */ + int capabilities ; /* low-level reply to SNDCTL_DSP_GETCAPS */ + SETTINGS default_hard ; /* open() or init() should set something valid */ + SETTINGS default_soft ; /* you can make it look like old OSS, if you want to */ +} MACHINE; + + /* + * Low level stuff + */ + +typedef struct { + ssize_t (*ct_ulaw)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); + ssize_t (*ct_alaw)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); + ssize_t (*ct_s8)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); + ssize_t (*ct_u8)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); + ssize_t (*ct_s16be)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); + ssize_t (*ct_u16be)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); + ssize_t (*ct_s16le)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); + ssize_t (*ct_u16le)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); +} TRANS; + +struct sound_settings { + MACHINE mach; /* machine dependent things */ + SETTINGS hard; /* hardware settings */ + SETTINGS soft; /* software settings */ + SETTINGS dsp; /* /dev/dsp default settings */ + TRANS *trans_write; /* supported translations */ + int volume_left; /* volume (range is machine dependent) */ + int volume_right; + int bass; /* tone (range is machine dependent) */ + int treble; + int gain; + int minDev; /* minor device number currently open */ + spinlock_t lock; +}; + +extern struct sound_settings dmasound; + +#ifdef HAS_8BIT_TABLES +extern char dmasound_ulaw2dma8[]; +extern char dmasound_alaw2dma8[]; +#endif + + /* + * Mid level stuff + */ + +static inline int dmasound_set_volume(int volume) +{ + return dmasound.mach.setVolume(volume); +} + +static inline int dmasound_set_bass(int bass) +{ + return dmasound.mach.setBass ? dmasound.mach.setBass(bass) : 50; +} + +static inline int dmasound_set_treble(int treble) +{ + return dmasound.mach.setTreble ? dmasound.mach.setTreble(treble) : 50; +} + +static inline int dmasound_set_gain(int gain) +{ + return dmasound.mach.setGain ? dmasound.mach.setGain(gain) : 100; +} + + + /* + * Sound queue stuff, the heart of the driver + */ + +struct sound_queue { + /* buffers allocated for this queue */ + int numBufs; /* real limits on what the user can have */ + int bufSize; /* in bytes */ + char **buffers; + + /* current parameters */ + int locked ; /* params cannot be modified when != 0 */ + int user_frags ; /* user requests this many */ + int user_frag_size ; /* of this size */ + int max_count; /* actual # fragments <= numBufs */ + int block_size; /* internal block size in bytes */ + int max_active; /* in-use fragments <= max_count */ + + /* it shouldn't be necessary to declare any of these volatile */ + int front, rear, count; + int rear_size; + /* + * The use of the playing field depends on the hardware + * + * Atari, PMac: The number of frames that are loaded/playing + * + * Amiga: Bit 0 is set: a frame is loaded + * Bit 1 is set: a frame is playing + */ + int active; + wait_queue_head_t action_queue, open_queue, sync_queue; + int non_blocking; + int busy, syncing, xruns, died; +}; + +#define SLEEP(queue) interruptible_sleep_on_timeout(&queue, HZ) +#define WAKE_UP(queue) (wake_up_interruptible(&queue)) + +extern struct sound_queue dmasound_write_sq; +#define write_sq dmasound_write_sq + +extern int dmasound_catchRadius; +#define catchRadius dmasound_catchRadius + +/* define the value to be put in the byte-swap reg in mac-io + when we want it to swap for us. +*/ +#define BS_VAL 1 + +#define SW_INPUT_VOLUME_SCALE 4 +#define SW_INPUT_VOLUME_DEFAULT (128 / SW_INPUT_VOLUME_SCALE) + +extern int expand_read_bal; /* Balance factor for reading */ +extern uint software_input_volume; /* software implemented recording volume! */ + +#endif /* _dmasound_h_ */ diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c new file mode 100644 index 00000000..13c21446 --- /dev/null +++ b/sound/oss/dmasound/dmasound_atari.c @@ -0,0 +1,1620 @@ +/* + * linux/sound/oss/dmasound/dmasound_atari.c + * + * Atari TT and Falcon DMA Sound Driver + * + * See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits + * prior to 28/01/2001 + * + * 28/01/2001 [0.1] Iain Sandoe + * - added versioning + * - put in and populated the hardware_afmts field. + * [0.2] - put in SNDCTL_DSP_GETCAPS value. + * 01/02/2001 [0.3] - put in default hard/soft settings. + */ + + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/soundcard.h> +#include <linux/mm.h> +#include <linux/spinlock.h> +#include <linux/interrupt.h> + +#include <asm/uaccess.h> +#include <asm/atariints.h> +#include <asm/atari_stram.h> + +#include "dmasound.h" + +#define DMASOUND_ATARI_REVISION 0 +#define DMASOUND_ATARI_EDITION 3 + +extern void atari_microwire_cmd(int cmd); + +static int is_falcon; +static int write_sq_ignore_int; /* ++TeSche: used for Falcon */ + +static int expand_bal; /* Balance factor for expanding (not volume!) */ +static int expand_data; /* Data for expanding */ + + +/*** Translations ************************************************************/ + + +/* ++TeSche: radically changed for new expanding purposes... + * + * These two routines now deal with copying/expanding/translating the samples + * from user space into our buffer at the right frequency. They take care about + * how much data there's actually to read, how much buffer space there is and + * to convert samples into the right frequency/encoding. They will only work on + * complete samples so it may happen they leave some bytes in the input stream + * if the user didn't write a multiple of the current sample size. They both + * return the number of bytes they've used from both streams so you may detect + * such a situation. Luckily all programs should be able to cope with that. + * + * I think I've optimized anything as far as one can do in plain C, all + * variables should fit in registers and the loops are really short. There's + * one loop for every possible situation. Writing a more generalized and thus + * parameterized loop would only produce slower code. Feel free to optimize + * this in assembler if you like. :) + * + * I think these routines belong here because they're not yet really hardware + * independent, especially the fact that the Falcon can play 16bit samples + * only in stereo is hardcoded in both of them! + * + * ++geert: split in even more functions (one per format) + */ + +static ssize_t ata_ct_law(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ct_s8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ct_u8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ct_s16be(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ct_u16be(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ct_s16le(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ct_u16le(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ctx_law(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ctx_s8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ctx_u8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ctx_s16be(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ctx_u16be(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ctx_s16le(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); +static ssize_t ata_ctx_u16le(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft); + + +/*** Low level stuff *********************************************************/ + + +static void *AtaAlloc(unsigned int size, gfp_t flags); +static void AtaFree(void *, unsigned int size); +static int AtaIrqInit(void); +#ifdef MODULE +static void AtaIrqCleanUp(void); +#endif /* MODULE */ +static int AtaSetBass(int bass); +static int AtaSetTreble(int treble); +static void TTSilence(void); +static void TTInit(void); +static int TTSetFormat(int format); +static int TTSetVolume(int volume); +static int TTSetGain(int gain); +static void FalconSilence(void); +static void FalconInit(void); +static int FalconSetFormat(int format); +static int FalconSetVolume(int volume); +static void AtaPlayNextFrame(int index); +static void AtaPlay(void); +static irqreturn_t AtaInterrupt(int irq, void *dummy); + +/*** Mid level stuff *********************************************************/ + +static void TTMixerInit(void); +static void FalconMixerInit(void); +static int AtaMixerIoctl(u_int cmd, u_long arg); +static int TTMixerIoctl(u_int cmd, u_long arg); +static int FalconMixerIoctl(u_int cmd, u_long arg); +static int AtaWriteSqSetup(void); +static int AtaSqOpen(fmode_t mode); +static int TTStateInfo(char *buffer, size_t space); +static int FalconStateInfo(char *buffer, size_t space); + + +/*** Translations ************************************************************/ + + +static ssize_t ata_ct_law(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + char *table = dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8 + : dmasound_alaw2dma8; + ssize_t count, used; + u_char *p = &frame[*frameUsed]; + + count = min_t(unsigned long, userCount, frameLeft); + if (dmasound.soft.stereo) + count &= ~1; + used = count; + while (count > 0) { + u_char data; + if (get_user(data, userPtr++)) + return -EFAULT; + *p++ = table[data]; + count--; + } + *frameUsed += used; + return used; +} + + +static ssize_t ata_ct_s8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + void *p = &frame[*frameUsed]; + + count = min_t(unsigned long, userCount, frameLeft); + if (dmasound.soft.stereo) + count &= ~1; + used = count; + if (copy_from_user(p, userPtr, count)) + return -EFAULT; + *frameUsed += used; + return used; +} + + +static ssize_t ata_ct_u8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + + if (!dmasound.soft.stereo) { + u_char *p = &frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft); + used = count; + while (count > 0) { + u_char data; + if (get_user(data, userPtr++)) + return -EFAULT; + *p++ = data ^ 0x80; + count--; + } + } else { + u_short *p = (u_short *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>1; + used = count*2; + while (count > 0) { + u_short data; + if (get_user(data, (u_short __user *)userPtr)) + return -EFAULT; + userPtr += 2; + *p++ = data ^ 0x8080; + count--; + } + } + *frameUsed += used; + return used; +} + + +static ssize_t ata_ct_s16be(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>1; + used = count*2; + while (count > 0) { + u_short data; + if (get_user(data, (u_short __user *)userPtr)) + return -EFAULT; + userPtr += 2; + *p++ = data; + *p++ = data; + count--; + } + *frameUsed += used*2; + } else { + void *p = (u_short *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft) & ~3; + used = count; + if (copy_from_user(p, userPtr, count)) + return -EFAULT; + *frameUsed += used; + } + return used; +} + + +static ssize_t ata_ct_u16be(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>1; + used = count*2; + while (count > 0) { + u_short data; + if (get_user(data, (u_short __user *)userPtr)) + return -EFAULT; + userPtr += 2; + data ^= 0x8000; + *p++ = data; + *p++ = data; + count--; + } + *frameUsed += used*2; + } else { + u_long *p = (u_long *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>2; + used = count*4; + while (count > 0) { + u_int data; + if (get_user(data, (u_int __user *)userPtr)) + return -EFAULT; + userPtr += 4; + *p++ = data ^ 0x80008000; + count--; + } + *frameUsed += used; + } + return used; +} + + +static ssize_t ata_ct_s16le(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + + count = frameLeft; + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>1; + used = count*2; + while (count > 0) { + u_short data; + if (get_user(data, (u_short __user *)userPtr)) + return -EFAULT; + userPtr += 2; + data = le2be16(data); + *p++ = data; + *p++ = data; + count--; + } + *frameUsed += used*2; + } else { + u_long *p = (u_long *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>2; + used = count*4; + while (count > 0) { + u_long data; + if (get_user(data, (u_int __user *)userPtr)) + return -EFAULT; + userPtr += 4; + data = le2be16dbl(data); + *p++ = data; + count--; + } + *frameUsed += used; + } + return used; +} + + +static ssize_t ata_ct_u16le(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + + count = frameLeft; + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>1; + used = count*2; + while (count > 0) { + u_short data; + if (get_user(data, (u_short __user *)userPtr)) + return -EFAULT; + userPtr += 2; + data = le2be16(data) ^ 0x8000; + *p++ = data; + *p++ = data; + } + *frameUsed += used*2; + } else { + u_long *p = (u_long *)&frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft)>>2; + used = count; + while (count > 0) { + u_long data; + if (get_user(data, (u_int __user *)userPtr)) + return -EFAULT; + userPtr += 4; + data = le2be16dbl(data) ^ 0x80008000; + *p++ = data; + count--; + } + *frameUsed += used; + } + return used; +} + + +static ssize_t ata_ctx_law(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + char *table = dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8 + : dmasound_alaw2dma8; + /* this should help gcc to stuff everything into registers */ + long bal = expand_bal; + long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + ssize_t used, usedf; + + used = userCount; + usedf = frameLeft; + if (!dmasound.soft.stereo) { + u_char *p = &frame[*frameUsed]; + u_char data = expand_data; + while (frameLeft) { + u_char c; + if (bal < 0) { + if (!userCount) + break; + if (get_user(c, userPtr++)) + return -EFAULT; + data = table[c]; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + expand_data = data; + } else { + u_short *p = (u_short *)&frame[*frameUsed]; + u_short data = expand_data; + while (frameLeft >= 2) { + u_char c; + if (bal < 0) { + if (userCount < 2) + break; + if (get_user(c, userPtr++)) + return -EFAULT; + data = table[c] << 8; + if (get_user(c, userPtr++)) + return -EFAULT; + data |= table[c]; + userCount -= 2; + bal += hSpeed; + } + *p++ = data; + frameLeft -= 2; + bal -= sSpeed; + } + expand_data = data; + } + expand_bal = bal; + used -= userCount; + *frameUsed += usedf-frameLeft; + return used; +} + + +static ssize_t ata_ctx_s8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + /* this should help gcc to stuff everything into registers */ + long bal = expand_bal; + long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + ssize_t used, usedf; + + used = userCount; + usedf = frameLeft; + if (!dmasound.soft.stereo) { + u_char *p = &frame[*frameUsed]; + u_char data = expand_data; + while (frameLeft) { + if (bal < 0) { + if (!userCount) + break; + if (get_user(data, userPtr++)) + return -EFAULT; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + expand_data = data; + } else { + u_short *p = (u_short *)&frame[*frameUsed]; + u_short data = expand_data; + while (frameLeft >= 2) { + if (bal < 0) { + if (userCount < 2) + break; + if (get_user(data, (u_short __user *)userPtr)) + return -EFAULT; + userPtr += 2; + userCount -= 2; + bal += hSpeed; + } + *p++ = data; + frameLeft -= 2; + bal -= sSpeed; + } + expand_data = data; + } + expand_bal = bal; + used -= userCount; + *frameUsed += usedf-frameLeft; + return used; +} + + +static ssize_t ata_ctx_u8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + /* this should help gcc to stuff everything into registers */ + long bal = expand_bal; + long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + ssize_t used, usedf; + + used = userCount; + usedf = frameLeft; + if (!dmasound.soft.stereo) { + u_char *p = &frame[*frameUsed]; + u_char data = expand_data; + while (frameLeft) { + if (bal < 0) { + if (!userCount) + break; + if (get_user(data, userPtr++)) + return -EFAULT; + data ^= 0x80; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + expand_data = data; + } else { + u_short *p = (u_short *)&frame[*frameUsed]; + u_short data = expand_data; + while (frameLeft >= 2) { + if (bal < 0) { + if (userCount < 2) + break; + if (get_user(data, (u_short __user *)userPtr)) + return -EFAULT; + userPtr += 2; + data ^= 0x8080; + userCount -= 2; + bal += hSpeed; + } + *p++ = data; + frameLeft -= 2; + bal -= sSpeed; + } + expand_data = data; + } + expand_bal = bal; + used -= userCount; + *frameUsed += usedf-frameLeft; + return used; +} + + +static ssize_t ata_ctx_s16be(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + /* this should help gcc to stuff everything into registers */ + long bal = expand_bal; + long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + ssize_t used, usedf; + + used = userCount; + usedf = frameLeft; + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + u_short data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 2) + break; + if (get_user(data, (u_short __user *)userPtr)) + return -EFAULT; + userPtr += 2; + userCount -= 2; + bal += hSpeed; + } + *p++ = data; + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } else { + u_long *p = (u_long *)&frame[*frameUsed]; + u_long data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 4) + break; + if (get_user(data, (u_int __user *)userPtr)) + return -EFAULT; + userPtr += 4; + userCount -= 4; + bal += hSpeed; + } + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } + expand_bal = bal; + used -= userCount; + *frameUsed += usedf-frameLeft; + return used; +} + + +static ssize_t ata_ctx_u16be(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + /* this should help gcc to stuff everything into registers */ + long bal = expand_bal; + long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + ssize_t used, usedf; + + used = userCount; + usedf = frameLeft; + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + u_short data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 2) + break; + if (get_user(data, (u_short __user *)userPtr)) + return -EFAULT; + userPtr += 2; + data ^= 0x8000; + userCount -= 2; + bal += hSpeed; + } + *p++ = data; + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } else { + u_long *p = (u_long *)&frame[*frameUsed]; + u_long data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 4) + break; + if (get_user(data, (u_int __user *)userPtr)) + return -EFAULT; + userPtr += 4; + data ^= 0x80008000; + userCount -= 4; + bal += hSpeed; + } + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } + expand_bal = bal; + used -= userCount; + *frameUsed += usedf-frameLeft; + return used; +} + + +static ssize_t ata_ctx_s16le(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + /* this should help gcc to stuff everything into registers */ + long bal = expand_bal; + long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + ssize_t used, usedf; + + used = userCount; + usedf = frameLeft; + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + u_short data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 2) + break; + if (get_user(data, (u_short __user *)userPtr)) + return -EFAULT; + userPtr += 2; + data = le2be16(data); + userCount -= 2; + bal += hSpeed; + } + *p++ = data; + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } else { + u_long *p = (u_long *)&frame[*frameUsed]; + u_long data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 4) + break; + if (get_user(data, (u_int __user *)userPtr)) + return -EFAULT; + userPtr += 4; + data = le2be16dbl(data); + userCount -= 4; + bal += hSpeed; + } + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } + expand_bal = bal; + used -= userCount; + *frameUsed += usedf-frameLeft; + return used; +} + + +static ssize_t ata_ctx_u16le(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + /* this should help gcc to stuff everything into registers */ + long bal = expand_bal; + long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + ssize_t used, usedf; + + used = userCount; + usedf = frameLeft; + if (!dmasound.soft.stereo) { + u_short *p = (u_short *)&frame[*frameUsed]; + u_short data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 2) + break; + if (get_user(data, (u_short __user *)userPtr)) + return -EFAULT; + userPtr += 2; + data = le2be16(data) ^ 0x8000; + userCount -= 2; + bal += hSpeed; + } + *p++ = data; + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } else { + u_long *p = (u_long *)&frame[*frameUsed]; + u_long data = expand_data; + while (frameLeft >= 4) { + if (bal < 0) { + if (userCount < 4) + break; + if (get_user(data, (u_int __user *)userPtr)) + return -EFAULT; + userPtr += 4; + data = le2be16dbl(data) ^ 0x80008000; + userCount -= 4; + bal += hSpeed; + } + *p++ = data; + frameLeft -= 4; + bal -= sSpeed; + } + expand_data = data; + } + expand_bal = bal; + used -= userCount; + *frameUsed += usedf-frameLeft; + return used; +} + + +static TRANS transTTNormal = { + .ct_ulaw = ata_ct_law, + .ct_alaw = ata_ct_law, + .ct_s8 = ata_ct_s8, + .ct_u8 = ata_ct_u8, +}; + +static TRANS transTTExpanding = { + .ct_ulaw = ata_ctx_law, + .ct_alaw = ata_ctx_law, + .ct_s8 = ata_ctx_s8, + .ct_u8 = ata_ctx_u8, +}; + +static TRANS transFalconNormal = { + .ct_ulaw = ata_ct_law, + .ct_alaw = ata_ct_law, + .ct_s8 = ata_ct_s8, + .ct_u8 = ata_ct_u8, + .ct_s16be = ata_ct_s16be, + .ct_u16be = ata_ct_u16be, + .ct_s16le = ata_ct_s16le, + .ct_u16le = ata_ct_u16le +}; + +static TRANS transFalconExpanding = { + .ct_ulaw = ata_ctx_law, + .ct_alaw = ata_ctx_law, + .ct_s8 = ata_ctx_s8, + .ct_u8 = ata_ctx_u8, + .ct_s16be = ata_ctx_s16be, + .ct_u16be = ata_ctx_u16be, + .ct_s16le = ata_ctx_s16le, + .ct_u16le = ata_ctx_u16le, +}; + + +/*** Low level stuff *********************************************************/ + + + +/* + * Atari (TT/Falcon) + */ + +static void *AtaAlloc(unsigned int size, gfp_t flags) +{ + return atari_stram_alloc(size, "dmasound"); +} + +static void AtaFree(void *obj, unsigned int size) +{ + atari_stram_free( obj ); +} + +static int __init AtaIrqInit(void) +{ + /* Set up timer A. Timer A + will receive a signal upon end of playing from the sound + hardware. Furthermore Timer A is able to count events + and will cause an interrupt after a programmed number + of events. So all we need to keep the music playing is + to provide the sound hardware with new data upon + an interrupt from timer A. */ + st_mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */ + st_mfp.tim_dt_a = 1; /* Cause interrupt after first event. */ + st_mfp.tim_ct_a = 8; /* Turn on event counting. */ + /* Register interrupt handler. */ + if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound", + AtaInterrupt)) + return 0; + st_mfp.int_en_a |= 0x20; /* Turn interrupt on. */ + st_mfp.int_mk_a |= 0x20; + return 1; +} + +#ifdef MODULE +static void AtaIrqCleanUp(void) +{ + st_mfp.tim_ct_a = 0; /* stop timer */ + st_mfp.int_en_a &= ~0x20; /* turn interrupt off */ + free_irq(IRQ_MFP_TIMA, AtaInterrupt); +} +#endif /* MODULE */ + + +#define TONE_VOXWARE_TO_DB(v) \ + (((v) < 0) ? -12 : ((v) > 100) ? 12 : ((v) - 50) * 6 / 25) +#define TONE_DB_TO_VOXWARE(v) (((v) * 25 + ((v) > 0 ? 5 : -5)) / 6 + 50) + + +static int AtaSetBass(int bass) +{ + dmasound.bass = TONE_VOXWARE_TO_DB(bass); + atari_microwire_cmd(MW_LM1992_BASS(dmasound.bass)); + return TONE_DB_TO_VOXWARE(dmasound.bass); +} + + +static int AtaSetTreble(int treble) +{ + dmasound.treble = TONE_VOXWARE_TO_DB(treble); + atari_microwire_cmd(MW_LM1992_TREBLE(dmasound.treble)); + return TONE_DB_TO_VOXWARE(dmasound.treble); +} + + + +/* + * TT + */ + + +static void TTSilence(void) +{ + tt_dmasnd.ctrl = DMASND_CTRL_OFF; + atari_microwire_cmd(MW_LM1992_PSG_HIGH); /* mix in PSG signal 1:1 */ +} + + +static void TTInit(void) +{ + int mode, i, idx; + const int freq[4] = {50066, 25033, 12517, 6258}; + + /* search a frequency that fits into the allowed error range */ + + idx = -1; + for (i = 0; i < ARRAY_SIZE(freq); i++) + /* this isn't as much useful for a TT than for a Falcon, but + * then it doesn't hurt very much to implement it for a TT too. + */ + if ((100 * abs(dmasound.soft.speed - freq[i]) / freq[i]) < catchRadius) + idx = i; + if (idx > -1) { + dmasound.soft.speed = freq[idx]; + dmasound.trans_write = &transTTNormal; + } else + dmasound.trans_write = &transTTExpanding; + + TTSilence(); + dmasound.hard = dmasound.soft; + + if (dmasound.hard.speed > 50066) { + /* we would need to squeeze the sound, but we won't do that */ + dmasound.hard.speed = 50066; + mode = DMASND_MODE_50KHZ; + dmasound.trans_write = &transTTNormal; + } else if (dmasound.hard.speed > 25033) { + dmasound.hard.speed = 50066; + mode = DMASND_MODE_50KHZ; + } else if (dmasound.hard.speed > 12517) { + dmasound.hard.speed = 25033; + mode = DMASND_MODE_25KHZ; + } else if (dmasound.hard.speed > 6258) { + dmasound.hard.speed = 12517; + mode = DMASND_MODE_12KHZ; + } else { + dmasound.hard.speed = 6258; + mode = DMASND_MODE_6KHZ; + } + + tt_dmasnd.mode = (dmasound.hard.stereo ? + DMASND_MODE_STEREO : DMASND_MODE_MONO) | + DMASND_MODE_8BIT | mode; + + expand_bal = -dmasound.soft.speed; +} + + +static int TTSetFormat(int format) +{ + /* TT sound DMA supports only 8bit modes */ + + switch (format) { + case AFMT_QUERY: + return dmasound.soft.format; + case AFMT_MU_LAW: + case AFMT_A_LAW: + case AFMT_S8: + case AFMT_U8: + break; + default: + format = AFMT_S8; + } + + dmasound.soft.format = format; + dmasound.soft.size = 8; + if (dmasound.minDev == SND_DEV_DSP) { + dmasound.dsp.format = format; + dmasound.dsp.size = 8; + } + TTInit(); + + return format; +} + + +#define VOLUME_VOXWARE_TO_DB(v) \ + (((v) < 0) ? -40 : ((v) > 100) ? 0 : ((v) * 2) / 5 - 40) +#define VOLUME_DB_TO_VOXWARE(v) ((((v) + 40) * 5 + 1) / 2) + + +static int TTSetVolume(int volume) +{ + dmasound.volume_left = VOLUME_VOXWARE_TO_DB(volume & 0xff); + atari_microwire_cmd(MW_LM1992_BALLEFT(dmasound.volume_left)); + dmasound.volume_right = VOLUME_VOXWARE_TO_DB((volume & 0xff00) >> 8); + atari_microwire_cmd(MW_LM1992_BALRIGHT(dmasound.volume_right)); + return VOLUME_DB_TO_VOXWARE(dmasound.volume_left) | + (VOLUME_DB_TO_VOXWARE(dmasound.volume_right) << 8); +} + + +#define GAIN_VOXWARE_TO_DB(v) \ + (((v) < 0) ? -80 : ((v) > 100) ? 0 : ((v) * 4) / 5 - 80) +#define GAIN_DB_TO_VOXWARE(v) ((((v) + 80) * 5 + 1) / 4) + +static int TTSetGain(int gain) +{ + dmasound.gain = GAIN_VOXWARE_TO_DB(gain); + atari_microwire_cmd(MW_LM1992_VOLUME(dmasound.gain)); + return GAIN_DB_TO_VOXWARE(dmasound.gain); +} + + + +/* + * Falcon + */ + + +static void FalconSilence(void) +{ + /* stop playback, set sample rate 50kHz for PSG sound */ + tt_dmasnd.ctrl = DMASND_CTRL_OFF; + tt_dmasnd.mode = DMASND_MODE_50KHZ | DMASND_MODE_STEREO | DMASND_MODE_8BIT; + tt_dmasnd.int_div = 0; /* STE compatible divider */ + tt_dmasnd.int_ctrl = 0x0; + tt_dmasnd.cbar_src = 0x0000; /* no matrix inputs */ + tt_dmasnd.cbar_dst = 0x0000; /* no matrix outputs */ + tt_dmasnd.dac_src = 1; /* connect ADC to DAC, disconnect matrix */ + tt_dmasnd.adc_src = 3; /* ADC Input = PSG */ +} + + +static void FalconInit(void) +{ + int divider, i, idx; + const int freq[8] = {49170, 32780, 24585, 19668, 16390, 12292, 9834, 8195}; + + /* search a frequency that fits into the allowed error range */ + + idx = -1; + for (i = 0; i < ARRAY_SIZE(freq); i++) + /* if we will tolerate 3% error 8000Hz->8195Hz (2.38%) would + * be playable without expanding, but that now a kernel runtime + * option + */ + if ((100 * abs(dmasound.soft.speed - freq[i]) / freq[i]) < catchRadius) + idx = i; + if (idx > -1) { + dmasound.soft.speed = freq[idx]; + dmasound.trans_write = &transFalconNormal; + } else + dmasound.trans_write = &transFalconExpanding; + + FalconSilence(); + dmasound.hard = dmasound.soft; + + if (dmasound.hard.size == 16) { + /* the Falcon can play 16bit samples only in stereo */ + dmasound.hard.stereo = 1; + } + + if (dmasound.hard.speed > 49170) { + /* we would need to squeeze the sound, but we won't do that */ + dmasound.hard.speed = 49170; + divider = 1; + dmasound.trans_write = &transFalconNormal; + } else if (dmasound.hard.speed > 32780) { + dmasound.hard.speed = 49170; + divider = 1; + } else if (dmasound.hard.speed > 24585) { + dmasound.hard.speed = 32780; + divider = 2; + } else if (dmasound.hard.speed > 19668) { + dmasound.hard.speed = 24585; + divider = 3; + } else if (dmasound.hard.speed > 16390) { + dmasound.hard.speed = 19668; + divider = 4; + } else if (dmasound.hard.speed > 12292) { + dmasound.hard.speed = 16390; + divider = 5; + } else if (dmasound.hard.speed > 9834) { + dmasound.hard.speed = 12292; + divider = 7; + } else if (dmasound.hard.speed > 8195) { + dmasound.hard.speed = 9834; + divider = 9; + } else { + dmasound.hard.speed = 8195; + divider = 11; + } + tt_dmasnd.int_div = divider; + + /* Setup Falcon sound DMA for playback */ + tt_dmasnd.int_ctrl = 0x4; /* Timer A int at play end */ + tt_dmasnd.track_select = 0x0; /* play 1 track, track 1 */ + tt_dmasnd.cbar_src = 0x0001; /* DMA(25MHz) --> DAC */ + tt_dmasnd.cbar_dst = 0x0000; + tt_dmasnd.rec_track_select = 0; + tt_dmasnd.dac_src = 2; /* connect matrix to DAC */ + tt_dmasnd.adc_src = 0; /* ADC Input = Mic */ + + tt_dmasnd.mode = (dmasound.hard.stereo ? + DMASND_MODE_STEREO : DMASND_MODE_MONO) | + ((dmasound.hard.size == 8) ? + DMASND_MODE_8BIT : DMASND_MODE_16BIT) | + DMASND_MODE_6KHZ; + + expand_bal = -dmasound.soft.speed; +} + + +static int FalconSetFormat(int format) +{ + int size; + /* Falcon sound DMA supports 8bit and 16bit modes */ + + switch (format) { + case AFMT_QUERY: + return dmasound.soft.format; + case AFMT_MU_LAW: + case AFMT_A_LAW: + case AFMT_U8: + case AFMT_S8: + size = 8; + break; + case AFMT_S16_BE: + case AFMT_U16_BE: + case AFMT_S16_LE: + case AFMT_U16_LE: + size = 16; + break; + default: /* :-) */ + size = 8; + format = AFMT_S8; + } + + dmasound.soft.format = format; + dmasound.soft.size = size; + if (dmasound.minDev == SND_DEV_DSP) { + dmasound.dsp.format = format; + dmasound.dsp.size = dmasound.soft.size; + } + + FalconInit(); + + return format; +} + + +/* This is for the Falcon output *attenuation* in 1.5dB steps, + * i.e. output level from 0 to -22.5dB in -1.5dB steps. + */ +#define VOLUME_VOXWARE_TO_ATT(v) \ + ((v) < 0 ? 15 : (v) > 100 ? 0 : 15 - (v) * 3 / 20) +#define VOLUME_ATT_TO_VOXWARE(v) (100 - (v) * 20 / 3) + + +static int FalconSetVolume(int volume) +{ + dmasound.volume_left = VOLUME_VOXWARE_TO_ATT(volume & 0xff); + dmasound.volume_right = VOLUME_VOXWARE_TO_ATT((volume & 0xff00) >> 8); + tt_dmasnd.output_atten = dmasound.volume_left << 8 | dmasound.volume_right << 4; + return VOLUME_ATT_TO_VOXWARE(dmasound.volume_left) | + VOLUME_ATT_TO_VOXWARE(dmasound.volume_right) << 8; +} + + +static void AtaPlayNextFrame(int index) +{ + char *start, *end; + + /* used by AtaPlay() if all doubts whether there really is something + * to be played are already wiped out. + */ + start = write_sq.buffers[write_sq.front]; + end = start+((write_sq.count == index) ? write_sq.rear_size + : write_sq.block_size); + /* end might not be a legal virtual address. */ + DMASNDSetEnd(virt_to_phys(end - 1) + 1); + DMASNDSetBase(virt_to_phys(start)); + /* Since only an even number of samples per frame can + be played, we might lose one byte here. (TO DO) */ + write_sq.front = (write_sq.front+1) % write_sq.max_count; + write_sq.active++; + tt_dmasnd.ctrl = DMASND_CTRL_ON | DMASND_CTRL_REPEAT; +} + + +static void AtaPlay(void) +{ + /* ++TeSche: Note that write_sq.active is no longer just a flag but + * holds the number of frames the DMA is currently programmed for + * instead, may be 0, 1 (currently being played) or 2 (pre-programmed). + * + * Changes done to write_sq.count and write_sq.active are a bit more + * subtle again so now I must admit I also prefer disabling the irq + * here rather than considering all possible situations. But the point + * is that disabling the irq doesn't have any bad influence on this + * version of the driver as we benefit from having pre-programmed the + * DMA wherever possible: There's no need to reload the DMA at the + * exact time of an interrupt but only at some time while the + * pre-programmed frame is playing! + */ + atari_disable_irq(IRQ_MFP_TIMA); + + if (write_sq.active == 2 || /* DMA is 'full' */ + write_sq.count <= 0) { /* nothing to do */ + atari_enable_irq(IRQ_MFP_TIMA); + return; + } + + if (write_sq.active == 0) { + /* looks like there's nothing 'in' the DMA yet, so try + * to put two frames into it (at least one is available). + */ + if (write_sq.count == 1 && + write_sq.rear_size < write_sq.block_size && + !write_sq.syncing) { + /* hmmm, the only existing frame is not + * yet filled and we're not syncing? + */ + atari_enable_irq(IRQ_MFP_TIMA); + return; + } + AtaPlayNextFrame(1); + if (write_sq.count == 1) { + /* no more frames */ + atari_enable_irq(IRQ_MFP_TIMA); + return; + } + if (write_sq.count == 2 && + write_sq.rear_size < write_sq.block_size && + !write_sq.syncing) { + /* hmmm, there were two frames, but the second + * one is not yet filled and we're not syncing? + */ + atari_enable_irq(IRQ_MFP_TIMA); + return; + } + AtaPlayNextFrame(2); + } else { + /* there's already a frame being played so we may only stuff + * one new into the DMA, but even if this may be the last + * frame existing the previous one is still on write_sq.count. + */ + if (write_sq.count == 2 && + write_sq.rear_size < write_sq.block_size && + !write_sq.syncing) { + /* hmmm, the only existing frame is not + * yet filled and we're not syncing? + */ + atari_enable_irq(IRQ_MFP_TIMA); + return; + } + AtaPlayNextFrame(2); + } + atari_enable_irq(IRQ_MFP_TIMA); +} + + +static irqreturn_t AtaInterrupt(int irq, void *dummy) +{ +#if 0 + /* ++TeSche: if you should want to test this... */ + static int cnt; + if (write_sq.active == 2) + if (++cnt == 10) { + /* simulate losing an interrupt */ + cnt = 0; + return IRQ_HANDLED; + } +#endif + spin_lock(&dmasound.lock); + if (write_sq_ignore_int && is_falcon) { + /* ++TeSche: Falcon only: ignore first irq because it comes + * immediately after starting a frame. after that, irqs come + * (almost) like on the TT. + */ + write_sq_ignore_int = 0; + goto out; + } + + if (!write_sq.active) { + /* playing was interrupted and sq_reset() has already cleared + * the sq variables, so better don't do anything here. + */ + WAKE_UP(write_sq.sync_queue); + goto out; + } + + /* Probably ;) one frame is finished. Well, in fact it may be that a + * pre-programmed one is also finished because there has been a long + * delay in interrupt delivery and we've completely lost one, but + * there's no way to detect such a situation. In such a case the last + * frame will be played more than once and the situation will recover + * as soon as the irq gets through. + */ + write_sq.count--; + write_sq.active--; + + if (!write_sq.active) { + tt_dmasnd.ctrl = DMASND_CTRL_OFF; + write_sq_ignore_int = 1; + } + + WAKE_UP(write_sq.action_queue); + /* At least one block of the queue is free now + so wake up a writing process blocked because + of a full queue. */ + + if ((write_sq.active != 1) || (write_sq.count != 1)) + /* We must be a bit carefully here: write_sq.count indicates the + * number of buffers used and not the number of frames to be + * played. If write_sq.count==1 and write_sq.active==1 that + * means the only remaining frame was already programmed + * earlier (and is currently running) so we mustn't call + * AtaPlay() here, otherwise we'll play one frame too much. + */ + AtaPlay(); + + if (!write_sq.active) WAKE_UP(write_sq.sync_queue); + /* We are not playing after AtaPlay(), so there + is nothing to play any more. Wake up a process + waiting for audio output to drain. */ +out: + spin_unlock(&dmasound.lock); + return IRQ_HANDLED; +} + + +/*** Mid level stuff *********************************************************/ + + +/* + * /dev/mixer abstraction + */ + +#define RECLEVEL_VOXWARE_TO_GAIN(v) \ + ((v) < 0 ? 0 : (v) > 100 ? 15 : (v) * 3 / 20) +#define RECLEVEL_GAIN_TO_VOXWARE(v) (((v) * 20 + 2) / 3) + + +static void __init TTMixerInit(void) +{ + atari_microwire_cmd(MW_LM1992_VOLUME(0)); + dmasound.volume_left = 0; + atari_microwire_cmd(MW_LM1992_BALLEFT(0)); + dmasound.volume_right = 0; + atari_microwire_cmd(MW_LM1992_BALRIGHT(0)); + atari_microwire_cmd(MW_LM1992_TREBLE(0)); + atari_microwire_cmd(MW_LM1992_BASS(0)); +} + +static void __init FalconMixerInit(void) +{ + dmasound.volume_left = (tt_dmasnd.output_atten & 0xf00) >> 8; + dmasound.volume_right = (tt_dmasnd.output_atten & 0xf0) >> 4; +} + +static int AtaMixerIoctl(u_int cmd, u_long arg) +{ + int data; + unsigned long flags; + switch (cmd) { + case SOUND_MIXER_READ_SPEAKER: + if (is_falcon || MACH_IS_TT) { + int porta; + spin_lock_irqsave(&dmasound.lock, flags); + sound_ym.rd_data_reg_sel = 14; + porta = sound_ym.rd_data_reg_sel; + spin_unlock_irqrestore(&dmasound.lock, flags); + return IOCTL_OUT(arg, porta & 0x40 ? 0 : 100); + } + break; + case SOUND_MIXER_WRITE_VOLUME: + IOCTL_IN(arg, data); + return IOCTL_OUT(arg, dmasound_set_volume(data)); + case SOUND_MIXER_WRITE_SPEAKER: + if (is_falcon || MACH_IS_TT) { + int porta; + IOCTL_IN(arg, data); + spin_lock_irqsave(&dmasound.lock, flags); + sound_ym.rd_data_reg_sel = 14; + porta = (sound_ym.rd_data_reg_sel & ~0x40) | + (data < 50 ? 0x40 : 0); + sound_ym.wd_data = porta; + spin_unlock_irqrestore(&dmasound.lock, flags); + return IOCTL_OUT(arg, porta & 0x40 ? 0 : 100); + } + } + return -EINVAL; +} + + +static int TTMixerIoctl(u_int cmd, u_long arg) +{ + int data; + switch (cmd) { + case SOUND_MIXER_READ_RECMASK: + return IOCTL_OUT(arg, 0); + case SOUND_MIXER_READ_DEVMASK: + return IOCTL_OUT(arg, + SOUND_MASK_VOLUME | SOUND_MASK_TREBLE | SOUND_MASK_BASS | + (MACH_IS_TT ? SOUND_MASK_SPEAKER : 0)); + case SOUND_MIXER_READ_STEREODEVS: + return IOCTL_OUT(arg, SOUND_MASK_VOLUME); + case SOUND_MIXER_READ_VOLUME: + return IOCTL_OUT(arg, + VOLUME_DB_TO_VOXWARE(dmasound.volume_left) | + (VOLUME_DB_TO_VOXWARE(dmasound.volume_right) << 8)); + case SOUND_MIXER_READ_BASS: + return IOCTL_OUT(arg, TONE_DB_TO_VOXWARE(dmasound.bass)); + case SOUND_MIXER_READ_TREBLE: + return IOCTL_OUT(arg, TONE_DB_TO_VOXWARE(dmasound.treble)); + case SOUND_MIXER_READ_OGAIN: + return IOCTL_OUT(arg, GAIN_DB_TO_VOXWARE(dmasound.gain)); + case SOUND_MIXER_WRITE_BASS: + IOCTL_IN(arg, data); + return IOCTL_OUT(arg, dmasound_set_bass(data)); + case SOUND_MIXER_WRITE_TREBLE: + IOCTL_IN(arg, data); + return IOCTL_OUT(arg, dmasound_set_treble(data)); + case SOUND_MIXER_WRITE_OGAIN: + IOCTL_IN(arg, data); + return IOCTL_OUT(arg, dmasound_set_gain(data)); + } + return AtaMixerIoctl(cmd, arg); +} + +static int FalconMixerIoctl(u_int cmd, u_long arg) +{ + int data; + switch (cmd) { + case SOUND_MIXER_READ_RECMASK: + return IOCTL_OUT(arg, SOUND_MASK_MIC); + case SOUND_MIXER_READ_DEVMASK: + return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_MIC | SOUND_MASK_SPEAKER); + case SOUND_MIXER_READ_STEREODEVS: + return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_MIC); + case SOUND_MIXER_READ_VOLUME: + return IOCTL_OUT(arg, + VOLUME_ATT_TO_VOXWARE(dmasound.volume_left) | + VOLUME_ATT_TO_VOXWARE(dmasound.volume_right) << 8); + case SOUND_MIXER_READ_CAPS: + return IOCTL_OUT(arg, SOUND_CAP_EXCL_INPUT); + case SOUND_MIXER_WRITE_MIC: + IOCTL_IN(arg, data); + tt_dmasnd.input_gain = + RECLEVEL_VOXWARE_TO_GAIN(data & 0xff) << 4 | + RECLEVEL_VOXWARE_TO_GAIN(data >> 8 & 0xff); + /* fall thru, return set value */ + case SOUND_MIXER_READ_MIC: + return IOCTL_OUT(arg, + RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain >> 4 & 0xf) | + RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain & 0xf) << 8); + } + return AtaMixerIoctl(cmd, arg); +} + +static int AtaWriteSqSetup(void) +{ + write_sq_ignore_int = 0; + return 0 ; +} + +static int AtaSqOpen(fmode_t mode) +{ + write_sq_ignore_int = 1; + return 0 ; +} + +static int TTStateInfo(char *buffer, size_t space) +{ + int len = 0; + len += sprintf(buffer+len, "\tvol left %ddB [-40... 0]\n", + dmasound.volume_left); + len += sprintf(buffer+len, "\tvol right %ddB [-40... 0]\n", + dmasound.volume_right); + len += sprintf(buffer+len, "\tbass %ddB [-12...+12]\n", + dmasound.bass); + len += sprintf(buffer+len, "\ttreble %ddB [-12...+12]\n", + dmasound.treble); + if (len >= space) { + printk(KERN_ERR "dmasound_atari: overflowed state buffer alloc.\n") ; + len = space ; + } + return len; +} + +static int FalconStateInfo(char *buffer, size_t space) +{ + int len = 0; + len += sprintf(buffer+len, "\tvol left %ddB [-22.5 ... 0]\n", + dmasound.volume_left); + len += sprintf(buffer+len, "\tvol right %ddB [-22.5 ... 0]\n", + dmasound.volume_right); + if (len >= space) { + printk(KERN_ERR "dmasound_atari: overflowed state buffer alloc.\n") ; + len = space ; + } + return len; +} + + +/*** Machine definitions *****************************************************/ + +static SETTINGS def_hard_falcon = { + .format = AFMT_S8, + .stereo = 0, + .size = 8, + .speed = 8195 +} ; + +static SETTINGS def_hard_tt = { + .format = AFMT_S8, + .stereo = 0, + .size = 8, + .speed = 12517 +} ; + +static SETTINGS def_soft = { + .format = AFMT_U8, + .stereo = 0, + .size = 8, + .speed = 8000 +} ; + +static __initdata MACHINE machTT = { + .name = "Atari", + .name2 = "TT", + .owner = THIS_MODULE, + .dma_alloc = AtaAlloc, + .dma_free = AtaFree, + .irqinit = AtaIrqInit, +#ifdef MODULE + .irqcleanup = AtaIrqCleanUp, +#endif /* MODULE */ + .init = TTInit, + .silence = TTSilence, + .setFormat = TTSetFormat, + .setVolume = TTSetVolume, + .setBass = AtaSetBass, + .setTreble = AtaSetTreble, + .setGain = TTSetGain, + .play = AtaPlay, + .mixer_init = TTMixerInit, + .mixer_ioctl = TTMixerIoctl, + .write_sq_setup = AtaWriteSqSetup, + .sq_open = AtaSqOpen, + .state_info = TTStateInfo, + .min_dsp_speed = 6258, + .version = ((DMASOUND_ATARI_REVISION<<8) | DMASOUND_ATARI_EDITION), + .hardware_afmts = AFMT_S8, /* h'ware-supported formats *only* here */ + .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */ +}; + +static __initdata MACHINE machFalcon = { + .name = "Atari", + .name2 = "FALCON", + .dma_alloc = AtaAlloc, + .dma_free = AtaFree, + .irqinit = AtaIrqInit, +#ifdef MODULE + .irqcleanup = AtaIrqCleanUp, +#endif /* MODULE */ + .init = FalconInit, + .silence = FalconSilence, + .setFormat = FalconSetFormat, + .setVolume = FalconSetVolume, + .setBass = AtaSetBass, + .setTreble = AtaSetTreble, + .play = AtaPlay, + .mixer_init = FalconMixerInit, + .mixer_ioctl = FalconMixerIoctl, + .write_sq_setup = AtaWriteSqSetup, + .sq_open = AtaSqOpen, + .state_info = FalconStateInfo, + .min_dsp_speed = 8195, + .version = ((DMASOUND_ATARI_REVISION<<8) | DMASOUND_ATARI_EDITION), + .hardware_afmts = (AFMT_S8 | AFMT_S16_BE), /* h'ware-supported formats *only* here */ + .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */ +}; + + +/*** Config & Setup **********************************************************/ + + +static int __init dmasound_atari_init(void) +{ + if (MACH_IS_ATARI && ATARIHW_PRESENT(PCM_8BIT)) { + if (ATARIHW_PRESENT(CODEC)) { + dmasound.mach = machFalcon; + dmasound.mach.default_soft = def_soft ; + dmasound.mach.default_hard = def_hard_falcon ; + is_falcon = 1; + } else if (ATARIHW_PRESENT(MICROWIRE)) { + dmasound.mach = machTT; + dmasound.mach.default_soft = def_soft ; + dmasound.mach.default_hard = def_hard_tt ; + is_falcon = 0; + } else + return -ENODEV; + if ((st_mfp.int_en_a & st_mfp.int_mk_a & 0x20) == 0) + return dmasound_init(); + else { + printk("DMA sound driver: Timer A interrupt already in use\n"); + return -EBUSY; + } + } + return -ENODEV; +} + +static void __exit dmasound_atari_cleanup(void) +{ + dmasound_deinit(); +} + +module_init(dmasound_atari_init); +module_exit(dmasound_atari_cleanup); +MODULE_LICENSE("GPL"); diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c new file mode 100644 index 00000000..c918313c --- /dev/null +++ b/sound/oss/dmasound/dmasound_core.c @@ -0,0 +1,1589 @@ +/* + * linux/sound/oss/dmasound/dmasound_core.c + * + * + * OSS/Free compatible Atari TT/Falcon and Amiga DMA sound driver for + * Linux/m68k + * Extended to support Power Macintosh for Linux/ppc by Paul Mackerras + * + * (c) 1995 by Michael Schlueter & Michael Marte + * + * Michael Schlueter (michael@duck.syd.de) did the basic structure of the VFS + * interface and the u-law to signed byte conversion. + * + * Michael Marte (marte@informatik.uni-muenchen.de) did the sound queue, + * /dev/mixer, /dev/sndstat and complemented the VFS interface. He would like + * to thank: + * - Michael Schlueter for initial ideas and documentation on the MFP and + * the DMA sound hardware. + * - Therapy? for their CD 'Troublegum' which really made me rock. + * + * /dev/sndstat is based on code by Hannu Savolainen, the author of the + * VoxWare family of drivers. + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file COPYING in the main directory of this archive + * for more details. + * + * History: + * + * 1995/8/25 First release + * + * 1995/9/02 Roman Hodek: + * - Fixed atari_stram_alloc() call, the timer + * programming and several race conditions + * 1995/9/14 Roman Hodek: + * - After some discussion with Michael Schlueter, + * revised the interrupt disabling + * - Slightly speeded up U8->S8 translation by using + * long operations where possible + * - Added 4:3 interpolation for /dev/audio + * + * 1995/9/20 Torsten Scherer: + * - Fixed a bug in sq_write and changed /dev/audio + * converting to play at 12517Hz instead of 6258Hz. + * + * 1995/9/23 Torsten Scherer: + * - Changed sq_interrupt() and sq_play() to pre-program + * the DMA for another frame while there's still one + * running. This allows the IRQ response to be + * arbitrarily delayed and playing will still continue. + * + * 1995/10/14 Guenther Kelleter, Torsten Scherer: + * - Better support for Falcon audio (the Falcon doesn't + * raise an IRQ at the end of a frame, but at the + * beginning instead!). uses 'if (codec_dma)' in lots + * of places to simply switch between Falcon and TT + * code. + * + * 1995/11/06 Torsten Scherer: + * - Started introducing a hardware abstraction scheme + * (may perhaps also serve for Amigas?) + * - Can now play samples at almost all frequencies by + * means of a more generalized expand routine + * - Takes a good deal of care to cut data only at + * sample sizes + * - Buffer size is now a kernel runtime option + * - Implemented fsync() & several minor improvements + * Guenther Kelleter: + * - Useful hints and bug fixes + * - Cross-checked it for Falcons + * + * 1996/3/9 Geert Uytterhoeven: + * - Support added for Amiga, A-law, 16-bit little + * endian. + * - Unification to drivers/sound/dmasound.c. + * + * 1996/4/6 Martin Mitchell: + * - Updated to 1.3 kernel. + * + * 1996/6/13 Topi Kanerva: + * - Fixed things that were broken (mainly the amiga + * 14-bit routines) + * - /dev/sndstat shows now the real hardware frequency + * - The lowpass filter is disabled by default now + * + * 1996/9/25 Geert Uytterhoeven: + * - Modularization + * + * 1998/6/10 Andreas Schwab: + * - Converted to use sound_core + * + * 1999/12/28 Richard Zidlicky: + * - Added support for Q40 + * + * 2000/2/27 Geert Uytterhoeven: + * - Clean up and split the code into 4 parts: + * o dmasound_core: machine-independent code + * o dmasound_atari: Atari TT and Falcon support + * o dmasound_awacs: Apple PowerMac support + * o dmasound_paula: Amiga support + * + * 2000/3/25 Geert Uytterhoeven: + * - Integration of dmasound_q40 + * - Small clean ups + * + * 2001/01/26 [1.0] Iain Sandoe + * - make /dev/sndstat show revision & edition info. + * - since dmasound.mach.sq_setup() can fail on pmac + * its type has been changed to int and the returns + * are checked. + * [1.1] - stop missing translations from being called. + * 2001/02/08 [1.2] - remove unused translation tables & move machine- + * specific tables to low-level. + * - return correct info. for SNDCTL_DSP_GETFMTS. + * [1.3] - implement SNDCTL_DSP_GETCAPS fully. + * [1.4] - make /dev/sndstat text length usage deterministic. + * - make /dev/sndstat call to low-level + * dmasound.mach.state_info() pass max space to ll driver. + * - tidy startup banners and output info. + * [1.5] - tidy up a little (removed some unused #defines in + * dmasound.h) + * - fix up HAS_RECORD conditionalisation. + * - add record code in places it is missing... + * - change buf-sizes to bytes to allow < 1kb for pmac + * if user param entry is < 256 the value is taken to + * be in kb > 256 is taken to be in bytes. + * - make default buff/frag params conditional on + * machine to allow smaller values for pmac. + * - made the ioctls, read & write comply with the OSS + * rules on setting params. + * - added parsing of _setup() params for record. + * 2001/04/04 [1.6] - fix bug where sample rates higher than maximum were + * being reported as OK. + * - fix open() to return -EBUSY as per OSS doc. when + * audio is in use - this is independent of O_NOBLOCK. + * - fix bug where SNDCTL_DSP_POST was blocking. + */ + + /* Record capability notes 30/01/2001: + * At present these observations apply only to pmac LL driver (the only one + * that can do record, at present). However, if other LL drivers for machines + * with record are added they may apply. + * + * The fragment parameters for the record and play channels are separate. + * However, if the driver is opened O_RDWR there is no way (in the current OSS + * API) to specify their values independently for the record and playback + * channels. Since the only common factor between the input & output is the + * sample rate (on pmac) it should be possible to open /dev/dspX O_WRONLY and + * /dev/dspY O_RDONLY. The input & output channels could then have different + * characteristics (other than the first that sets sample rate claiming the + * right to set it for ever). As it stands, the format, channels, number of + * bits & sample rate are assumed to be common. In the future perhaps these + * should be the responsibility of the LL driver - and then if a card really + * does not share items between record & playback they can be specified + * separately. +*/ + +/* Thread-safeness of shared_resources notes: 31/01/2001 + * If the user opens O_RDWR and then splits record & play between two threads + * both of which inherit the fd - and then starts changing things from both + * - we will have difficulty telling. + * + * It's bad application coding - but ... + * TODO: think about how to sort this out... without bogging everything down in + * semaphores. + * + * Similarly, the OSS spec says "all changes to parameters must be between + * open() and the first read() or write(). - and a bit later on (by + * implication) "between SNDCTL_DSP_RESET and the first read() or write() after + * it". If the app is multi-threaded and this rule is broken between threads + * we will have trouble spotting it - and the fault will be rather obscure :-( + * + * We will try and put out at least a kmsg if we see it happen... but I think + * it will be quite hard to trap it with an -EXXX return... because we can't + * see the fault until after the damage is done. +*/ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/sound.h> +#include <linux/init.h> +#include <linux/soundcard.h> +#include <linux/poll.h> +#include <linux/mutex.h> + +#include <asm/uaccess.h> + +#include "dmasound.h" + +#define DMASOUND_CORE_REVISION 1 +#define DMASOUND_CORE_EDITION 6 + + /* + * Declarations + */ + +static DEFINE_MUTEX(dmasound_core_mutex); +int dmasound_catchRadius = 0; +module_param(dmasound_catchRadius, int, 0); + +static unsigned int numWriteBufs = DEFAULT_N_BUFFERS; +module_param(numWriteBufs, int, 0); +static unsigned int writeBufSize = DEFAULT_BUFF_SIZE ; /* in bytes */ +module_param(writeBufSize, int, 0); + +MODULE_LICENSE("GPL"); + +#ifdef MODULE +static int sq_unit = -1; +static int mixer_unit = -1; +static int state_unit = -1; +static int irq_installed; +#endif /* MODULE */ + +/* control over who can modify resources shared between play/record */ +static fmode_t shared_resource_owner; +static int shared_resources_initialised; + + /* + * Mid level stuff + */ + +struct sound_settings dmasound = { + .lock = __SPIN_LOCK_UNLOCKED(dmasound.lock) +}; + +static inline void sound_silence(void) +{ + dmasound.mach.silence(); /* _MUST_ stop DMA */ +} + +static inline int sound_set_format(int format) +{ + return dmasound.mach.setFormat(format); +} + + +static int sound_set_speed(int speed) +{ + if (speed < 0) + return dmasound.soft.speed; + + /* trap out-of-range speed settings. + at present we allow (arbitrarily) low rates - using soft + up-conversion - but we can't allow > max because there is + no soft down-conversion. + */ + if (dmasound.mach.max_dsp_speed && + (speed > dmasound.mach.max_dsp_speed)) + speed = dmasound.mach.max_dsp_speed ; + + dmasound.soft.speed = speed; + + if (dmasound.minDev == SND_DEV_DSP) + dmasound.dsp.speed = dmasound.soft.speed; + + return dmasound.soft.speed; +} + +static int sound_set_stereo(int stereo) +{ + if (stereo < 0) + return dmasound.soft.stereo; + + stereo = !!stereo; /* should be 0 or 1 now */ + + dmasound.soft.stereo = stereo; + if (dmasound.minDev == SND_DEV_DSP) + dmasound.dsp.stereo = stereo; + + return stereo; +} + +static ssize_t sound_copy_translate(TRANS *trans, const u_char __user *userPtr, + size_t userCount, u_char frame[], + ssize_t *frameUsed, ssize_t frameLeft) +{ + ssize_t (*ct_func)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t); + + switch (dmasound.soft.format) { + case AFMT_MU_LAW: + ct_func = trans->ct_ulaw; + break; + case AFMT_A_LAW: + ct_func = trans->ct_alaw; + break; + case AFMT_S8: + ct_func = trans->ct_s8; + break; + case AFMT_U8: + ct_func = trans->ct_u8; + break; + case AFMT_S16_BE: + ct_func = trans->ct_s16be; + break; + case AFMT_U16_BE: + ct_func = trans->ct_u16be; + break; + case AFMT_S16_LE: + ct_func = trans->ct_s16le; + break; + case AFMT_U16_LE: + ct_func = trans->ct_u16le; + break; + default: + return 0; + } + /* if the user has requested a non-existent translation don't try + to call it but just return 0 bytes moved + */ + if (ct_func) + return ct_func(userPtr, userCount, frame, frameUsed, frameLeft); + return 0; +} + + /* + * /dev/mixer abstraction + */ + +static struct { + int busy; + int modify_counter; +} mixer; + +static int mixer_open(struct inode *inode, struct file *file) +{ + mutex_lock(&dmasound_core_mutex); + if (!try_module_get(dmasound.mach.owner)) { + mutex_unlock(&dmasound_core_mutex); + return -ENODEV; + } + mixer.busy = 1; + mutex_unlock(&dmasound_core_mutex); + return 0; +} + +static int mixer_release(struct inode *inode, struct file *file) +{ + mutex_lock(&dmasound_core_mutex); + mixer.busy = 0; + module_put(dmasound.mach.owner); + mutex_unlock(&dmasound_core_mutex); + return 0; +} + +static int mixer_ioctl(struct file *file, u_int cmd, u_long arg) +{ + if (_SIOC_DIR(cmd) & _SIOC_WRITE) + mixer.modify_counter++; + switch (cmd) { + case OSS_GETVERSION: + return IOCTL_OUT(arg, SOUND_VERSION); + case SOUND_MIXER_INFO: + { + mixer_info info; + memset(&info, 0, sizeof(info)); + strlcpy(info.id, dmasound.mach.name2, sizeof(info.id)); + strlcpy(info.name, dmasound.mach.name2, sizeof(info.name)); + info.modify_counter = mixer.modify_counter; + if (copy_to_user((void __user *)arg, &info, sizeof(info))) + return -EFAULT; + return 0; + } + } + if (dmasound.mach.mixer_ioctl) + return dmasound.mach.mixer_ioctl(cmd, arg); + return -EINVAL; +} + +static long mixer_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) +{ + int ret; + + mutex_lock(&dmasound_core_mutex); + ret = mixer_ioctl(file, cmd, arg); + mutex_unlock(&dmasound_core_mutex); + + return ret; +} + +static const struct file_operations mixer_fops = +{ + .owner = THIS_MODULE, + .llseek = no_llseek, + .unlocked_ioctl = mixer_unlocked_ioctl, + .open = mixer_open, + .release = mixer_release, +}; + +static void mixer_init(void) +{ +#ifndef MODULE + int mixer_unit; +#endif + mixer_unit = register_sound_mixer(&mixer_fops, -1); + if (mixer_unit < 0) + return; + + mixer.busy = 0; + dmasound.treble = 0; + dmasound.bass = 0; + if (dmasound.mach.mixer_init) + dmasound.mach.mixer_init(); +} + + + /* + * Sound queue stuff, the heart of the driver + */ + +struct sound_queue dmasound_write_sq; +static void sq_reset_output(void) ; + +static int sq_allocate_buffers(struct sound_queue *sq, int num, int size) +{ + int i; + + if (sq->buffers) + return 0; + sq->numBufs = num; + sq->bufSize = size; + sq->buffers = kmalloc (num * sizeof(char *), GFP_KERNEL); + if (!sq->buffers) + return -ENOMEM; + for (i = 0; i < num; i++) { + sq->buffers[i] = dmasound.mach.dma_alloc(size, GFP_KERNEL); + if (!sq->buffers[i]) { + while (i--) + dmasound.mach.dma_free(sq->buffers[i], size); + kfree(sq->buffers); + sq->buffers = NULL; + return -ENOMEM; + } + } + return 0; +} + +static void sq_release_buffers(struct sound_queue *sq) +{ + int i; + + if (sq->buffers) { + for (i = 0; i < sq->numBufs; i++) + dmasound.mach.dma_free(sq->buffers[i], sq->bufSize); + kfree(sq->buffers); + sq->buffers = NULL; + } +} + + +static int sq_setup(struct sound_queue *sq) +{ + int (*setup_func)(void) = NULL; + int hard_frame ; + + if (sq->locked) { /* are we already set? - and not changeable */ +#ifdef DEBUG_DMASOUND +printk("dmasound_core: tried to sq_setup a locked queue\n") ; +#endif + return -EINVAL ; + } + sq->locked = 1 ; /* don't think we have a race prob. here _check_ */ + + /* make sure that the parameters are set up + This should have been done already... + */ + + dmasound.mach.init(); + + /* OK. If the user has set fragment parameters explicitly, then we + should leave them alone... as long as they are valid. + Invalid user fragment params can occur if we allow the whole buffer + to be used when the user requests the fragments sizes (with no soft + x-lation) and then the user subsequently sets a soft x-lation that + requires increased internal buffering. + + Othwerwise (if the user did not set them) OSS says that we should + select frag params on the basis of 0.5 s output & 0.1 s input + latency. (TODO. For now we will copy in the defaults.) + */ + + if (sq->user_frags <= 0) { + sq->max_count = sq->numBufs ; + sq->max_active = sq->numBufs ; + sq->block_size = sq->bufSize; + /* set up the user info */ + sq->user_frags = sq->numBufs ; + sq->user_frag_size = sq->bufSize ; + sq->user_frag_size *= + (dmasound.soft.size * (dmasound.soft.stereo+1) ) ; + sq->user_frag_size /= + (dmasound.hard.size * (dmasound.hard.stereo+1) ) ; + } else { + /* work out requested block size */ + sq->block_size = sq->user_frag_size ; + sq->block_size *= + (dmasound.hard.size * (dmasound.hard.stereo+1) ) ; + sq->block_size /= + (dmasound.soft.size * (dmasound.soft.stereo+1) ) ; + /* the user wants to write frag-size chunks */ + sq->block_size *= dmasound.hard.speed ; + sq->block_size /= dmasound.soft.speed ; + /* this only works for size values which are powers of 2 */ + hard_frame = + (dmasound.hard.size * (dmasound.hard.stereo+1))/8 ; + sq->block_size += (hard_frame - 1) ; + sq->block_size &= ~(hard_frame - 1) ; /* make sure we are aligned */ + /* let's just check for obvious mistakes */ + if ( sq->block_size <= 0 || sq->block_size > sq->bufSize) { +#ifdef DEBUG_DMASOUND +printk("dmasound_core: invalid frag size (user set %d)\n", sq->user_frag_size) ; +#endif + sq->block_size = sq->bufSize ; + } + if ( sq->user_frags <= sq->numBufs ) { + sq->max_count = sq->user_frags ; + /* if user has set max_active - then use it */ + sq->max_active = (sq->max_active <= sq->max_count) ? + sq->max_active : sq->max_count ; + } else { +#ifdef DEBUG_DMASOUND +printk("dmasound_core: invalid frag count (user set %d)\n", sq->user_frags) ; +#endif + sq->max_count = + sq->max_active = sq->numBufs ; + } + } + sq->front = sq->count = sq->rear_size = 0; + sq->syncing = 0; + sq->active = 0; + + if (sq == &write_sq) { + sq->rear = -1; + setup_func = dmasound.mach.write_sq_setup; + } + if (setup_func) + return setup_func(); + return 0 ; +} + +static inline void sq_play(void) +{ + dmasound.mach.play(); +} + +static ssize_t sq_write(struct file *file, const char __user *src, size_t uLeft, + loff_t *ppos) +{ + ssize_t uWritten = 0; + u_char *dest; + ssize_t uUsed = 0, bUsed, bLeft; + unsigned long flags ; + + /* ++TeSche: Is something like this necessary? + * Hey, that's an honest question! Or does any other part of the + * filesystem already checks this situation? I really don't know. + */ + if (uLeft == 0) + return 0; + + /* implement any changes we have made to the soft/hard params. + this is not satisfactory really, all we have done up to now is to + say what we would like - there hasn't been any real checking of capability + */ + + if (shared_resources_initialised == 0) { + dmasound.mach.init() ; + shared_resources_initialised = 1 ; + } + + /* set up the sq if it is not already done. This may seem a dumb place + to do it - but it is what OSS requires. It means that write() can + return memory allocation errors. To avoid this possibility use the + GETBLKSIZE or GETOSPACE ioctls (after you've fiddled with all the + params you want to change) - these ioctls also force the setup. + */ + + if (write_sq.locked == 0) { + if ((uWritten = sq_setup(&write_sq)) < 0) return uWritten ; + uWritten = 0 ; + } + +/* FIXME: I think that this may be the wrong behaviour when we get strapped + for time and the cpu is close to being (or actually) behind in sending data. + - because we've lost the time that the N samples, already in the buffer, + would have given us to get here with the next lot from the user. +*/ + /* The interrupt doesn't start to play the last, incomplete frame. + * Thus we can append to it without disabling the interrupts! (Note + * also that write_sq.rear isn't affected by the interrupt.) + */ + + /* as of 1.6 this behaviour changes if SNDCTL_DSP_POST has been issued: + this will mimic the behaviour of syncing and allow the sq_play() to + queue a partial fragment. Since sq_play() may/will be called from + the IRQ handler - at least on Pmac we have to deal with it. + The strategy - possibly not optimum - is to kill _POST status if we + get here. This seems, at least, reasonable - in the sense that POST + is supposed to indicate that we might not write before the queue + is drained - and if we get here in time then it does not apply. + */ + + spin_lock_irqsave(&dmasound.lock, flags); + write_sq.syncing &= ~2 ; /* take out POST status */ + spin_unlock_irqrestore(&dmasound.lock, flags); + + if (write_sq.count > 0 && + (bLeft = write_sq.block_size-write_sq.rear_size) > 0) { + dest = write_sq.buffers[write_sq.rear]; + bUsed = write_sq.rear_size; + uUsed = sound_copy_translate(dmasound.trans_write, src, uLeft, + dest, &bUsed, bLeft); + if (uUsed <= 0) + return uUsed; + src += uUsed; + uWritten += uUsed; + uLeft = (uUsed <= uLeft) ? (uLeft - uUsed) : 0 ; /* paranoia */ + write_sq.rear_size = bUsed; + } + + while (uLeft) { + while (write_sq.count >= write_sq.max_active) { + sq_play(); + if (write_sq.non_blocking) + return uWritten > 0 ? uWritten : -EAGAIN; + SLEEP(write_sq.action_queue); + if (signal_pending(current)) + return uWritten > 0 ? uWritten : -EINTR; + } + + /* Here, we can avoid disabling the interrupt by first + * copying and translating the data, and then updating + * the write_sq variables. Until this is done, the interrupt + * won't see the new frame and we can work on it + * undisturbed. + */ + + dest = write_sq.buffers[(write_sq.rear+1) % write_sq.max_count]; + bUsed = 0; + bLeft = write_sq.block_size; + uUsed = sound_copy_translate(dmasound.trans_write, src, uLeft, + dest, &bUsed, bLeft); + if (uUsed <= 0) + break; + src += uUsed; + uWritten += uUsed; + uLeft = (uUsed <= uLeft) ? (uLeft - uUsed) : 0 ; /* paranoia */ + if (bUsed) { + write_sq.rear = (write_sq.rear+1) % write_sq.max_count; + write_sq.rear_size = bUsed; + write_sq.count++; + } + } /* uUsed may have been 0 */ + + sq_play(); + + return uUsed < 0? uUsed: uWritten; +} + +static unsigned int sq_poll(struct file *file, struct poll_table_struct *wait) +{ + unsigned int mask = 0; + int retVal; + + if (write_sq.locked == 0) { + if ((retVal = sq_setup(&write_sq)) < 0) + return retVal; + return 0; + } + if (file->f_mode & FMODE_WRITE ) + poll_wait(file, &write_sq.action_queue, wait); + if (file->f_mode & FMODE_WRITE) + if (write_sq.count < write_sq.max_active || write_sq.block_size - write_sq.rear_size > 0) + mask |= POLLOUT | POLLWRNORM; + return mask; + +} + +static inline void sq_init_waitqueue(struct sound_queue *sq) +{ + init_waitqueue_head(&sq->action_queue); + init_waitqueue_head(&sq->open_queue); + init_waitqueue_head(&sq->sync_queue); + sq->busy = 0; +} + +#if 0 /* blocking open() */ +static inline void sq_wake_up(struct sound_queue *sq, struct file *file, + fmode_t mode) +{ + if (file->f_mode & mode) { + sq->busy = 0; /* CHECK: IS THIS OK??? */ + WAKE_UP(sq->open_queue); + } +} +#endif + +static int sq_open2(struct sound_queue *sq, struct file *file, fmode_t mode, + int numbufs, int bufsize) +{ + int rc = 0; + + if (file->f_mode & mode) { + if (sq->busy) { +#if 0 /* blocking open() */ + rc = -EBUSY; + if (file->f_flags & O_NONBLOCK) + return rc; + rc = -EINTR; + while (sq->busy) { + SLEEP(sq->open_queue); + if (signal_pending(current)) + return rc; + } + rc = 0; +#else + /* OSS manual says we will return EBUSY regardless + of O_NOBLOCK. + */ + return -EBUSY ; +#endif + } + sq->busy = 1; /* Let's play spot-the-race-condition */ + + /* allocate the default number & size of buffers. + (i.e. specified in _setup() or as module params) + can't be changed at the moment - but _could_ be perhaps + in the setfragments ioctl. + */ + if (( rc = sq_allocate_buffers(sq, numbufs, bufsize))) { +#if 0 /* blocking open() */ + sq_wake_up(sq, file, mode); +#else + sq->busy = 0 ; +#endif + return rc; + } + + sq->non_blocking = file->f_flags & O_NONBLOCK; + } + return rc; +} + +#define write_sq_init_waitqueue() sq_init_waitqueue(&write_sq) +#if 0 /* blocking open() */ +#define write_sq_wake_up(file) sq_wake_up(&write_sq, file, FMODE_WRITE) +#endif +#define write_sq_release_buffers() sq_release_buffers(&write_sq) +#define write_sq_open(file) \ + sq_open2(&write_sq, file, FMODE_WRITE, numWriteBufs, writeBufSize ) + +static int sq_open(struct inode *inode, struct file *file) +{ + int rc; + + mutex_lock(&dmasound_core_mutex); + if (!try_module_get(dmasound.mach.owner)) { + mutex_unlock(&dmasound_core_mutex); + return -ENODEV; + } + + rc = write_sq_open(file); /* checks the f_mode */ + if (rc) + goto out; + if (file->f_mode & FMODE_READ) { + /* TODO: if O_RDWR, release any resources grabbed by write part */ + rc = -ENXIO ; /* I think this is what is required by open(2) */ + goto out; + } + + if (dmasound.mach.sq_open) + dmasound.mach.sq_open(file->f_mode); + + /* CHECK whether this is sensible - in the case that dsp0 could be opened + O_RDONLY and dsp1 could be opened O_WRONLY + */ + + dmasound.minDev = iminor(inode) & 0x0f; + + /* OK. - we should make some attempt at consistency. At least the H'ware + options should be set with a valid mode. We will make it that the LL + driver must supply defaults for hard & soft params. + */ + + if (shared_resource_owner == 0) { + /* you can make this AFMT_U8/mono/8K if you want to mimic old + OSS behaviour - while we still have soft translations ;-) */ + dmasound.soft = dmasound.mach.default_soft ; + dmasound.dsp = dmasound.mach.default_soft ; + dmasound.hard = dmasound.mach.default_hard ; + } + +#ifndef DMASOUND_STRICT_OSS_COMPLIANCE + /* none of the current LL drivers can actually do this "native" at the moment + OSS does not really require us to supply /dev/audio if we can't do it. + */ + if (dmasound.minDev == SND_DEV_AUDIO) { + sound_set_speed(8000); + sound_set_stereo(0); + sound_set_format(AFMT_MU_LAW); + } +#endif + mutex_unlock(&dmasound_core_mutex); + return 0; + out: + module_put(dmasound.mach.owner); + mutex_unlock(&dmasound_core_mutex); + return rc; +} + +static void sq_reset_output(void) +{ + sound_silence(); /* this _must_ stop DMA, we might be about to lose the buffers */ + write_sq.active = 0; + write_sq.count = 0; + write_sq.rear_size = 0; + /* write_sq.front = (write_sq.rear+1) % write_sq.max_count;*/ + write_sq.front = 0 ; + write_sq.rear = -1 ; /* same as for set-up */ + + /* OK - we can unlock the parameters and fragment settings */ + write_sq.locked = 0 ; + write_sq.user_frags = 0 ; + write_sq.user_frag_size = 0 ; +} + +static void sq_reset(void) +{ + sq_reset_output() ; + /* we could consider resetting the shared_resources_owner here... but I + think it is probably still rather non-obvious to application writer + */ + + /* we release everything else though */ + shared_resources_initialised = 0 ; +} + +static int sq_fsync(struct file *filp, struct dentry *dentry) +{ + int rc = 0; + int timeout = 5; + + write_sq.syncing |= 1; + sq_play(); /* there may be an incomplete frame waiting */ + + while (write_sq.active) { + SLEEP(write_sq.sync_queue); + if (signal_pending(current)) { + /* While waiting for audio output to drain, an + * interrupt occurred. Stop audio output immediately + * and clear the queue. */ + sq_reset_output(); + rc = -EINTR; + break; + } + if (!--timeout) { + printk(KERN_WARNING "dmasound: Timeout draining output\n"); + sq_reset_output(); + rc = -EIO; + break; + } + } + + /* flag no sync regardless of whether we had a DSP_POST or not */ + write_sq.syncing = 0 ; + return rc; +} + +static int sq_release(struct inode *inode, struct file *file) +{ + int rc = 0; + + mutex_lock(&dmasound_core_mutex); + + if (file->f_mode & FMODE_WRITE) { + if (write_sq.busy) + rc = sq_fsync(file, file->f_path.dentry); + + sq_reset_output() ; /* make sure dma is stopped and all is quiet */ + write_sq_release_buffers(); + write_sq.busy = 0; + } + + if (file->f_mode & shared_resource_owner) { /* it's us that has them */ + shared_resource_owner = 0 ; + shared_resources_initialised = 0 ; + dmasound.hard = dmasound.mach.default_hard ; + } + + module_put(dmasound.mach.owner); + +#if 0 /* blocking open() */ + /* Wake up a process waiting for the queue being released. + * Note: There may be several processes waiting for a call + * to open() returning. */ + + /* Iain: hmm I don't understand this next comment ... */ + /* There is probably a DOS atack here. They change the mode flag. */ + /* XXX add check here,*/ + read_sq_wake_up(file); /* checks f_mode */ + write_sq_wake_up(file); /* checks f_mode */ +#endif /* blocking open() */ + + mutex_unlock(&dmasound_core_mutex); + + return rc; +} + +/* here we see if we have a right to modify format, channels, size and so on + if no-one else has claimed it already then we do... + + TODO: We might change this to mask O_RDWR such that only one or the other channel + is the owner - if we have problems. +*/ + +static int shared_resources_are_mine(fmode_t md) +{ + if (shared_resource_owner) + return (shared_resource_owner & md) != 0; + else { + shared_resource_owner = md ; + return 1 ; + } +} + +/* if either queue is locked we must deny the right to change shared params +*/ + +static int queues_are_quiescent(void) +{ + if (write_sq.locked) + return 0 ; + return 1 ; +} + +/* check and set a queue's fragments per user's wishes... + we will check against the pre-defined literals and the actual sizes. + This is a bit fraught - because soft translations can mess with our + buffer requirements *after* this call - OSS says "call setfrags first" +*/ + +/* It is possible to replace all the -EINVAL returns with an override that + just puts the allowable value in. This may be what many OSS apps require +*/ + +static int set_queue_frags(struct sound_queue *sq, int bufs, int size) +{ + if (sq->locked) { +#ifdef DEBUG_DMASOUND +printk("dmasound_core: tried to set_queue_frags on a locked queue\n") ; +#endif + return -EINVAL ; + } + + if ((size < MIN_FRAG_SIZE) || (size > MAX_FRAG_SIZE)) + return -EINVAL ; + size = (1<<size) ; /* now in bytes */ + if (size > sq->bufSize) + return -EINVAL ; /* this might still not work */ + + if (bufs <= 0) + return -EINVAL ; + if (bufs > sq->numBufs) /* the user is allowed say "don't care" with 0x7fff */ + bufs = sq->numBufs ; + + /* there is, currently, no way to specify max_active separately + from max_count. This could be a LL driver issue - I guess + if there is a requirement for these values to be different then + we will have to pass that info. up to this level. + */ + sq->user_frags = + sq->max_active = bufs ; + sq->user_frag_size = size ; + + return 0 ; +} + +static int sq_ioctl(struct file *file, u_int cmd, u_long arg) +{ + int val, result; + u_long fmt; + int data; + int size, nbufs; + audio_buf_info info; + + switch (cmd) { + case SNDCTL_DSP_RESET: + sq_reset(); + return 0; + break ; + case SNDCTL_DSP_GETFMTS: + fmt = dmasound.mach.hardware_afmts ; /* this is what OSS says.. */ + return IOCTL_OUT(arg, fmt); + break ; + case SNDCTL_DSP_GETBLKSIZE: + /* this should tell the caller about bytes that the app can + read/write - the app doesn't care about our internal buffers. + We force sq_setup() here as per OSS 1.1 (which should + compute the values necessary). + Since there is no mechanism to specify read/write separately, for + fds opened O_RDWR, the write_sq values will, arbitrarily, overwrite + the read_sq ones. + */ + size = 0 ; + if (file->f_mode & FMODE_WRITE) { + if ( !write_sq.locked ) + sq_setup(&write_sq) ; + size = write_sq.user_frag_size ; + } + return IOCTL_OUT(arg, size); + break ; + case SNDCTL_DSP_POST: + /* all we are going to do is to tell the LL that any + partial frags can be queued for output. + The LL will have to clear this flag when last output + is queued. + */ + write_sq.syncing |= 0x2 ; + sq_play() ; + return 0 ; + case SNDCTL_DSP_SYNC: + /* This call, effectively, has the same behaviour as SNDCTL_DSP_RESET + except that it waits for output to finish before resetting + everything - read, however, is killed immediately. + */ + result = 0 ; + if (file->f_mode & FMODE_WRITE) { + result = sq_fsync(file, file->f_path.dentry); + sq_reset_output() ; + } + /* if we are the shared resource owner then release them */ + if (file->f_mode & shared_resource_owner) + shared_resources_initialised = 0 ; + return result ; + break ; + case SOUND_PCM_READ_RATE: + return IOCTL_OUT(arg, dmasound.soft.speed); + case SNDCTL_DSP_SPEED: + /* changing this on the fly will have weird effects on the sound. + Where there are rate conversions implemented in soft form - it + will cause the _ctx_xxx() functions to be substituted. + However, there doesn't appear to be any reason to dis-allow it from + a driver pov. + */ + if (shared_resources_are_mine(file->f_mode)) { + IOCTL_IN(arg, data); + data = sound_set_speed(data) ; + shared_resources_initialised = 0 ; + return IOCTL_OUT(arg, data); + } else + return -EINVAL ; + break ; + /* OSS says these next 4 actions are undefined when the device is + busy/active - we will just return -EINVAL. + To be allowed to change one - (a) you have to own the right + (b) the queue(s) must be quiescent + */ + case SNDCTL_DSP_STEREO: + if (shared_resources_are_mine(file->f_mode) && + queues_are_quiescent()) { + IOCTL_IN(arg, data); + shared_resources_initialised = 0 ; + return IOCTL_OUT(arg, sound_set_stereo(data)); + } else + return -EINVAL ; + break ; + case SOUND_PCM_WRITE_CHANNELS: + if (shared_resources_are_mine(file->f_mode) && + queues_are_quiescent()) { + IOCTL_IN(arg, data); + /* the user might ask for 20 channels, we will return 1 or 2 */ + shared_resources_initialised = 0 ; + return IOCTL_OUT(arg, sound_set_stereo(data-1)+1); + } else + return -EINVAL ; + break ; + case SNDCTL_DSP_SETFMT: + if (shared_resources_are_mine(file->f_mode) && + queues_are_quiescent()) { + int format; + IOCTL_IN(arg, data); + shared_resources_initialised = 0 ; + format = sound_set_format(data); + result = IOCTL_OUT(arg, format); + if (result < 0) + return result; + if (format != data && data != AFMT_QUERY) + return -EINVAL; + return 0; + } else + return -EINVAL ; + case SNDCTL_DSP_SUBDIVIDE: + return -EINVAL ; + case SNDCTL_DSP_SETFRAGMENT: + /* we can do this independently for the two queues - with the + proviso that for fds opened O_RDWR we cannot separate the + actions and both queues will be set per the last call. + NOTE: this does *NOT* actually set the queue up - merely + registers our intentions. + */ + IOCTL_IN(arg, data); + result = 0 ; + nbufs = (data >> 16) & 0x7fff ; /* 0x7fff is 'use maximum' */ + size = data & 0xffff; + if (file->f_mode & FMODE_WRITE) { + result = set_queue_frags(&write_sq, nbufs, size) ; + if (result) + return result ; + } + /* NOTE: this return value is irrelevant - OSS specifically says that + the value is 'random' and that the user _must_ check the actual + frags values using SNDCTL_DSP_GETBLKSIZE or similar */ + return IOCTL_OUT(arg, data); + break ; + case SNDCTL_DSP_GETOSPACE: + /* + */ + if (file->f_mode & FMODE_WRITE) { + if ( !write_sq.locked ) + sq_setup(&write_sq) ; + info.fragments = write_sq.max_active - write_sq.count; + info.fragstotal = write_sq.max_active; + info.fragsize = write_sq.user_frag_size; + info.bytes = info.fragments * info.fragsize; + if (copy_to_user((void __user *)arg, &info, sizeof(info))) + return -EFAULT; + return 0; + } else + return -EINVAL ; + break ; + case SNDCTL_DSP_GETCAPS: + val = dmasound.mach.capabilities & 0xffffff00; + return IOCTL_OUT(arg,val); + + default: + return mixer_ioctl(file, cmd, arg); + } + return -EINVAL; +} + +static long sq_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) +{ + int ret; + + mutex_lock(&dmasound_core_mutex); + ret = sq_ioctl(file, cmd, arg); + mutex_unlock(&dmasound_core_mutex); + + return ret; +} + +static const struct file_operations sq_fops = +{ + .owner = THIS_MODULE, + .llseek = no_llseek, + .write = sq_write, + .poll = sq_poll, + .unlocked_ioctl = sq_unlocked_ioctl, + .open = sq_open, + .release = sq_release, +}; + +static int sq_init(void) +{ + const struct file_operations *fops = &sq_fops; +#ifndef MODULE + int sq_unit; +#endif + + sq_unit = register_sound_dsp(fops, -1); + if (sq_unit < 0) { + printk(KERN_ERR "dmasound_core: couldn't register fops\n") ; + return sq_unit ; + } + + write_sq_init_waitqueue(); + + /* These parameters will be restored for every clean open() + * in the case of multiple open()s (e.g. dsp0 & dsp1) they + * will be set so long as the shared resources have no owner. + */ + + if (shared_resource_owner == 0) { + dmasound.soft = dmasound.mach.default_soft ; + dmasound.hard = dmasound.mach.default_hard ; + dmasound.dsp = dmasound.mach.default_soft ; + shared_resources_initialised = 0 ; + } + return 0 ; +} + + + /* + * /dev/sndstat + */ + +/* we allow more space for record-enabled because there are extra output lines. + the number here must include the amount we are prepared to give to the low-level + driver. +*/ + +#define STAT_BUFF_LEN 768 + +/* this is how much space we will allow the low-level driver to use + in the stat buffer. Currently, 2 * (80 character line + <NL>). + We do not police this (it is up to the ll driver to be honest). +*/ + +#define LOW_LEVEL_STAT_ALLOC 162 + +static struct { + int busy; + char buf[STAT_BUFF_LEN]; /* state.buf should not overflow! */ + int len, ptr; +} state; + +/* publish this function for use by low-level code, if required */ + +static char *get_afmt_string(int afmt) +{ + switch(afmt) { + case AFMT_MU_LAW: + return "mu-law"; + break; + case AFMT_A_LAW: + return "A-law"; + break; + case AFMT_U8: + return "unsigned 8 bit"; + break; + case AFMT_S8: + return "signed 8 bit"; + break; + case AFMT_S16_BE: + return "signed 16 bit BE"; + break; + case AFMT_U16_BE: + return "unsigned 16 bit BE"; + break; + case AFMT_S16_LE: + return "signed 16 bit LE"; + break; + case AFMT_U16_LE: + return "unsigned 16 bit LE"; + break; + case 0: + return "format not set" ; + break ; + default: + break ; + } + return "ERROR: Unsupported AFMT_XXXX code" ; +} + +static int state_open(struct inode *inode, struct file *file) +{ + char *buffer = state.buf; + int len = 0; + int ret; + + mutex_lock(&dmasound_core_mutex); + ret = -EBUSY; + if (state.busy) + goto out; + + ret = -ENODEV; + if (!try_module_get(dmasound.mach.owner)) + goto out; + + state.ptr = 0; + state.busy = 1; + + len += sprintf(buffer+len, "%sDMA sound driver rev %03d :\n", + dmasound.mach.name, (DMASOUND_CORE_REVISION<<4) + + ((dmasound.mach.version>>8) & 0x0f)); + len += sprintf(buffer+len, + "Core driver edition %02d.%02d : %s driver edition %02d.%02d\n", + DMASOUND_CORE_REVISION, DMASOUND_CORE_EDITION, dmasound.mach.name2, + (dmasound.mach.version >> 8), (dmasound.mach.version & 0xff)) ; + + /* call the low-level module to fill in any stat info. that it has + if present. Maximum buffer usage is specified. + */ + + if (dmasound.mach.state_info) + len += dmasound.mach.state_info(buffer+len, + (size_t) LOW_LEVEL_STAT_ALLOC) ; + + /* make usage of the state buffer as deterministic as poss. + exceptional conditions could cause overrun - and this is flagged as + a kernel error. + */ + + /* formats and settings */ + + len += sprintf(buffer+len,"\t\t === Formats & settings ===\n") ; + len += sprintf(buffer+len,"Parameter %20s%20s\n","soft","hard") ; + len += sprintf(buffer+len,"Format :%20s%20s\n", + get_afmt_string(dmasound.soft.format), + get_afmt_string(dmasound.hard.format)); + + len += sprintf(buffer+len,"Samp Rate:%14d s/sec%14d s/sec\n", + dmasound.soft.speed, dmasound.hard.speed); + + len += sprintf(buffer+len,"Channels :%20s%20s\n", + dmasound.soft.stereo ? "stereo" : "mono", + dmasound.hard.stereo ? "stereo" : "mono" ); + + /* sound queue status */ + + len += sprintf(buffer+len,"\t\t === Sound Queue status ===\n"); + len += sprintf(buffer+len,"Allocated:%8s%6s\n","Buffers","Size") ; + len += sprintf(buffer+len,"%9s:%8d%6d\n", + "write", write_sq.numBufs, write_sq.bufSize) ; + len += sprintf(buffer+len, + "Current : MaxFrg FragSiz MaxAct Frnt Rear " + "Cnt RrSize A B S L xruns\n") ; + len += sprintf(buffer+len,"%9s:%7d%8d%7d%5d%5d%4d%7d%2d%2d%2d%2d%7d\n", + "write", write_sq.max_count, write_sq.block_size, + write_sq.max_active, write_sq.front, write_sq.rear, + write_sq.count, write_sq.rear_size, write_sq.active, + write_sq.busy, write_sq.syncing, write_sq.locked, write_sq.xruns) ; +#ifdef DEBUG_DMASOUND +printk("dmasound: stat buffer used %d bytes\n", len) ; +#endif + + if (len >= STAT_BUFF_LEN) + printk(KERN_ERR "dmasound_core: stat buffer overflowed!\n"); + + state.len = len; + ret = 0; +out: + mutex_unlock(&dmasound_core_mutex); + return ret; +} + +static int state_release(struct inode *inode, struct file *file) +{ + mutex_lock(&dmasound_core_mutex); + state.busy = 0; + module_put(dmasound.mach.owner); + mutex_unlock(&dmasound_core_mutex); + return 0; +} + +static ssize_t state_read(struct file *file, char __user *buf, size_t count, + loff_t *ppos) +{ + int n = state.len - state.ptr; + if (n > count) + n = count; + if (n <= 0) + return 0; + if (copy_to_user(buf, &state.buf[state.ptr], n)) + return -EFAULT; + state.ptr += n; + return n; +} + +static const struct file_operations state_fops = { + .owner = THIS_MODULE, + .llseek = no_llseek, + .read = state_read, + .open = state_open, + .release = state_release, +}; + +static int state_init(void) +{ +#ifndef MODULE + int state_unit; +#endif + state_unit = register_sound_special(&state_fops, SND_DEV_STATUS); + if (state_unit < 0) + return state_unit ; + state.busy = 0; + return 0 ; +} + + + /* + * Config & Setup + * + * This function is called by _one_ chipset-specific driver + */ + +int dmasound_init(void) +{ + int res ; +#ifdef MODULE + if (irq_installed) + return -EBUSY; +#endif + + /* Set up sound queue, /dev/audio and /dev/dsp. */ + + /* Set default settings. */ + if ((res = sq_init()) < 0) + return res ; + + /* Set up /dev/sndstat. */ + if ((res = state_init()) < 0) + return res ; + + /* Set up /dev/mixer. */ + mixer_init(); + + if (!dmasound.mach.irqinit()) { + printk(KERN_ERR "DMA sound driver: Interrupt initialization failed\n"); + return -ENODEV; + } +#ifdef MODULE + irq_installed = 1; +#endif + + printk(KERN_INFO "%s DMA sound driver rev %03d installed\n", + dmasound.mach.name, (DMASOUND_CORE_REVISION<<4) + + ((dmasound.mach.version>>8) & 0x0f)); + printk(KERN_INFO + "Core driver edition %02d.%02d : %s driver edition %02d.%02d\n", + DMASOUND_CORE_REVISION, DMASOUND_CORE_EDITION, dmasound.mach.name2, + (dmasound.mach.version >> 8), (dmasound.mach.version & 0xff)) ; + printk(KERN_INFO "Write will use %4d fragments of %7d bytes as default\n", + numWriteBufs, writeBufSize) ; + return 0; +} + +#ifdef MODULE + +void dmasound_deinit(void) +{ + if (irq_installed) { + sound_silence(); + dmasound.mach.irqcleanup(); + irq_installed = 0; + } + + write_sq_release_buffers(); + + if (mixer_unit >= 0) + unregister_sound_mixer(mixer_unit); + if (state_unit >= 0) + unregister_sound_special(state_unit); + if (sq_unit >= 0) + unregister_sound_dsp(sq_unit); +} + +#else /* !MODULE */ + +static int dmasound_setup(char *str) +{ + int ints[6], size; + + str = get_options(str, ARRAY_SIZE(ints), ints); + + /* check the bootstrap parameter for "dmasound=" */ + + /* FIXME: other than in the most naive of cases there is no sense in these + * buffers being other than powers of two. This is not checked yet. + */ + + switch (ints[0]) { + case 3: + if ((ints[3] < 0) || (ints[3] > MAX_CATCH_RADIUS)) + printk("dmasound_setup: invalid catch radius, using default = %d\n", catchRadius); + else + catchRadius = ints[3]; + /* fall through */ + case 2: + if (ints[1] < MIN_BUFFERS) + printk("dmasound_setup: invalid number of buffers, using default = %d\n", numWriteBufs); + else + numWriteBufs = ints[1]; + /* fall through */ + case 1: + if ((size = ints[2]) < 256) /* check for small buffer specs */ + size <<= 10 ; + if (size < MIN_BUFSIZE || size > MAX_BUFSIZE) + printk("dmasound_setup: invalid write buffer size, using default = %d\n", writeBufSize); + else + writeBufSize = size; + case 0: + break; + default: + printk("dmasound_setup: invalid number of arguments\n"); + return 0; + } + return 1; +} + +__setup("dmasound=", dmasound_setup); + +#endif /* !MODULE */ + + /* + * Conversion tables + */ + +#ifdef HAS_8BIT_TABLES +/* 8 bit mu-law */ + +char dmasound_ulaw2dma8[] = { + -126, -122, -118, -114, -110, -106, -102, -98, + -94, -90, -86, -82, -78, -74, -70, -66, + -63, -61, -59, -57, -55, -53, -51, -49, + -47, -45, -43, -41, -39, -37, -35, -33, + -31, -30, -29, -28, -27, -26, -25, -24, + -23, -22, -21, -20, -19, -18, -17, -16, + -16, -15, -15, -14, -14, -13, -13, -12, + -12, -11, -11, -10, -10, -9, -9, -8, + -8, -8, -7, -7, -7, -7, -6, -6, + -6, -6, -5, -5, -5, -5, -4, -4, + -4, -4, -4, -4, -3, -3, -3, -3, + -3, -3, -3, -3, -2, -2, -2, -2, + -2, -2, -2, -2, -2, -2, -2, -2, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, 0, + 125, 121, 117, 113, 109, 105, 101, 97, + 93, 89, 85, 81, 77, 73, 69, 65, + 62, 60, 58, 56, 54, 52, 50, 48, + 46, 44, 42, 40, 38, 36, 34, 32, + 30, 29, 28, 27, 26, 25, 24, 23, + 22, 21, 20, 19, 18, 17, 16, 15, + 15, 14, 14, 13, 13, 12, 12, 11, + 11, 10, 10, 9, 9, 8, 8, 7, + 7, 7, 6, 6, 6, 6, 5, 5, + 5, 5, 4, 4, 4, 4, 3, 3, + 3, 3, 3, 3, 2, 2, 2, 2, + 2, 2, 2, 2, 1, 1, 1, 1, + 1, 1, 1, 1, 1, 1, 1, 1, + 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0 +}; + +/* 8 bit A-law */ + +char dmasound_alaw2dma8[] = { + -22, -21, -24, -23, -18, -17, -20, -19, + -30, -29, -32, -31, -26, -25, -28, -27, + -11, -11, -12, -12, -9, -9, -10, -10, + -15, -15, -16, -16, -13, -13, -14, -14, + -86, -82, -94, -90, -70, -66, -78, -74, + -118, -114, -126, -122, -102, -98, -110, -106, + -43, -41, -47, -45, -35, -33, -39, -37, + -59, -57, -63, -61, -51, -49, -55, -53, + -2, -2, -2, -2, -2, -2, -2, -2, + -2, -2, -2, -2, -2, -2, -2, -2, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -6, -6, -6, -6, -5, -5, -5, -5, + -8, -8, -8, -8, -7, -7, -7, -7, + -3, -3, -3, -3, -3, -3, -3, -3, + -4, -4, -4, -4, -4, -4, -4, -4, + 21, 20, 23, 22, 17, 16, 19, 18, + 29, 28, 31, 30, 25, 24, 27, 26, + 10, 10, 11, 11, 8, 8, 9, 9, + 14, 14, 15, 15, 12, 12, 13, 13, + 86, 82, 94, 90, 70, 66, 78, 74, + 118, 114, 126, 122, 102, 98, 110, 106, + 43, 41, 47, 45, 35, 33, 39, 37, + 59, 57, 63, 61, 51, 49, 55, 53, + 1, 1, 1, 1, 1, 1, 1, 1, + 1, 1, 1, 1, 1, 1, 1, 1, + 0, 0, 0, 0, 0, 0, 0, 0, + 0, 0, 0, 0, 0, 0, 0, 0, + 5, 5, 5, 5, 4, 4, 4, 4, + 7, 7, 7, 7, 6, 6, 6, 6, + 2, 2, 2, 2, 2, 2, 2, 2, + 3, 3, 3, 3, 3, 3, 3, 3 +}; +#endif /* HAS_8BIT_TABLES */ + + /* + * Visible symbols for modules + */ + +EXPORT_SYMBOL(dmasound); +EXPORT_SYMBOL(dmasound_init); +#ifdef MODULE +EXPORT_SYMBOL(dmasound_deinit); +#endif +EXPORT_SYMBOL(dmasound_write_sq); +EXPORT_SYMBOL(dmasound_catchRadius); +#ifdef HAS_8BIT_TABLES +EXPORT_SYMBOL(dmasound_ulaw2dma8); +EXPORT_SYMBOL(dmasound_alaw2dma8); +#endif diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c new file mode 100644 index 00000000..87910e99 --- /dev/null +++ b/sound/oss/dmasound/dmasound_paula.c @@ -0,0 +1,751 @@ +/* + * linux/sound/oss/dmasound/dmasound_paula.c + * + * Amiga `Paula' DMA Sound Driver + * + * See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits + * prior to 28/01/2001 + * + * 28/01/2001 [0.1] Iain Sandoe + * - added versioning + * - put in and populated the hardware_afmts field. + * [0.2] - put in SNDCTL_DSP_GETCAPS value. + * [0.3] - put in constraint on state buffer usage. + * [0.4] - put in default hard/soft settings +*/ + + +#include <linux/module.h> +#include <linux/mm.h> +#include <linux/init.h> +#include <linux/ioport.h> +#include <linux/soundcard.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <asm/uaccess.h> +#include <asm/setup.h> +#include <asm/amigahw.h> +#include <asm/amigaints.h> +#include <asm/machdep.h> + +#include "dmasound.h" + +#define DMASOUND_PAULA_REVISION 0 +#define DMASOUND_PAULA_EDITION 4 + +#define custom amiga_custom + /* + * The minimum period for audio depends on htotal (for OCS/ECS/AGA) + * (Imported from arch/m68k/amiga/amisound.c) + */ + +extern volatile u_short amiga_audio_min_period; + + + /* + * amiga_mksound() should be able to restore the period after beeping + * (Imported from arch/m68k/amiga/amisound.c) + */ + +extern u_short amiga_audio_period; + + + /* + * Audio DMA masks + */ + +#define AMI_AUDIO_OFF (DMAF_AUD0 | DMAF_AUD1 | DMAF_AUD2 | DMAF_AUD3) +#define AMI_AUDIO_8 (DMAF_SETCLR | DMAF_MASTER | DMAF_AUD0 | DMAF_AUD1) +#define AMI_AUDIO_14 (AMI_AUDIO_8 | DMAF_AUD2 | DMAF_AUD3) + + + /* + * Helper pointers for 16(14)-bit sound + */ + +static int write_sq_block_size_half, write_sq_block_size_quarter; + + +/*** Low level stuff *********************************************************/ + + +static void *AmiAlloc(unsigned int size, gfp_t flags); +static void AmiFree(void *obj, unsigned int size); +static int AmiIrqInit(void); +#ifdef MODULE +static void AmiIrqCleanUp(void); +#endif +static void AmiSilence(void); +static void AmiInit(void); +static int AmiSetFormat(int format); +static int AmiSetVolume(int volume); +static int AmiSetTreble(int treble); +static void AmiPlayNextFrame(int index); +static void AmiPlay(void); +static irqreturn_t AmiInterrupt(int irq, void *dummy); + +#ifdef CONFIG_HEARTBEAT + + /* + * Heartbeat interferes with sound since the 7 kHz low-pass filter and the + * power LED are controlled by the same line. + */ + +static void (*saved_heartbeat)(int) = NULL; + +static inline void disable_heartbeat(void) +{ + if (mach_heartbeat) { + saved_heartbeat = mach_heartbeat; + mach_heartbeat = NULL; + } + AmiSetTreble(dmasound.treble); +} + +static inline void enable_heartbeat(void) +{ + if (saved_heartbeat) + mach_heartbeat = saved_heartbeat; +} +#else /* !CONFIG_HEARTBEAT */ +#define disable_heartbeat() do { } while (0) +#define enable_heartbeat() do { } while (0) +#endif /* !CONFIG_HEARTBEAT */ + + +/*** Mid level stuff *********************************************************/ + +static void AmiMixerInit(void); +static int AmiMixerIoctl(u_int cmd, u_long arg); +static int AmiWriteSqSetup(void); +static int AmiStateInfo(char *buffer, size_t space); + + +/*** Translations ************************************************************/ + +/* ++TeSche: radically changed for new expanding purposes... + * + * These two routines now deal with copying/expanding/translating the samples + * from user space into our buffer at the right frequency. They take care about + * how much data there's actually to read, how much buffer space there is and + * to convert samples into the right frequency/encoding. They will only work on + * complete samples so it may happen they leave some bytes in the input stream + * if the user didn't write a multiple of the current sample size. They both + * return the number of bytes they've used from both streams so you may detect + * such a situation. Luckily all programs should be able to cope with that. + * + * I think I've optimized anything as far as one can do in plain C, all + * variables should fit in registers and the loops are really short. There's + * one loop for every possible situation. Writing a more generalized and thus + * parameterized loop would only produce slower code. Feel free to optimize + * this in assembler if you like. :) + * + * I think these routines belong here because they're not yet really hardware + * independent, especially the fact that the Falcon can play 16bit samples + * only in stereo is hardcoded in both of them! + * + * ++geert: split in even more functions (one per format) + */ + + + /* + * Native format + */ + +static ssize_t ami_ct_s8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, ssize_t frameLeft) +{ + ssize_t count, used; + + if (!dmasound.soft.stereo) { + void *p = &frame[*frameUsed]; + count = min_t(unsigned long, userCount, frameLeft) & ~1; + used = count; + if (copy_from_user(p, userPtr, count)) + return -EFAULT; + } else { + u_char *left = &frame[*frameUsed>>1]; + u_char *right = left+write_sq_block_size_half; + count = min_t(unsigned long, userCount, frameLeft)>>1 & ~1; + used = count*2; + while (count > 0) { + if (get_user(*left++, userPtr++) + || get_user(*right++, userPtr++)) + return -EFAULT; + count--; + } + } + *frameUsed += used; + return used; +} + + + /* + * Copy and convert 8 bit data + */ + +#define GENERATE_AMI_CT8(funcname, convsample) \ +static ssize_t funcname(const u_char __user *userPtr, size_t userCount, \ + u_char frame[], ssize_t *frameUsed, \ + ssize_t frameLeft) \ +{ \ + ssize_t count, used; \ + \ + if (!dmasound.soft.stereo) { \ + u_char *p = &frame[*frameUsed]; \ + count = min_t(size_t, userCount, frameLeft) & ~1; \ + used = count; \ + while (count > 0) { \ + u_char data; \ + if (get_user(data, userPtr++)) \ + return -EFAULT; \ + *p++ = convsample(data); \ + count--; \ + } \ + } else { \ + u_char *left = &frame[*frameUsed>>1]; \ + u_char *right = left+write_sq_block_size_half; \ + count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \ + used = count*2; \ + while (count > 0) { \ + u_char data; \ + if (get_user(data, userPtr++)) \ + return -EFAULT; \ + *left++ = convsample(data); \ + if (get_user(data, userPtr++)) \ + return -EFAULT; \ + *right++ = convsample(data); \ + count--; \ + } \ + } \ + *frameUsed += used; \ + return used; \ +} + +#define AMI_CT_ULAW(x) (dmasound_ulaw2dma8[(x)]) +#define AMI_CT_ALAW(x) (dmasound_alaw2dma8[(x)]) +#define AMI_CT_U8(x) ((x) ^ 0x80) + +GENERATE_AMI_CT8(ami_ct_ulaw, AMI_CT_ULAW) +GENERATE_AMI_CT8(ami_ct_alaw, AMI_CT_ALAW) +GENERATE_AMI_CT8(ami_ct_u8, AMI_CT_U8) + + + /* + * Copy and convert 16 bit data + */ + +#define GENERATE_AMI_CT_16(funcname, convsample) \ +static ssize_t funcname(const u_char __user *userPtr, size_t userCount, \ + u_char frame[], ssize_t *frameUsed, \ + ssize_t frameLeft) \ +{ \ + const u_short __user *ptr = (const u_short __user *)userPtr; \ + ssize_t count, used; \ + u_short data; \ + \ + if (!dmasound.soft.stereo) { \ + u_char *high = &frame[*frameUsed>>1]; \ + u_char *low = high+write_sq_block_size_half; \ + count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \ + used = count*2; \ + while (count > 0) { \ + if (get_user(data, ptr++)) \ + return -EFAULT; \ + data = convsample(data); \ + *high++ = data>>8; \ + *low++ = (data>>2) & 0x3f; \ + count--; \ + } \ + } else { \ + u_char *lefth = &frame[*frameUsed>>2]; \ + u_char *leftl = lefth+write_sq_block_size_quarter; \ + u_char *righth = lefth+write_sq_block_size_half; \ + u_char *rightl = righth+write_sq_block_size_quarter; \ + count = min_t(size_t, userCount, frameLeft)>>2 & ~1; \ + used = count*4; \ + while (count > 0) { \ + if (get_user(data, ptr++)) \ + return -EFAULT; \ + data = convsample(data); \ + *lefth++ = data>>8; \ + *leftl++ = (data>>2) & 0x3f; \ + if (get_user(data, ptr++)) \ + return -EFAULT; \ + data = convsample(data); \ + *righth++ = data>>8; \ + *rightl++ = (data>>2) & 0x3f; \ + count--; \ + } \ + } \ + *frameUsed += used; \ + return used; \ +} + +#define AMI_CT_S16BE(x) (x) +#define AMI_CT_U16BE(x) ((x) ^ 0x8000) +#define AMI_CT_S16LE(x) (le2be16((x))) +#define AMI_CT_U16LE(x) (le2be16((x)) ^ 0x8000) + +GENERATE_AMI_CT_16(ami_ct_s16be, AMI_CT_S16BE) +GENERATE_AMI_CT_16(ami_ct_u16be, AMI_CT_U16BE) +GENERATE_AMI_CT_16(ami_ct_s16le, AMI_CT_S16LE) +GENERATE_AMI_CT_16(ami_ct_u16le, AMI_CT_U16LE) + + +static TRANS transAmiga = { + .ct_ulaw = ami_ct_ulaw, + .ct_alaw = ami_ct_alaw, + .ct_s8 = ami_ct_s8, + .ct_u8 = ami_ct_u8, + .ct_s16be = ami_ct_s16be, + .ct_u16be = ami_ct_u16be, + .ct_s16le = ami_ct_s16le, + .ct_u16le = ami_ct_u16le, +}; + +/*** Low level stuff *********************************************************/ + +static inline void StopDMA(void) +{ + custom.aud[0].audvol = custom.aud[1].audvol = 0; + custom.aud[2].audvol = custom.aud[3].audvol = 0; + custom.dmacon = AMI_AUDIO_OFF; + enable_heartbeat(); +} + +static void *AmiAlloc(unsigned int size, gfp_t flags) +{ + return amiga_chip_alloc((long)size, "dmasound [Paula]"); +} + +static void AmiFree(void *obj, unsigned int size) +{ + amiga_chip_free (obj); +} + +static int __init AmiIrqInit(void) +{ + /* turn off DMA for audio channels */ + StopDMA(); + + /* Register interrupt handler. */ + if (request_irq(IRQ_AMIGA_AUD0, AmiInterrupt, 0, "DMA sound", + AmiInterrupt)) + return 0; + return 1; +} + +#ifdef MODULE +static void AmiIrqCleanUp(void) +{ + /* turn off DMA for audio channels */ + StopDMA(); + /* release the interrupt */ + free_irq(IRQ_AMIGA_AUD0, AmiInterrupt); +} +#endif /* MODULE */ + +static void AmiSilence(void) +{ + /* turn off DMA for audio channels */ + StopDMA(); +} + + +static void AmiInit(void) +{ + int period, i; + + AmiSilence(); + + if (dmasound.soft.speed) + period = amiga_colorclock/dmasound.soft.speed-1; + else + period = amiga_audio_min_period; + dmasound.hard = dmasound.soft; + dmasound.trans_write = &transAmiga; + + if (period < amiga_audio_min_period) { + /* we would need to squeeze the sound, but we won't do that */ + period = amiga_audio_min_period; + } else if (period > 65535) { + period = 65535; + } + dmasound.hard.speed = amiga_colorclock/(period+1); + + for (i = 0; i < 4; i++) + custom.aud[i].audper = period; + amiga_audio_period = period; +} + + +static int AmiSetFormat(int format) +{ + int size; + + /* Amiga sound DMA supports 8bit and 16bit (pseudo 14 bit) modes */ + + switch (format) { + case AFMT_QUERY: + return dmasound.soft.format; + case AFMT_MU_LAW: + case AFMT_A_LAW: + case AFMT_U8: + case AFMT_S8: + size = 8; + break; + case AFMT_S16_BE: + case AFMT_U16_BE: + case AFMT_S16_LE: + case AFMT_U16_LE: + size = 16; + break; + default: /* :-) */ + size = 8; + format = AFMT_S8; + } + + dmasound.soft.format = format; + dmasound.soft.size = size; + if (dmasound.minDev == SND_DEV_DSP) { + dmasound.dsp.format = format; + dmasound.dsp.size = dmasound.soft.size; + } + AmiInit(); + + return format; +} + + +#define VOLUME_VOXWARE_TO_AMI(v) \ + (((v) < 0) ? 0 : ((v) > 100) ? 64 : ((v) * 64)/100) +#define VOLUME_AMI_TO_VOXWARE(v) ((v)*100/64) + +static int AmiSetVolume(int volume) +{ + dmasound.volume_left = VOLUME_VOXWARE_TO_AMI(volume & 0xff); + custom.aud[0].audvol = dmasound.volume_left; + dmasound.volume_right = VOLUME_VOXWARE_TO_AMI((volume & 0xff00) >> 8); + custom.aud[1].audvol = dmasound.volume_right; + if (dmasound.hard.size == 16) { + if (dmasound.volume_left == 64 && dmasound.volume_right == 64) { + custom.aud[2].audvol = 1; + custom.aud[3].audvol = 1; + } else { + custom.aud[2].audvol = 0; + custom.aud[3].audvol = 0; + } + } + return VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) | + (VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8); +} + +static int AmiSetTreble(int treble) +{ + dmasound.treble = treble; + if (treble < 50) + ciaa.pra &= ~0x02; + else + ciaa.pra |= 0x02; + return treble; +} + + +#define AMI_PLAY_LOADED 1 +#define AMI_PLAY_PLAYING 2 +#define AMI_PLAY_MASK 3 + + +static void AmiPlayNextFrame(int index) +{ + u_char *start, *ch0, *ch1, *ch2, *ch3; + u_long size; + + /* used by AmiPlay() if all doubts whether there really is something + * to be played are already wiped out. + */ + start = write_sq.buffers[write_sq.front]; + size = (write_sq.count == index ? write_sq.rear_size + : write_sq.block_size)>>1; + + if (dmasound.hard.stereo) { + ch0 = start; + ch1 = start+write_sq_block_size_half; + size >>= 1; + } else { + ch0 = start; + ch1 = start; + } + + disable_heartbeat(); + custom.aud[0].audvol = dmasound.volume_left; + custom.aud[1].audvol = dmasound.volume_right; + if (dmasound.hard.size == 8) { + custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0); + custom.aud[0].audlen = size; + custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1); + custom.aud[1].audlen = size; + custom.dmacon = AMI_AUDIO_8; + } else { + size >>= 1; + custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0); + custom.aud[0].audlen = size; + custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1); + custom.aud[1].audlen = size; + if (dmasound.volume_left == 64 && dmasound.volume_right == 64) { + /* We can play pseudo 14-bit only with the maximum volume */ + ch3 = ch0+write_sq_block_size_quarter; + ch2 = ch1+write_sq_block_size_quarter; + custom.aud[2].audvol = 1; /* we are being affected by the beeps */ + custom.aud[3].audvol = 1; /* restoring volume here helps a bit */ + custom.aud[2].audlc = (u_short *)ZTWO_PADDR(ch2); + custom.aud[2].audlen = size; + custom.aud[3].audlc = (u_short *)ZTWO_PADDR(ch3); + custom.aud[3].audlen = size; + custom.dmacon = AMI_AUDIO_14; + } else { + custom.aud[2].audvol = 0; + custom.aud[3].audvol = 0; + custom.dmacon = AMI_AUDIO_8; + } + } + write_sq.front = (write_sq.front+1) % write_sq.max_count; + write_sq.active |= AMI_PLAY_LOADED; +} + + +static void AmiPlay(void) +{ + int minframes = 1; + + custom.intena = IF_AUD0; + + if (write_sq.active & AMI_PLAY_LOADED) { + /* There's already a frame loaded */ + custom.intena = IF_SETCLR | IF_AUD0; + return; + } + + if (write_sq.active & AMI_PLAY_PLAYING) + /* Increase threshold: frame 1 is already being played */ + minframes = 2; + + if (write_sq.count < minframes) { + /* Nothing to do */ + custom.intena = IF_SETCLR | IF_AUD0; + return; + } + + if (write_sq.count <= minframes && + write_sq.rear_size < write_sq.block_size && !write_sq.syncing) { + /* hmmm, the only existing frame is not + * yet filled and we're not syncing? + */ + custom.intena = IF_SETCLR | IF_AUD0; + return; + } + + AmiPlayNextFrame(minframes); + + custom.intena = IF_SETCLR | IF_AUD0; +} + + +static irqreturn_t AmiInterrupt(int irq, void *dummy) +{ + int minframes = 1; + + custom.intena = IF_AUD0; + + if (!write_sq.active) { + /* Playing was interrupted and sq_reset() has already cleared + * the sq variables, so better don't do anything here. + */ + WAKE_UP(write_sq.sync_queue); + return IRQ_HANDLED; + } + + if (write_sq.active & AMI_PLAY_PLAYING) { + /* We've just finished a frame */ + write_sq.count--; + WAKE_UP(write_sq.action_queue); + } + + if (write_sq.active & AMI_PLAY_LOADED) + /* Increase threshold: frame 1 is already being played */ + minframes = 2; + + /* Shift the flags */ + write_sq.active = (write_sq.active<<1) & AMI_PLAY_MASK; + + if (!write_sq.active) + /* No frame is playing, disable audio DMA */ + StopDMA(); + + custom.intena = IF_SETCLR | IF_AUD0; + + if (write_sq.count >= minframes) + /* Try to play the next frame */ + AmiPlay(); + + if (!write_sq.active) + /* Nothing to play anymore. + Wake up a process waiting for audio output to drain. */ + WAKE_UP(write_sq.sync_queue); + return IRQ_HANDLED; +} + +/*** Mid level stuff *********************************************************/ + + +/* + * /dev/mixer abstraction + */ + +static void __init AmiMixerInit(void) +{ + dmasound.volume_left = 64; + dmasound.volume_right = 64; + custom.aud[0].audvol = dmasound.volume_left; + custom.aud[3].audvol = 1; /* For pseudo 14bit */ + custom.aud[1].audvol = dmasound.volume_right; + custom.aud[2].audvol = 1; /* For pseudo 14bit */ + dmasound.treble = 50; +} + +static int AmiMixerIoctl(u_int cmd, u_long arg) +{ + int data; + switch (cmd) { + case SOUND_MIXER_READ_DEVMASK: + return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_TREBLE); + case SOUND_MIXER_READ_RECMASK: + return IOCTL_OUT(arg, 0); + case SOUND_MIXER_READ_STEREODEVS: + return IOCTL_OUT(arg, SOUND_MASK_VOLUME); + case SOUND_MIXER_READ_VOLUME: + return IOCTL_OUT(arg, + VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) | + VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8); + case SOUND_MIXER_WRITE_VOLUME: + IOCTL_IN(arg, data); + return IOCTL_OUT(arg, dmasound_set_volume(data)); + case SOUND_MIXER_READ_TREBLE: + return IOCTL_OUT(arg, dmasound.treble); + case SOUND_MIXER_WRITE_TREBLE: + IOCTL_IN(arg, data); + return IOCTL_OUT(arg, dmasound_set_treble(data)); + } + return -EINVAL; +} + + +static int AmiWriteSqSetup(void) +{ + write_sq_block_size_half = write_sq.block_size>>1; + write_sq_block_size_quarter = write_sq_block_size_half>>1; + return 0; +} + + +static int AmiStateInfo(char *buffer, size_t space) +{ + int len = 0; + len += sprintf(buffer+len, "\tsound.volume_left = %d [0...64]\n", + dmasound.volume_left); + len += sprintf(buffer+len, "\tsound.volume_right = %d [0...64]\n", + dmasound.volume_right); + if (len >= space) { + printk(KERN_ERR "dmasound_paula: overflowed state buffer alloc.\n") ; + len = space ; + } + return len; +} + + +/*** Machine definitions *****************************************************/ + +static SETTINGS def_hard = { + .format = AFMT_S8, + .stereo = 0, + .size = 8, + .speed = 8000 +} ; + +static SETTINGS def_soft = { + .format = AFMT_U8, + .stereo = 0, + .size = 8, + .speed = 8000 +} ; + +static MACHINE machAmiga = { + .name = "Amiga", + .name2 = "AMIGA", + .owner = THIS_MODULE, + .dma_alloc = AmiAlloc, + .dma_free = AmiFree, + .irqinit = AmiIrqInit, +#ifdef MODULE + .irqcleanup = AmiIrqCleanUp, +#endif /* MODULE */ + .init = AmiInit, + .silence = AmiSilence, + .setFormat = AmiSetFormat, + .setVolume = AmiSetVolume, + .setTreble = AmiSetTreble, + .play = AmiPlay, + .mixer_init = AmiMixerInit, + .mixer_ioctl = AmiMixerIoctl, + .write_sq_setup = AmiWriteSqSetup, + .state_info = AmiStateInfo, + .min_dsp_speed = 8000, + .version = ((DMASOUND_PAULA_REVISION<<8) | DMASOUND_PAULA_EDITION), + .hardware_afmts = (AFMT_S8 | AFMT_S16_BE), /* h'ware-supported formats *only* here */ + .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */ +}; + + +/*** Config & Setup **********************************************************/ + + +static int __init amiga_audio_probe(struct platform_device *pdev) +{ + dmasound.mach = machAmiga; + dmasound.mach.default_hard = def_hard ; + dmasound.mach.default_soft = def_soft ; + return dmasound_init(); +} + +static int __exit amiga_audio_remove(struct platform_device *pdev) +{ + dmasound_deinit(); + return 0; +} + +static struct platform_driver amiga_audio_driver = { + .remove = __exit_p(amiga_audio_remove), + .driver = { + .name = "amiga-audio", + .owner = THIS_MODULE, + }, +}; + +static int __init amiga_audio_init(void) +{ + return platform_driver_probe(&amiga_audio_driver, amiga_audio_probe); +} + +module_init(amiga_audio_init); + +static void __exit amiga_audio_exit(void) +{ + platform_driver_unregister(&amiga_audio_driver); +} + +module_exit(amiga_audio_exit); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:amiga-audio"); diff --git a/sound/oss/dmasound/dmasound_q40.c b/sound/oss/dmasound/dmasound_q40.c new file mode 100644 index 00000000..99bcb21c --- /dev/null +++ b/sound/oss/dmasound/dmasound_q40.c @@ -0,0 +1,638 @@ +/* + * linux/sound/oss/dmasound/dmasound_q40.c + * + * Q40 DMA Sound Driver + * + * See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits + * prior to 28/01/2001 + * + * 28/01/2001 [0.1] Iain Sandoe + * - added versioning + * - put in and populated the hardware_afmts field. + * [0.2] - put in SNDCTL_DSP_GETCAPS value. + * [0.3] - put in default hard/soft settings. + */ + + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/soundcard.h> +#include <linux/interrupt.h> + +#include <asm/uaccess.h> +#include <asm/q40ints.h> +#include <asm/q40_master.h> + +#include "dmasound.h" + +#define DMASOUND_Q40_REVISION 0 +#define DMASOUND_Q40_EDITION 3 + +static int expand_bal; /* Balance factor for expanding (not volume!) */ +static int expand_data; /* Data for expanding */ + + +/*** Low level stuff *********************************************************/ + + +static void *Q40Alloc(unsigned int size, gfp_t flags); +static void Q40Free(void *, unsigned int); +static int Q40IrqInit(void); +#ifdef MODULE +static void Q40IrqCleanUp(void); +#endif +static void Q40Silence(void); +static void Q40Init(void); +static int Q40SetFormat(int format); +static int Q40SetVolume(int volume); +static void Q40PlayNextFrame(int index); +static void Q40Play(void); +static irqreturn_t Q40StereoInterrupt(int irq, void *dummy); +static irqreturn_t Q40MonoInterrupt(int irq, void *dummy); +static void Q40Interrupt(void); + + +/*** Mid level stuff *********************************************************/ + + + +/* userCount, frameUsed, frameLeft == byte counts */ +static ssize_t q40_ct_law(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + char *table = dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8: dmasound_alaw2dma8; + ssize_t count, used; + u_char *p = (u_char *) &frame[*frameUsed]; + + used = count = min_t(size_t, userCount, frameLeft); + if (copy_from_user(p,userPtr,count)) + return -EFAULT; + while (count > 0) { + *p = table[*p]+128; + p++; + count--; + } + *frameUsed += used ; + return used; +} + + +static ssize_t q40_ct_s8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + u_char *p = (u_char *) &frame[*frameUsed]; + + used = count = min_t(size_t, userCount, frameLeft); + if (copy_from_user(p,userPtr,count)) + return -EFAULT; + while (count > 0) { + *p = *p + 128; + p++; + count--; + } + *frameUsed += used; + return used; +} + +static ssize_t q40_ct_u8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + ssize_t count, used; + u_char *p = (u_char *) &frame[*frameUsed]; + + used = count = min_t(size_t, userCount, frameLeft); + if (copy_from_user(p,userPtr,count)) + return -EFAULT; + *frameUsed += used; + return used; +} + + +/* a bit too complicated to optimise right now ..*/ +static ssize_t q40_ctx_law(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + unsigned char *table = (unsigned char *) + (dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8: dmasound_alaw2dma8); + unsigned int data = expand_data; + u_char *p = (u_char *) &frame[*frameUsed]; + int bal = expand_bal; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + ftotal = frameLeft; + utotal = userCount; + while (frameLeft) { + u_char c; + if (bal < 0) { + if (userCount == 0) + break; + if (get_user(c, userPtr++)) + return -EFAULT; + data = table[c]; + data += 0x80; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + expand_bal = bal; + expand_data = data; + *frameUsed += (ftotal - frameLeft); + utotal -= userCount; + return utotal; +} + + +static ssize_t q40_ctx_s8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + u_char *p = (u_char *) &frame[*frameUsed]; + unsigned int data = expand_data; + int bal = expand_bal; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + + ftotal = frameLeft; + utotal = userCount; + while (frameLeft) { + u_char c; + if (bal < 0) { + if (userCount == 0) + break; + if (get_user(c, userPtr++)) + return -EFAULT; + data = c ; + data += 0x80; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + expand_bal = bal; + expand_data = data; + *frameUsed += (ftotal - frameLeft); + utotal -= userCount; + return utotal; +} + + +static ssize_t q40_ctx_u8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + u_char *p = (u_char *) &frame[*frameUsed]; + unsigned int data = expand_data; + int bal = expand_bal; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + ftotal = frameLeft; + utotal = userCount; + while (frameLeft) { + u_char c; + if (bal < 0) { + if (userCount == 0) + break; + if (get_user(c, userPtr++)) + return -EFAULT; + data = c ; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + expand_bal = bal; + expand_data = data; + *frameUsed += (ftotal - frameLeft) ; + utotal -= userCount; + return utotal; +} + +/* compressing versions */ +static ssize_t q40_ctc_law(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + unsigned char *table = (unsigned char *) + (dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8: dmasound_alaw2dma8); + unsigned int data = expand_data; + u_char *p = (u_char *) &frame[*frameUsed]; + int bal = expand_bal; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + ftotal = frameLeft; + utotal = userCount; + while (frameLeft) { + u_char c; + while(bal<0) { + if (userCount == 0) + goto lout; + if (!(bal<(-hSpeed))) { + if (get_user(c, userPtr)) + return -EFAULT; + data = 0x80 + table[c]; + } + userPtr++; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + lout: + expand_bal = bal; + expand_data = data; + *frameUsed += (ftotal - frameLeft); + utotal -= userCount; + return utotal; +} + + +static ssize_t q40_ctc_s8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + u_char *p = (u_char *) &frame[*frameUsed]; + unsigned int data = expand_data; + int bal = expand_bal; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + ftotal = frameLeft; + utotal = userCount; + while (frameLeft) { + u_char c; + while (bal < 0) { + if (userCount == 0) + goto lout; + if (!(bal<(-hSpeed))) { + if (get_user(c, userPtr)) + return -EFAULT; + data = c + 0x80; + } + userPtr++; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + lout: + expand_bal = bal; + expand_data = data; + *frameUsed += (ftotal - frameLeft); + utotal -= userCount; + return utotal; +} + + +static ssize_t q40_ctc_u8(const u_char __user *userPtr, size_t userCount, + u_char frame[], ssize_t *frameUsed, + ssize_t frameLeft) +{ + u_char *p = (u_char *) &frame[*frameUsed]; + unsigned int data = expand_data; + int bal = expand_bal; + int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed; + int utotal, ftotal; + + ftotal = frameLeft; + utotal = userCount; + while (frameLeft) { + u_char c; + while (bal < 0) { + if (userCount == 0) + goto lout; + if (!(bal<(-hSpeed))) { + if (get_user(c, userPtr)) + return -EFAULT; + data = c ; + } + userPtr++; + userCount--; + bal += hSpeed; + } + *p++ = data; + frameLeft--; + bal -= sSpeed; + } + lout: + expand_bal = bal; + expand_data = data; + *frameUsed += (ftotal - frameLeft) ; + utotal -= userCount; + return utotal; +} + + +static TRANS transQ40Normal = { + q40_ct_law, q40_ct_law, q40_ct_s8, q40_ct_u8, NULL, NULL, NULL, NULL +}; + +static TRANS transQ40Expanding = { + q40_ctx_law, q40_ctx_law, q40_ctx_s8, q40_ctx_u8, NULL, NULL, NULL, NULL +}; + +static TRANS transQ40Compressing = { + q40_ctc_law, q40_ctc_law, q40_ctc_s8, q40_ctc_u8, NULL, NULL, NULL, NULL +}; + + +/*** Low level stuff *********************************************************/ + +static void *Q40Alloc(unsigned int size, gfp_t flags) +{ + return kmalloc(size, flags); /* change to vmalloc */ +} + +static void Q40Free(void *ptr, unsigned int size) +{ + kfree(ptr); +} + +static int __init Q40IrqInit(void) +{ + /* Register interrupt handler. */ + if (request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0, + "DMA sound", Q40Interrupt)) + return 0; + + return(1); +} + + +#ifdef MODULE +static void Q40IrqCleanUp(void) +{ + master_outb(0,SAMPLE_ENABLE_REG); + free_irq(Q40_IRQ_SAMPLE, Q40Interrupt); +} +#endif /* MODULE */ + + +static void Q40Silence(void) +{ + master_outb(0,SAMPLE_ENABLE_REG); + *DAC_LEFT=*DAC_RIGHT=127; +} + +static char *q40_pp; +static unsigned int q40_sc; + +static void Q40PlayNextFrame(int index) +{ + u_char *start; + u_long size; + u_char speed; + int error; + + /* used by Q40Play() if all doubts whether there really is something + * to be played are already wiped out. + */ + start = write_sq.buffers[write_sq.front]; + size = (write_sq.count == index ? write_sq.rear_size : write_sq.block_size); + + q40_pp=start; + q40_sc=size; + + write_sq.front = (write_sq.front+1) % write_sq.max_count; + write_sq.active++; + + speed=(dmasound.hard.speed==10000 ? 0 : 1); + + master_outb( 0,SAMPLE_ENABLE_REG); + free_irq(Q40_IRQ_SAMPLE, Q40Interrupt); + if (dmasound.soft.stereo) + error = request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0, + "Q40 sound", Q40Interrupt); + else + error = request_irq(Q40_IRQ_SAMPLE, Q40MonoInterrupt, 0, + "Q40 sound", Q40Interrupt); + if (error && printk_ratelimit()) + pr_err("Couldn't register sound interrupt\n"); + + master_outb( speed, SAMPLE_RATE_REG); + master_outb( 1,SAMPLE_CLEAR_REG); + master_outb( 1,SAMPLE_ENABLE_REG); +} + +static void Q40Play(void) +{ + unsigned long flags; + + if (write_sq.active || write_sq.count<=0 ) { + /* There's already a frame loaded */ + return; + } + + /* nothing in the queue */ + if (write_sq.count <= 1 && write_sq.rear_size < write_sq.block_size && !write_sq.syncing) { + /* hmmm, the only existing frame is not + * yet filled and we're not syncing? + */ + return; + } + spin_lock_irqsave(&dmasound.lock, flags); + Q40PlayNextFrame(1); + spin_unlock_irqrestore(&dmasound.lock, flags); +} + +static irqreturn_t Q40StereoInterrupt(int irq, void *dummy) +{ + spin_lock(&dmasound.lock); + if (q40_sc>1){ + *DAC_LEFT=*q40_pp++; + *DAC_RIGHT=*q40_pp++; + q40_sc -=2; + master_outb(1,SAMPLE_CLEAR_REG); + }else Q40Interrupt(); + spin_unlock(&dmasound.lock); + return IRQ_HANDLED; +} +static irqreturn_t Q40MonoInterrupt(int irq, void *dummy) +{ + spin_lock(&dmasound.lock); + if (q40_sc>0){ + *DAC_LEFT=*q40_pp; + *DAC_RIGHT=*q40_pp++; + q40_sc --; + master_outb(1,SAMPLE_CLEAR_REG); + }else Q40Interrupt(); + spin_unlock(&dmasound.lock); + return IRQ_HANDLED; +} +static void Q40Interrupt(void) +{ + if (!write_sq.active) { + /* playing was interrupted and sq_reset() has already cleared + * the sq variables, so better don't do anything here. + */ + WAKE_UP(write_sq.sync_queue); + master_outb(0,SAMPLE_ENABLE_REG); /* better safe */ + goto exit; + } else write_sq.active=0; + write_sq.count--; + Q40Play(); + + if (q40_sc<2) + { /* there was nothing to play, disable irq */ + master_outb(0,SAMPLE_ENABLE_REG); + *DAC_LEFT=*DAC_RIGHT=127; + } + WAKE_UP(write_sq.action_queue); + + exit: + master_outb(1,SAMPLE_CLEAR_REG); +} + + +static void Q40Init(void) +{ + int i, idx; + const int freq[] = {10000, 20000}; + + /* search a frequency that fits into the allowed error range */ + + idx = -1; + for (i = 0; i < 2; i++) + if ((100 * abs(dmasound.soft.speed - freq[i]) / freq[i]) <= catchRadius) + idx = i; + + dmasound.hard = dmasound.soft; + /*sound.hard.stereo=1;*/ /* no longer true */ + dmasound.hard.size=8; + + if (idx > -1) { + dmasound.soft.speed = freq[idx]; + dmasound.trans_write = &transQ40Normal; + } else + dmasound.trans_write = &transQ40Expanding; + + Q40Silence(); + + if (dmasound.hard.speed > 20200) { + /* squeeze the sound, we do that */ + dmasound.hard.speed = 20000; + dmasound.trans_write = &transQ40Compressing; + } else if (dmasound.hard.speed > 10000) { + dmasound.hard.speed = 20000; + } else { + dmasound.hard.speed = 10000; + } + expand_bal = -dmasound.soft.speed; +} + + +static int Q40SetFormat(int format) +{ + /* Q40 sound supports only 8bit modes */ + + switch (format) { + case AFMT_QUERY: + return(dmasound.soft.format); + case AFMT_MU_LAW: + case AFMT_A_LAW: + case AFMT_S8: + case AFMT_U8: + break; + default: + format = AFMT_S8; + } + + dmasound.soft.format = format; + dmasound.soft.size = 8; + if (dmasound.minDev == SND_DEV_DSP) { + dmasound.dsp.format = format; + dmasound.dsp.size = 8; + } + Q40Init(); + + return(format); +} + +static int Q40SetVolume(int volume) +{ + return 0; +} + + +/*** Machine definitions *****************************************************/ + +static SETTINGS def_hard = { + .format = AFMT_U8, + .stereo = 0, + .size = 8, + .speed = 10000 +} ; + +static SETTINGS def_soft = { + .format = AFMT_U8, + .stereo = 0, + .size = 8, + .speed = 8000 +} ; + +static MACHINE machQ40 = { + .name = "Q40", + .name2 = "Q40", + .owner = THIS_MODULE, + .dma_alloc = Q40Alloc, + .dma_free = Q40Free, + .irqinit = Q40IrqInit, +#ifdef MODULE + .irqcleanup = Q40IrqCleanUp, +#endif /* MODULE */ + .init = Q40Init, + .silence = Q40Silence, + .setFormat = Q40SetFormat, + .setVolume = Q40SetVolume, + .play = Q40Play, + .min_dsp_speed = 10000, + .version = ((DMASOUND_Q40_REVISION<<8) | DMASOUND_Q40_EDITION), + .hardware_afmts = AFMT_U8, /* h'ware-supported formats *only* here */ + .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */ +}; + + +/*** Config & Setup **********************************************************/ + + +static int __init dmasound_q40_init(void) +{ + if (MACH_IS_Q40) { + dmasound.mach = machQ40; + dmasound.mach.default_hard = def_hard ; + dmasound.mach.default_soft = def_soft ; + return dmasound_init(); + } else + return -ENODEV; +} + +static void __exit dmasound_q40_cleanup(void) +{ + dmasound_deinit(); +} + +module_init(dmasound_q40_init); +module_exit(dmasound_q40_cleanup); + +MODULE_DESCRIPTION("Q40/Q60 sound driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/oss/hex2hex.c b/sound/oss/hex2hex.c new file mode 100644 index 00000000..041ef5c5 --- /dev/null +++ b/sound/oss/hex2hex.c @@ -0,0 +1,101 @@ +/* + * hex2hex reads stdin in Intel HEX format and produces an + * (unsigned char) array which contains the bytes and writes it + * to stdout using C syntax + */ + +#include <stdio.h> +#include <string.h> +#include <stdlib.h> + +#define ABANDON(why) { fprintf(stderr, "%s\n", why); exit(1); } +#define MAX_SIZE (256*1024) +unsigned char buf[MAX_SIZE]; + +static int loadhex(FILE *inf, unsigned char *buf) +{ + int l=0, c, i; + + while ((c=getc(inf))!=EOF) + { + if (c == ':') /* Sync with beginning of line */ + { + int n, check; + unsigned char sum; + int addr; + int linetype; + + if (fscanf(inf, "%02x", &n) != 1) + ABANDON("File format error"); + sum = n; + + if (fscanf(inf, "%04x", &addr) != 1) + ABANDON("File format error"); + sum += addr/256; + sum += addr%256; + + if (fscanf(inf, "%02x", &linetype) != 1) + ABANDON("File format error"); + sum += linetype; + + if (linetype != 0) + continue; + + for (i=0;i<n;i++) + { + if (fscanf(inf, "%02x", &c) != 1) + ABANDON("File format error"); + if (addr >= MAX_SIZE) + ABANDON("File too large"); + buf[addr++] = c; + if (addr > l) + l = addr; + sum += c; + } + + if (fscanf(inf, "%02x", &check) != 1) + ABANDON("File format error"); + + sum = ~sum + 1; + if (check != sum) + ABANDON("Line checksum error"); + } + } + + return l; +} + +int main( int argc, const char * argv [] ) +{ + const char * varline; + int i,l; + int id=0; + + if(argv[1] && strcmp(argv[1], "-i")==0) + { + argv++; + argc--; + id=1; + } + if(argv[1]==NULL) + { + fprintf(stderr,"hex2hex: [-i] filename\n"); + exit(1); + } + varline = argv[1]; + l = loadhex(stdin, buf); + + printf("/*\n *\t Computer generated file. Do not edit.\n */\n"); + printf("static int %s_len = %d;\n", varline, l); + printf("static unsigned char %s[] %s = {\n", varline, id?"__initdata":""); + + for (i=0;i<l;i++) + { + if (i) printf(","); + if (i && !(i % 16)) printf("\n"); + printf("0x%02x", buf[i]); + } + + printf("\n};\n\n"); + return 0; +} diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c new file mode 100644 index 00000000..52d06a33 --- /dev/null +++ b/sound/oss/kahlua.c @@ -0,0 +1,231 @@ +/* + * Initialisation code for Cyrix/NatSemi VSA1 softaudio + * + * (C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk> + * + * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. + * The older version (VSA1) provides fairly good soundblaster emulation + * although there are a couple of bugs: large DMA buffers break record, + * and the MPU event handling seems suspect. VSA2 allows the native driver + * to control the AC97 audio engine directly and requires a different driver. + * + * Thanks to National Semiconductor for providing the needed information + * on the XpressAudio(tm) internals. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2, or (at your option) any + * later version. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * TO DO: + * Investigate whether we can portably support Cognac (5520) in the + * same manner. + */ + +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <linux/slab.h> + +#include "sound_config.h" + +#include "sb.h" + +/* + * Read a soundblaster compatible mixer register. + * In this case we are actually reading an SMI trap + * not real hardware. + */ + +static u8 __devinit mixer_read(unsigned long io, u8 reg) +{ + outb(reg, io + 4); + udelay(20); + reg = inb(io + 5); + udelay(20); + return reg; +} + +static int __devinit probe_one(struct pci_dev *pdev, const struct pci_device_id *ent) +{ + struct address_info *hw_config; + unsigned long base; + void __iomem *mem; + unsigned long io; + u16 map; + u8 irq, dma8, dma16; + int oldquiet; + extern int sb_be_quiet; + + base = pci_resource_start(pdev, 0); + if(base == 0UL) + return 1; + + mem = ioremap(base, 128); + if (!mem) + return 1; + map = readw(mem + 0x18); /* Read the SMI enables */ + iounmap(mem); + + /* Map bits + 0:1 * 0x20 + 0x200 = sb base + 2 sb enable + 3 adlib enable + 5 MPU enable 0x330 + 6 MPU enable 0x300 + + The other bits may be used internally so must be masked */ + + io = 0x220 + 0x20 * (map & 3); + + if(map & (1<<2)) + printk(KERN_INFO "kahlua: XpressAudio at 0x%lx\n", io); + else + return 1; + + if(map & (1<<5)) + printk(KERN_INFO "kahlua: MPU at 0x300\n"); + else if(map & (1<<6)) + printk(KERN_INFO "kahlua: MPU at 0x330\n"); + + irq = mixer_read(io, 0x80) & 0x0F; + dma8 = mixer_read(io, 0x81); + + // printk("IRQ=%x MAP=%x DMA=%x\n", irq, map, dma8); + + if(dma8 & 0x20) + dma16 = 5; + else if(dma8 & 0x40) + dma16 = 6; + else if(dma8 & 0x80) + dma16 = 7; + else + { + printk(KERN_ERR "kahlua: No 16bit DMA enabled.\n"); + return 1; + } + + if(dma8 & 0x01) + dma8 = 0; + else if(dma8 & 0x02) + dma8 = 1; + else if(dma8 & 0x08) + dma8 = 3; + else + { + printk(KERN_ERR "kahlua: No 8bit DMA enabled.\n"); + return 1; + } + + if(irq & 1) + irq = 9; + else if(irq & 2) + irq = 5; + else if(irq & 4) + irq = 7; + else if(irq & 8) + irq = 10; + else + { + printk(KERN_ERR "kahlua: SB IRQ not set.\n"); + return 1; + } + + printk(KERN_INFO "kahlua: XpressAudio on IRQ %d, DMA %d, %d\n", + irq, dma8, dma16); + + hw_config = kzalloc(sizeof(struct address_info), GFP_KERNEL); + if(hw_config == NULL) + { + printk(KERN_ERR "kahlua: out of memory.\n"); + return 1; + } + + pci_set_drvdata(pdev, hw_config); + + hw_config->io_base = io; + hw_config->irq = irq; + hw_config->dma = dma8; + hw_config->dma2 = dma16; + hw_config->name = "Cyrix XpressAudio"; + hw_config->driver_use_1 = SB_NO_MIDI | SB_PCI_IRQ; + + if (!request_region(io, 16, "soundblaster")) + goto err_out_free; + + if(sb_dsp_detect(hw_config, 0, 0, NULL)==0) + { + printk(KERN_ERR "kahlua: audio not responding.\n"); + release_region(io, 16); + goto err_out_free; + } + + oldquiet = sb_be_quiet; + sb_be_quiet = 1; + if(sb_dsp_init(hw_config, THIS_MODULE)) + { + sb_be_quiet = oldquiet; + goto err_out_free; + } + sb_be_quiet = oldquiet; + + return 0; + +err_out_free: + pci_set_drvdata(pdev, NULL); + kfree(hw_config); + return 1; +} + +static void __devexit remove_one(struct pci_dev *pdev) +{ + struct address_info *hw_config = pci_get_drvdata(pdev); + sb_dsp_unload(hw_config, 0); + pci_set_drvdata(pdev, NULL); + kfree(hw_config); +} + +MODULE_AUTHOR("Alan Cox"); +MODULE_DESCRIPTION("Kahlua VSA1 PCI Audio"); +MODULE_LICENSE("GPL"); + +/* + * 5530 only. The 5510/5520 decode is different. + */ + +static DEFINE_PCI_DEVICE_TABLE(id_tbl) = { + { PCI_VDEVICE(CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO), 0 }, + { } +}; + +MODULE_DEVICE_TABLE(pci, id_tbl); + +static struct pci_driver kahlua_driver = { + .name = "kahlua", + .id_table = id_tbl, + .probe = probe_one, + .remove = __devexit_p(remove_one), +}; + + +static int __init kahlua_init_module(void) +{ + printk(KERN_INFO "Cyrix Kahlua VSA1 XpressAudio support (c) Copyright 2003 Red Hat Inc\n"); + return pci_register_driver(&kahlua_driver); +} + +static void __devexit kahlua_cleanup_module(void) +{ + pci_unregister_driver(&kahlua_driver); +} + + +module_init(kahlua_init_module); +module_exit(kahlua_cleanup_module); + diff --git a/sound/oss/midi_ctrl.h b/sound/oss/midi_ctrl.h new file mode 100644 index 00000000..3353e5a6 --- /dev/null +++ b/sound/oss/midi_ctrl.h @@ -0,0 +1,22 @@ +static unsigned char ctrl_def_values[128] = +{ + 0x40,0x00,0x40,0x40, 0x40,0x40,0x40,0x7f, /* 0 to 7 */ + 0x40,0x40,0x40,0x7f, 0x40,0x40,0x40,0x40, /* 8 to 15 */ + 0x40,0x40,0x40,0x40, 0x40,0x40,0x40,0x40, /* 16 to 23 */ + 0x40,0x40,0x40,0x40, 0x40,0x40,0x40,0x40, /* 24 to 31 */ + + 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 32 to 39 */ + 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 40 to 47 */ + 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 48 to 55 */ + 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 56 to 63 */ + + 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 64 to 71 */ + 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 72 to 79 */ + 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 80 to 87 */ + 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 88 to 95 */ + + 0x00,0x00,0x7f,0x7f, 0x7f,0x7f,0x00,0x00, /* 96 to 103 */ + 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 104 to 111 */ + 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 112 to 119 */ + 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 120 to 127 */ +}; diff --git a/sound/oss/midi_synth.c b/sound/oss/midi_synth.c new file mode 100644 index 00000000..2292c230 --- /dev/null +++ b/sound/oss/midi_synth.c @@ -0,0 +1,712 @@ +/* + * sound/oss/midi_synth.c + * + * High level midi sequencer manager for dumb MIDI interfaces. + */ +/* + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + */ +/* + * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed) + * Andrew Veliath : fixed running status in MIDI input state machine + */ +#define USE_SEQ_MACROS +#define USE_SIMPLE_MACROS + +#include "sound_config.h" + +#define _MIDI_SYNTH_C_ + +#include "midi_synth.h" + +static int midi2synth[MAX_MIDI_DEV]; +static int sysex_state[MAX_MIDI_DEV] = +{0}; +static unsigned char prev_out_status[MAX_MIDI_DEV]; + +#define STORE(cmd) \ +{ \ + int len; \ + unsigned char obuf[8]; \ + cmd; \ + seq_input_event(obuf, len); \ +} + +#define _seqbuf obuf +#define _seqbufptr 0 +#define _SEQ_ADVBUF(x) len=x + +void +do_midi_msg(int synthno, unsigned char *msg, int mlen) +{ + switch (msg[0] & 0xf0) + { + case 0x90: + if (msg[2] != 0) + { + STORE(SEQ_START_NOTE(synthno, msg[0] & 0x0f, msg[1], msg[2])); + break; + } + msg[2] = 64; + + case 0x80: + STORE(SEQ_STOP_NOTE(synthno, msg[0] & 0x0f, msg[1], msg[2])); + break; + + case 0xA0: + STORE(SEQ_KEY_PRESSURE(synthno, msg[0] & 0x0f, msg[1], msg[2])); + break; + + case 0xB0: + STORE(SEQ_CONTROL(synthno, msg[0] & 0x0f, + msg[1], msg[2])); + break; + + case 0xC0: + STORE(SEQ_SET_PATCH(synthno, msg[0] & 0x0f, msg[1])); + break; + + case 0xD0: + STORE(SEQ_CHN_PRESSURE(synthno, msg[0] & 0x0f, msg[1])); + break; + + case 0xE0: + STORE(SEQ_BENDER(synthno, msg[0] & 0x0f, + (msg[1] & 0x7f) | ((msg[2] & 0x7f) << 7))); + break; + + default: + /* printk( "MPU: Unknown midi channel message %02x\n", msg[0]); */ + ; + } +} +EXPORT_SYMBOL(do_midi_msg); + +static void +midi_outc(int midi_dev, int data) +{ + int timeout; + + for (timeout = 0; timeout < 3200; timeout++) + if (midi_devs[midi_dev]->outputc(midi_dev, (unsigned char) (data & 0xff))) + { + if (data & 0x80) /* + * Status byte + */ + prev_out_status[midi_dev] = + (unsigned char) (data & 0xff); /* + * Store for running status + */ + return; /* + * Mission complete + */ + } + /* + * Sorry! No space on buffers. + */ + printk("Midi send timed out\n"); +} + +static int +prefix_cmd(int midi_dev, unsigned char status) +{ + if ((char *) midi_devs[midi_dev]->prefix_cmd == NULL) + return 1; + + return midi_devs[midi_dev]->prefix_cmd(midi_dev, status); +} + +static void +midi_synth_input(int orig_dev, unsigned char data) +{ + int dev; + struct midi_input_info *inc; + + static unsigned char len_tab[] = /* # of data bytes following a status + */ + { + 2, /* 8x */ + 2, /* 9x */ + 2, /* Ax */ + 2, /* Bx */ + 1, /* Cx */ + 1, /* Dx */ + 2, /* Ex */ + 0 /* Fx */ + }; + + if (orig_dev < 0 || orig_dev > num_midis || midi_devs[orig_dev] == NULL) + return; + + if (data == 0xfe) /* Ignore active sensing */ + return; + + dev = midi2synth[orig_dev]; + inc = &midi_devs[orig_dev]->in_info; + + switch (inc->m_state) + { + case MST_INIT: + if (data & 0x80) /* MIDI status byte */ + { + if ((data & 0xf0) == 0xf0) /* Common message */ + { + switch (data) + { + case 0xf0: /* Sysex */ + inc->m_state = MST_SYSEX; + break; /* Sysex */ + + case 0xf1: /* MTC quarter frame */ + case 0xf3: /* Song select */ + inc->m_state = MST_DATA; + inc->m_ptr = 1; + inc->m_left = 1; + inc->m_buf[0] = data; + break; + + case 0xf2: /* Song position pointer */ + inc->m_state = MST_DATA; + inc->m_ptr = 1; + inc->m_left = 2; + inc->m_buf[0] = data; + break; + + default: + inc->m_buf[0] = data; + inc->m_ptr = 1; + do_midi_msg(dev, inc->m_buf, inc->m_ptr); + inc->m_ptr = 0; + inc->m_left = 0; + } + } else + { + inc->m_state = MST_DATA; + inc->m_ptr = 1; + inc->m_left = len_tab[(data >> 4) - 8]; + inc->m_buf[0] = inc->m_prev_status = data; + } + } else if (inc->m_prev_status & 0x80) { + /* Data byte (use running status) */ + inc->m_ptr = 2; + inc->m_buf[1] = data; + inc->m_buf[0] = inc->m_prev_status; + inc->m_left = len_tab[(inc->m_buf[0] >> 4) - 8] - 1; + if (inc->m_left > 0) + inc->m_state = MST_DATA; /* Not done yet */ + else { + inc->m_state = MST_INIT; + do_midi_msg(dev, inc->m_buf, inc->m_ptr); + inc->m_ptr = 0; + } + } + break; /* MST_INIT */ + + case MST_DATA: + inc->m_buf[inc->m_ptr++] = data; + if (--inc->m_left <= 0) + { + inc->m_state = MST_INIT; + do_midi_msg(dev, inc->m_buf, inc->m_ptr); + inc->m_ptr = 0; + } + break; /* MST_DATA */ + + case MST_SYSEX: + if (data == 0xf7) /* Sysex end */ + { + inc->m_state = MST_INIT; + inc->m_left = 0; + inc->m_ptr = 0; + } + break; /* MST_SYSEX */ + + default: + printk("MIDI%d: Unexpected state %d (%02x)\n", orig_dev, inc->m_state, (int) data); + inc->m_state = MST_INIT; + } +} + +static void +leave_sysex(int dev) +{ + int orig_dev = synth_devs[dev]->midi_dev; + int timeout = 0; + + if (!sysex_state[dev]) + return; + + sysex_state[dev] = 0; + + while (!midi_devs[orig_dev]->outputc(orig_dev, 0xf7) && + timeout < 1000) + timeout++; + + sysex_state[dev] = 0; +} + +static void +midi_synth_output(int dev) +{ + /* + * Currently NOP + */ +} + +int midi_synth_ioctl(int dev, unsigned int cmd, void __user *arg) +{ + /* + * int orig_dev = synth_devs[dev]->midi_dev; + */ + + switch (cmd) { + + case SNDCTL_SYNTH_INFO: + if (__copy_to_user(arg, synth_devs[dev]->info, sizeof(struct synth_info))) + return -EFAULT; + return 0; + + case SNDCTL_SYNTH_MEMAVL: + return 0x7fffffff; + + default: + return -EINVAL; + } +} +EXPORT_SYMBOL(midi_synth_ioctl); + +int +midi_synth_kill_note(int dev, int channel, int note, int velocity) +{ + int orig_dev = synth_devs[dev]->midi_dev; + int msg, chn; + + if (note < 0 || note > 127) + return 0; + if (channel < 0 || channel > 15) + return 0; + if (velocity < 0) + velocity = 0; + if (velocity > 127) + velocity = 127; + + leave_sysex(dev); + + msg = prev_out_status[orig_dev] & 0xf0; + chn = prev_out_status[orig_dev] & 0x0f; + + if (chn == channel && ((msg == 0x90 && velocity == 64) || msg == 0x80)) + { /* + * Use running status + */ + if (!prefix_cmd(orig_dev, note)) + return 0; + + midi_outc(orig_dev, note); + + if (msg == 0x90) /* + * Running status = Note on + */ + midi_outc(orig_dev, 0); /* + * Note on with velocity 0 == note + * off + */ + else + midi_outc(orig_dev, velocity); + } else + { + if (velocity == 64) + { + if (!prefix_cmd(orig_dev, 0x90 | (channel & 0x0f))) + return 0; + midi_outc(orig_dev, 0x90 | (channel & 0x0f)); /* + * Note on + */ + midi_outc(orig_dev, note); + midi_outc(orig_dev, 0); /* + * Zero G + */ + } else + { + if (!prefix_cmd(orig_dev, 0x80 | (channel & 0x0f))) + return 0; + midi_outc(orig_dev, 0x80 | (channel & 0x0f)); /* + * Note off + */ + midi_outc(orig_dev, note); + midi_outc(orig_dev, velocity); + } + } + + return 0; +} +EXPORT_SYMBOL(midi_synth_kill_note); + +int +midi_synth_set_instr(int dev, int channel, int instr_no) +{ + int orig_dev = synth_devs[dev]->midi_dev; + + if (instr_no < 0 || instr_no > 127) + instr_no = 0; + if (channel < 0 || channel > 15) + return 0; + + leave_sysex(dev); + + if (!prefix_cmd(orig_dev, 0xc0 | (channel & 0x0f))) + return 0; + midi_outc(orig_dev, 0xc0 | (channel & 0x0f)); /* + * Program change + */ + midi_outc(orig_dev, instr_no); + + return 0; +} +EXPORT_SYMBOL(midi_synth_set_instr); + +int +midi_synth_start_note(int dev, int channel, int note, int velocity) +{ + int orig_dev = synth_devs[dev]->midi_dev; + int msg, chn; + + if (note < 0 || note > 127) + return 0; + if (channel < 0 || channel > 15) + return 0; + if (velocity < 0) + velocity = 0; + if (velocity > 127) + velocity = 127; + + leave_sysex(dev); + + msg = prev_out_status[orig_dev] & 0xf0; + chn = prev_out_status[orig_dev] & 0x0f; + + if (chn == channel && msg == 0x90) + { /* + * Use running status + */ + if (!prefix_cmd(orig_dev, note)) + return 0; + midi_outc(orig_dev, note); + midi_outc(orig_dev, velocity); + } else + { + if (!prefix_cmd(orig_dev, 0x90 | (channel & 0x0f))) + return 0; + midi_outc(orig_dev, 0x90 | (channel & 0x0f)); /* + * Note on + */ + midi_outc(orig_dev, note); + midi_outc(orig_dev, velocity); + } + return 0; +} +EXPORT_SYMBOL(midi_synth_start_note); + +void +midi_synth_reset(int dev) +{ + + leave_sysex(dev); +} +EXPORT_SYMBOL(midi_synth_reset); + +int +midi_synth_open(int dev, int mode) +{ + int orig_dev = synth_devs[dev]->midi_dev; + int err; + struct midi_input_info *inc; + + if (orig_dev < 0 || orig_dev >= num_midis || midi_devs[orig_dev] == NULL) + return -ENXIO; + + midi2synth[orig_dev] = dev; + sysex_state[dev] = 0; + prev_out_status[orig_dev] = 0; + + if ((err = midi_devs[orig_dev]->open(orig_dev, mode, + midi_synth_input, midi_synth_output)) < 0) + return err; + inc = &midi_devs[orig_dev]->in_info; + + /* save_flags(flags); + cli(); + don't know against what irqhandler to protect*/ + inc->m_busy = 0; + inc->m_state = MST_INIT; + inc->m_ptr = 0; + inc->m_left = 0; + inc->m_prev_status = 0x00; + /* restore_flags(flags); */ + + return 1; +} +EXPORT_SYMBOL(midi_synth_open); + +void +midi_synth_close(int dev) +{ + int orig_dev = synth_devs[dev]->midi_dev; + + leave_sysex(dev); + + /* + * Shut up the synths by sending just single active sensing message. + */ + midi_devs[orig_dev]->outputc(orig_dev, 0xfe); + + midi_devs[orig_dev]->close(orig_dev); +} +EXPORT_SYMBOL(midi_synth_close); + +void +midi_synth_hw_control(int dev, unsigned char *event) +{ +} +EXPORT_SYMBOL(midi_synth_hw_control); + +int +midi_synth_load_patch(int dev, int format, const char __user *addr, + int count, int pmgr_flag) +{ + int orig_dev = synth_devs[dev]->midi_dev; + + struct sysex_info sysex; + int i; + unsigned long left, src_offs, eox_seen = 0; + int first_byte = 1; + int hdr_size = (unsigned long) &sysex.data[0] - (unsigned long) &sysex; + + leave_sysex(dev); + + if (!prefix_cmd(orig_dev, 0xf0)) + return 0; + + /* Invalid patch format */ + if (format != SYSEX_PATCH) + return -EINVAL; + + /* Patch header too short */ + if (count < hdr_size) + return -EINVAL; + + count -= hdr_size; + + /* + * Copy the header from user space + */ + + if (copy_from_user(&sysex, addr, hdr_size)) + return -EFAULT; + + /* Sysex record too short */ + if ((unsigned)count < (unsigned)sysex.len) + sysex.len = count; + + left = sysex.len; + src_offs = 0; + + for (i = 0; i < left && !signal_pending(current); i++) + { + unsigned char data; + + if (get_user(data, + (unsigned char __user *)(addr + hdr_size + i))) + return -EFAULT; + + eox_seen = (i > 0 && data & 0x80); /* End of sysex */ + + if (eox_seen && data != 0xf7) + data = 0xf7; + + if (i == 0) + { + if (data != 0xf0) + { + printk(KERN_WARNING "midi_synth: Sysex start missing\n"); + return -EINVAL; + } + } + while (!midi_devs[orig_dev]->outputc(orig_dev, (unsigned char) (data & 0xff)) && + !signal_pending(current)) + schedule(); + + if (!first_byte && data & 0x80) + return 0; + first_byte = 0; + } + + if (!eox_seen) + midi_outc(orig_dev, 0xf7); + return 0; +} +EXPORT_SYMBOL(midi_synth_load_patch); + +void midi_synth_panning(int dev, int channel, int pressure) +{ +} +EXPORT_SYMBOL(midi_synth_panning); + +void midi_synth_aftertouch(int dev, int channel, int pressure) +{ + int orig_dev = synth_devs[dev]->midi_dev; + int msg, chn; + + if (pressure < 0 || pressure > 127) + return; + if (channel < 0 || channel > 15) + return; + + leave_sysex(dev); + + msg = prev_out_status[orig_dev] & 0xf0; + chn = prev_out_status[orig_dev] & 0x0f; + + if (msg != 0xd0 || chn != channel) /* + * Test for running status + */ + { + if (!prefix_cmd(orig_dev, 0xd0 | (channel & 0x0f))) + return; + midi_outc(orig_dev, 0xd0 | (channel & 0x0f)); /* + * Channel pressure + */ + } else if (!prefix_cmd(orig_dev, pressure)) + return; + + midi_outc(orig_dev, pressure); +} +EXPORT_SYMBOL(midi_synth_aftertouch); + +void +midi_synth_controller(int dev, int channel, int ctrl_num, int value) +{ + int orig_dev = synth_devs[dev]->midi_dev; + int chn, msg; + + if (ctrl_num < 0 || ctrl_num > 127) + return; + if (channel < 0 || channel > 15) + return; + + leave_sysex(dev); + + msg = prev_out_status[orig_dev] & 0xf0; + chn = prev_out_status[orig_dev] & 0x0f; + + if (msg != 0xb0 || chn != channel) + { + if (!prefix_cmd(orig_dev, 0xb0 | (channel & 0x0f))) + return; + midi_outc(orig_dev, 0xb0 | (channel & 0x0f)); + } else if (!prefix_cmd(orig_dev, ctrl_num)) + return; + + midi_outc(orig_dev, ctrl_num); + midi_outc(orig_dev, value & 0x7f); +} +EXPORT_SYMBOL(midi_synth_controller); + +void +midi_synth_bender(int dev, int channel, int value) +{ + int orig_dev = synth_devs[dev]->midi_dev; + int msg, prev_chn; + + if (channel < 0 || channel > 15) + return; + + if (value < 0 || value > 16383) + return; + + leave_sysex(dev); + + msg = prev_out_status[orig_dev] & 0xf0; + prev_chn = prev_out_status[orig_dev] & 0x0f; + + if (msg != 0xd0 || prev_chn != channel) /* + * Test for running status + */ + { + if (!prefix_cmd(orig_dev, 0xe0 | (channel & 0x0f))) + return; + midi_outc(orig_dev, 0xe0 | (channel & 0x0f)); + } else if (!prefix_cmd(orig_dev, value & 0x7f)) + return; + + midi_outc(orig_dev, value & 0x7f); + midi_outc(orig_dev, (value >> 7) & 0x7f); +} +EXPORT_SYMBOL(midi_synth_bender); + +void +midi_synth_setup_voice(int dev, int voice, int channel) +{ +} +EXPORT_SYMBOL(midi_synth_setup_voice); + +int +midi_synth_send_sysex(int dev, unsigned char *bytes, int len) +{ + int orig_dev = synth_devs[dev]->midi_dev; + int i; + + for (i = 0; i < len; i++) + { + switch (bytes[i]) + { + case 0xf0: /* Start sysex */ + if (!prefix_cmd(orig_dev, 0xf0)) + return 0; + sysex_state[dev] = 1; + break; + + case 0xf7: /* End sysex */ + if (!sysex_state[dev]) /* Orphan sysex end */ + return 0; + sysex_state[dev] = 0; + break; + + default: + if (!sysex_state[dev]) + return 0; + + if (bytes[i] & 0x80) /* Error. Another message before sysex end */ + { + bytes[i] = 0xf7; /* Sysex end */ + sysex_state[dev] = 0; + } + } + + if (!midi_devs[orig_dev]->outputc(orig_dev, bytes[i])) + { +/* + * Hardware level buffer is full. Abort the sysex message. + */ + + int timeout = 0; + + bytes[i] = 0xf7; + sysex_state[dev] = 0; + + while (!midi_devs[orig_dev]->outputc(orig_dev, bytes[i]) && + timeout < 1000) + timeout++; + } + if (!sysex_state[dev]) + return 0; + } + + return 0; +} +EXPORT_SYMBOL(midi_synth_send_sysex); + diff --git a/sound/oss/midi_synth.h b/sound/oss/midi_synth.h new file mode 100644 index 00000000..b64ddd6c --- /dev/null +++ b/sound/oss/midi_synth.h @@ -0,0 +1,47 @@ +int midi_synth_ioctl (int dev, + unsigned int cmd, void __user * arg); +int midi_synth_kill_note (int dev, int channel, int note, int velocity); +int midi_synth_set_instr (int dev, int channel, int instr_no); +int midi_synth_start_note (int dev, int channel, int note, int volume); +void midi_synth_reset (int dev); +int midi_synth_open (int dev, int mode); +void midi_synth_close (int dev); +void midi_synth_hw_control (int dev, unsigned char *event); +int midi_synth_load_patch (int dev, int format, const char __user * addr, + int count, int pmgr_flag); +void midi_synth_panning (int dev, int channel, int pressure); +void midi_synth_aftertouch (int dev, int channel, int pressure); +void midi_synth_controller (int dev, int channel, int ctrl_num, int value); +void midi_synth_bender (int dev, int chn, int value); +void midi_synth_setup_voice (int dev, int voice, int chn); +int midi_synth_send_sysex(int dev, unsigned char *bytes,int len); + +#ifndef _MIDI_SYNTH_C_ +static struct synth_info std_synth_info = +{MIDI_SYNTH_NAME, 0, SYNTH_TYPE_MIDI, 0, 0, 128, 0, 128, MIDI_SYNTH_CAPS}; + +static struct synth_operations std_midi_synth = +{ + .owner = THIS_MODULE, + .id = "MIDI", + .info = &std_synth_info, + .midi_dev = 0, + .synth_type = SYNTH_TYPE_MIDI, + .synth_subtype = 0, + .open = midi_synth_open, + .close = midi_synth_close, + .ioctl = midi_synth_ioctl, + .kill_note = midi_synth_kill_note, + .start_note = midi_synth_start_note, + .set_instr = midi_synth_set_instr, + .reset = midi_synth_reset, + .hw_control = midi_synth_hw_control, + .load_patch = midi_synth_load_patch, + .aftertouch = midi_synth_aftertouch, + .controller = midi_synth_controller, + .panning = midi_synth_panning, + .bender = midi_synth_bender, + .setup_voice = midi_synth_setup_voice, + .send_sysex = midi_synth_send_sysex +}; +#endif diff --git a/sound/oss/midibuf.c b/sound/oss/midibuf.c new file mode 100644 index 00000000..8cdb2cfe --- /dev/null +++ b/sound/oss/midibuf.c @@ -0,0 +1,425 @@ +/* + * sound/oss/midibuf.c + * + * Device file manager for /dev/midi# + */ +/* + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + */ +/* + * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed) + */ +#include <linux/stddef.h> +#include <linux/kmod.h> +#include <linux/spinlock.h> +#define MIDIBUF_C + +#include "sound_config.h" + + +/* + * Don't make MAX_QUEUE_SIZE larger than 4000 + */ + +#define MAX_QUEUE_SIZE 4000 + +static wait_queue_head_t midi_sleeper[MAX_MIDI_DEV]; +static wait_queue_head_t input_sleeper[MAX_MIDI_DEV]; + +struct midi_buf +{ + int len, head, tail; + unsigned char queue[MAX_QUEUE_SIZE]; +}; + +struct midi_parms +{ + long prech_timeout; /* + * Timeout before the first ch + */ +}; + +static struct midi_buf *midi_out_buf[MAX_MIDI_DEV] = {NULL}; +static struct midi_buf *midi_in_buf[MAX_MIDI_DEV] = {NULL}; +static struct midi_parms parms[MAX_MIDI_DEV]; + +static void midi_poll(unsigned long dummy); + + +static DEFINE_TIMER(poll_timer, midi_poll, 0, 0); + +static volatile int open_devs; +static DEFINE_SPINLOCK(lock); + +#define DATA_AVAIL(q) (q->len) +#define SPACE_AVAIL(q) (MAX_QUEUE_SIZE - q->len) + +#define QUEUE_BYTE(q, data) \ + if (SPACE_AVAIL(q)) \ + { \ + unsigned long flags; \ + spin_lock_irqsave(&lock, flags); \ + q->queue[q->tail] = (data); \ + q->len++; q->tail = (q->tail+1) % MAX_QUEUE_SIZE; \ + spin_unlock_irqrestore(&lock, flags); \ + } + +#define REMOVE_BYTE(q, data) \ + if (DATA_AVAIL(q)) \ + { \ + unsigned long flags; \ + spin_lock_irqsave(&lock, flags); \ + data = q->queue[q->head]; \ + q->len--; q->head = (q->head+1) % MAX_QUEUE_SIZE; \ + spin_unlock_irqrestore(&lock, flags); \ + } + +static void drain_midi_queue(int dev) +{ + + /* + * Give the Midi driver time to drain its output queues + */ + + if (midi_devs[dev]->buffer_status != NULL) + while (!signal_pending(current) && midi_devs[dev]->buffer_status(dev)) + interruptible_sleep_on_timeout(&midi_sleeper[dev], + HZ/10); +} + +static void midi_input_intr(int dev, unsigned char data) +{ + if (midi_in_buf[dev] == NULL) + return; + + if (data == 0xfe) /* + * Active sensing + */ + return; /* + * Ignore + */ + + if (SPACE_AVAIL(midi_in_buf[dev])) { + QUEUE_BYTE(midi_in_buf[dev], data); + wake_up(&input_sleeper[dev]); + } +} + +static void midi_output_intr(int dev) +{ + /* + * Currently NOP + */ +} + +static void midi_poll(unsigned long dummy) +{ + unsigned long flags; + int dev; + + spin_lock_irqsave(&lock, flags); + if (open_devs) + { + for (dev = 0; dev < num_midis; dev++) + if (midi_devs[dev] != NULL && midi_out_buf[dev] != NULL) + { + while (DATA_AVAIL(midi_out_buf[dev])) + { + int ok; + int c = midi_out_buf[dev]->queue[midi_out_buf[dev]->head]; + + spin_unlock_irqrestore(&lock,flags);/* Give some time to others */ + ok = midi_devs[dev]->outputc(dev, c); + spin_lock_irqsave(&lock, flags); + if (!ok) + break; + midi_out_buf[dev]->head = (midi_out_buf[dev]->head + 1) % MAX_QUEUE_SIZE; + midi_out_buf[dev]->len--; + } + + if (DATA_AVAIL(midi_out_buf[dev]) < 100) + wake_up(&midi_sleeper[dev]); + } + poll_timer.expires = (1) + jiffies; + add_timer(&poll_timer); + /* + * Come back later + */ + } + spin_unlock_irqrestore(&lock, flags); +} + +int MIDIbuf_open(int dev, struct file *file) +{ + int mode, err; + + dev = dev >> 4; + mode = translate_mode(file); + + if (num_midis > MAX_MIDI_DEV) + { + printk(KERN_ERR "midi: Too many midi interfaces\n"); + num_midis = MAX_MIDI_DEV; + } + if (dev < 0 || dev >= num_midis || midi_devs[dev] == NULL) + return -ENXIO; + /* + * Interrupts disabled. Be careful + */ + + module_put(midi_devs[dev]->owner); + + if ((err = midi_devs[dev]->open(dev, mode, + midi_input_intr, midi_output_intr)) < 0) + return err; + + parms[dev].prech_timeout = MAX_SCHEDULE_TIMEOUT; + midi_in_buf[dev] = vmalloc(sizeof(struct midi_buf)); + + if (midi_in_buf[dev] == NULL) + { + printk(KERN_WARNING "midi: Can't allocate buffer\n"); + midi_devs[dev]->close(dev); + return -EIO; + } + midi_in_buf[dev]->len = midi_in_buf[dev]->head = midi_in_buf[dev]->tail = 0; + + midi_out_buf[dev] = vmalloc(sizeof(struct midi_buf)); + + if (midi_out_buf[dev] == NULL) + { + printk(KERN_WARNING "midi: Can't allocate buffer\n"); + midi_devs[dev]->close(dev); + vfree(midi_in_buf[dev]); + midi_in_buf[dev] = NULL; + return -EIO; + } + midi_out_buf[dev]->len = midi_out_buf[dev]->head = midi_out_buf[dev]->tail = 0; + open_devs++; + + init_waitqueue_head(&midi_sleeper[dev]); + init_waitqueue_head(&input_sleeper[dev]); + + if (open_devs < 2) /* This was first open */ + { + poll_timer.expires = 1 + jiffies; + add_timer(&poll_timer); /* Start polling */ + } + return err; +} + +void MIDIbuf_release(int dev, struct file *file) +{ + int mode; + + dev = dev >> 4; + mode = translate_mode(file); + + if (dev < 0 || dev >= num_midis || midi_devs[dev] == NULL) + return; + + /* + * Wait until the queue is empty + */ + + if (mode != OPEN_READ) + { + midi_devs[dev]->outputc(dev, 0xfe); /* + * Active sensing to shut the + * devices + */ + + while (!signal_pending(current) && DATA_AVAIL(midi_out_buf[dev])) + interruptible_sleep_on(&midi_sleeper[dev]); + /* + * Sync + */ + + drain_midi_queue(dev); /* + * Ensure the output queues are empty + */ + } + + midi_devs[dev]->close(dev); + + open_devs--; + if (open_devs == 0) + del_timer_sync(&poll_timer); + vfree(midi_in_buf[dev]); + vfree(midi_out_buf[dev]); + midi_in_buf[dev] = NULL; + midi_out_buf[dev] = NULL; + + module_put(midi_devs[dev]->owner); +} + +int MIDIbuf_write(int dev, struct file *file, const char __user *buf, int count) +{ + int c, n, i; + unsigned char tmp_data; + + dev = dev >> 4; + + if (!count) + return 0; + + c = 0; + + while (c < count) + { + n = SPACE_AVAIL(midi_out_buf[dev]); + + if (n == 0) { /* + * No space just now. + */ + + if (file->f_flags & O_NONBLOCK) { + c = -EAGAIN; + goto out; + } + + interruptible_sleep_on(&midi_sleeper[dev]); + if (signal_pending(current)) + { + c = -EINTR; + goto out; + } + n = SPACE_AVAIL(midi_out_buf[dev]); + } + if (n > (count - c)) + n = count - c; + + for (i = 0; i < n; i++) + { + /* BROKE BROKE BROKE - CAN'T DO THIS WITH CLI !! */ + /* yes, think the same, so I removed the cli() brackets + QUEUE_BYTE is protected against interrupts */ + if (copy_from_user((char *) &tmp_data, &(buf)[c], 1)) { + c = -EFAULT; + goto out; + } + QUEUE_BYTE(midi_out_buf[dev], tmp_data); + c++; + } + } +out: + return c; +} + + +int MIDIbuf_read(int dev, struct file *file, char __user *buf, int count) +{ + int n, c = 0; + unsigned char tmp_data; + + dev = dev >> 4; + + if (!DATA_AVAIL(midi_in_buf[dev])) { /* + * No data yet, wait + */ + if (file->f_flags & O_NONBLOCK) { + c = -EAGAIN; + goto out; + } + interruptible_sleep_on_timeout(&input_sleeper[dev], + parms[dev].prech_timeout); + + if (signal_pending(current)) + c = -EINTR; /* The user is getting restless */ + } + if (c == 0 && DATA_AVAIL(midi_in_buf[dev])) /* + * Got some bytes + */ + { + n = DATA_AVAIL(midi_in_buf[dev]); + if (n > count) + n = count; + c = 0; + + while (c < n) + { + char *fixit; + REMOVE_BYTE(midi_in_buf[dev], tmp_data); + fixit = (char *) &tmp_data; + /* BROKE BROKE BROKE */ + /* yes removed the cli() brackets again + should q->len,tail&head be atomic_t? */ + if (copy_to_user(&(buf)[c], fixit, 1)) { + c = -EFAULT; + goto out; + } + c++; + } + } +out: + return c; +} + +int MIDIbuf_ioctl(int dev, struct file *file, + unsigned int cmd, void __user *arg) +{ + int val; + + dev = dev >> 4; + + if (((cmd >> 8) & 0xff) == 'C') + { + if (midi_devs[dev]->coproc) /* Coprocessor ioctl */ + return midi_devs[dev]->coproc->ioctl(midi_devs[dev]->coproc->devc, cmd, arg, 0); +/* printk("/dev/midi%d: No coprocessor for this device\n", dev);*/ + return -ENXIO; + } + else + { + switch (cmd) + { + case SNDCTL_MIDI_PRETIME: + if (get_user(val, (int __user *)arg)) + return -EFAULT; + if (val < 0) + val = 0; + val = (HZ * val) / 10; + parms[dev].prech_timeout = val; + return put_user(val, (int __user *)arg); + + default: + if (!midi_devs[dev]->ioctl) + return -EINVAL; + return midi_devs[dev]->ioctl(dev, cmd, arg); + } + } +} + +/* No kernel lock - fine */ +unsigned int MIDIbuf_poll(int dev, struct file *file, poll_table * wait) +{ + unsigned int mask = 0; + + dev = dev >> 4; + + /* input */ + poll_wait(file, &input_sleeper[dev], wait); + if (DATA_AVAIL(midi_in_buf[dev])) + mask |= POLLIN | POLLRDNORM; + + /* output */ + poll_wait(file, &midi_sleeper[dev], wait); + if (!SPACE_AVAIL(midi_out_buf[dev])) + mask |= POLLOUT | POLLWRNORM; + + return mask; +} + + +int MIDIbuf_avail(int dev) +{ + if (midi_in_buf[dev]) + return DATA_AVAIL (midi_in_buf[dev]); + return 0; +} +EXPORT_SYMBOL(MIDIbuf_avail); + diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c new file mode 100644 index 00000000..25e4609f --- /dev/null +++ b/sound/oss/mpu401.c @@ -0,0 +1,1806 @@ +/* + * sound/oss/mpu401.c + * + * The low level driver for Roland MPU-401 compatible Midi cards. + */ +/* + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * + * + * Thomas Sailer ioctl code reworked (vmalloc/vfree removed) + * Alan Cox modularisation, use normal request_irq, use dev_id + * Bartlomiej Zolnierkiewicz removed some __init to allow using many drivers + * Chris Rankin Update the module-usage counter for the coprocessor + * Zwane Mwaikambo Changed attach/unload resource freeing + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/spinlock.h> +#define USE_SEQ_MACROS +#define USE_SIMPLE_MACROS + +#include "sound_config.h" + +#include "coproc.h" +#include "mpu401.h" + +static int timer_mode = TMR_INTERNAL, timer_caps = TMR_INTERNAL; + +struct mpu_config +{ + int base; /* + * I/O base + */ + int irq; + int opened; /* + * Open mode + */ + int devno; + int synthno; + int uart_mode; + int initialized; + int mode; +#define MODE_MIDI 1 +#define MODE_SYNTH 2 + unsigned char version, revision; + unsigned int capabilities; +#define MPU_CAP_INTLG 0x10000000 +#define MPU_CAP_SYNC 0x00000010 +#define MPU_CAP_FSK 0x00000020 +#define MPU_CAP_CLS 0x00000040 +#define MPU_CAP_SMPTE 0x00000080 +#define MPU_CAP_2PORT 0x00000001 + int timer_flag; + +#define MBUF_MAX 10 +#define BUFTEST(dc) if (dc->m_ptr >= MBUF_MAX || dc->m_ptr < 0) \ + {printk( "MPU: Invalid buffer pointer %d/%d, s=%d\n", dc->m_ptr, dc->m_left, dc->m_state);dc->m_ptr--;} + int m_busy; + unsigned char m_buf[MBUF_MAX]; + int m_ptr; + int m_state; + int m_left; + unsigned char last_status; + void (*inputintr) (int dev, unsigned char data); + int shared_irq; + int *osp; + spinlock_t lock; + }; + +#define DATAPORT(base) (base) +#define COMDPORT(base) (base+1) +#define STATPORT(base) (base+1) + + +static void mpu401_close(int dev); + +static inline int mpu401_status(struct mpu_config *devc) +{ + return inb(STATPORT(devc->base)); +} + +#define input_avail(devc) (!(mpu401_status(devc)&INPUT_AVAIL)) +#define output_ready(devc) (!(mpu401_status(devc)&OUTPUT_READY)) + +static inline void write_command(struct mpu_config *devc, unsigned char cmd) +{ + outb(cmd, COMDPORT(devc->base)); +} + +static inline int read_data(struct mpu_config *devc) +{ + return inb(DATAPORT(devc->base)); +} + +static inline void write_data(struct mpu_config *devc, unsigned char byte) +{ + outb(byte, DATAPORT(devc->base)); +} + +#define OUTPUT_READY 0x40 +#define INPUT_AVAIL 0x80 +#define MPU_ACK 0xFE +#define MPU_RESET 0xFF +#define UART_MODE_ON 0x3F + +static struct mpu_config dev_conf[MAX_MIDI_DEV]; + +static int n_mpu_devs; + +static int reset_mpu401(struct mpu_config *devc); +static void set_uart_mode(int dev, struct mpu_config *devc, int arg); + +static int mpu_timer_init(int midi_dev); +static void mpu_timer_interrupt(void); +static void timer_ext_event(struct mpu_config *devc, int event, int parm); + +static struct synth_info mpu_synth_info_proto = { + "MPU-401 MIDI interface", + 0, + SYNTH_TYPE_MIDI, + MIDI_TYPE_MPU401, + 0, 128, + 0, 128, + SYNTH_CAP_INPUT +}; + +static struct synth_info mpu_synth_info[MAX_MIDI_DEV]; + +/* + * States for the input scanner + */ + +#define ST_INIT 0 /* Ready for timing byte or msg */ +#define ST_TIMED 1 /* Leading timing byte rcvd */ +#define ST_DATABYTE 2 /* Waiting for (nr_left) data bytes */ + +#define ST_SYSMSG 100 /* System message (sysx etc). */ +#define ST_SYSEX 101 /* System exclusive msg */ +#define ST_MTC 102 /* Midi Time Code (MTC) qframe msg */ +#define ST_SONGSEL 103 /* Song select */ +#define ST_SONGPOS 104 /* Song position pointer */ + +static unsigned char len_tab[] = /* # of data bytes following a status + */ +{ + 2, /* 8x */ + 2, /* 9x */ + 2, /* Ax */ + 2, /* Bx */ + 1, /* Cx */ + 1, /* Dx */ + 2, /* Ex */ + 0 /* Fx */ +}; + +#define STORE(cmd) \ +{ \ + int len; \ + unsigned char obuf[8]; \ + cmd; \ + seq_input_event(obuf, len); \ +} + +#define _seqbuf obuf +#define _seqbufptr 0 +#define _SEQ_ADVBUF(x) len=x + +static int mpu_input_scanner(struct mpu_config *devc, unsigned char midic) +{ + + switch (devc->m_state) + { + case ST_INIT: + switch (midic) + { + case 0xf8: + /* Timer overflow */ + break; + + case 0xfc: + printk("<all end>"); + break; + + case 0xfd: + if (devc->timer_flag) + mpu_timer_interrupt(); + break; + + case 0xfe: + return MPU_ACK; + + case 0xf0: + case 0xf1: + case 0xf2: + case 0xf3: + case 0xf4: + case 0xf5: + case 0xf6: + case 0xf7: + printk("<Trk data rq #%d>", midic & 0x0f); + break; + + case 0xf9: + printk("<conductor rq>"); + break; + + case 0xff: + devc->m_state = ST_SYSMSG; + break; + + default: + if (midic <= 0xef) + { + /* printk( "mpu time: %d ", midic); */ + devc->m_state = ST_TIMED; + } + else + printk("<MPU: Unknown event %02x> ", midic); + } + break; + + case ST_TIMED: + { + int msg = ((int) (midic & 0xf0) >> 4); + + devc->m_state = ST_DATABYTE; + + if (msg < 8) /* Data byte */ + { + /* printk( "midi msg (running status) "); */ + msg = ((int) (devc->last_status & 0xf0) >> 4); + msg -= 8; + devc->m_left = len_tab[msg] - 1; + + devc->m_ptr = 2; + devc->m_buf[0] = devc->last_status; + devc->m_buf[1] = midic; + + if (devc->m_left <= 0) + { + devc->m_state = ST_INIT; + do_midi_msg(devc->synthno, devc->m_buf, devc->m_ptr); + devc->m_ptr = 0; + } + } + else if (msg == 0xf) /* MPU MARK */ + { + devc->m_state = ST_INIT; + + switch (midic) + { + case 0xf8: + /* printk( "NOP "); */ + break; + + case 0xf9: + /* printk( "meas end "); */ + break; + + case 0xfc: + /* printk( "data end "); */ + break; + + default: + printk("Unknown MPU mark %02x\n", midic); + } + } + else + { + devc->last_status = midic; + /* printk( "midi msg "); */ + msg -= 8; + devc->m_left = len_tab[msg]; + + devc->m_ptr = 1; + devc->m_buf[0] = midic; + + if (devc->m_left <= 0) + { + devc->m_state = ST_INIT; + do_midi_msg(devc->synthno, devc->m_buf, devc->m_ptr); + devc->m_ptr = 0; + } + } + } + break; + + case ST_SYSMSG: + switch (midic) + { + case 0xf0: + printk("<SYX>"); + devc->m_state = ST_SYSEX; + break; + + case 0xf1: + devc->m_state = ST_MTC; + break; + + case 0xf2: + devc->m_state = ST_SONGPOS; + devc->m_ptr = 0; + break; + + case 0xf3: + devc->m_state = ST_SONGSEL; + break; + + case 0xf6: + /* printk( "tune_request\n"); */ + devc->m_state = ST_INIT; + + /* + * Real time messages + */ + case 0xf8: + /* midi clock */ + devc->m_state = ST_INIT; + timer_ext_event(devc, TMR_CLOCK, 0); + break; + + case 0xfA: + devc->m_state = ST_INIT; + timer_ext_event(devc, TMR_START, 0); + break; + + case 0xFB: + devc->m_state = ST_INIT; + timer_ext_event(devc, TMR_CONTINUE, 0); + break; + + case 0xFC: + devc->m_state = ST_INIT; + timer_ext_event(devc, TMR_STOP, 0); + break; + + case 0xFE: + /* active sensing */ + devc->m_state = ST_INIT; + break; + + case 0xff: + /* printk( "midi hard reset"); */ + devc->m_state = ST_INIT; + break; + + default: + printk("unknown MIDI sysmsg %0x\n", midic); + devc->m_state = ST_INIT; + } + break; + + case ST_MTC: + devc->m_state = ST_INIT; + printk("MTC frame %x02\n", midic); + break; + + case ST_SYSEX: + if (midic == 0xf7) + { + printk("<EOX>"); + devc->m_state = ST_INIT; + } + else + printk("%02x ", midic); + break; + + case ST_SONGPOS: + BUFTEST(devc); + devc->m_buf[devc->m_ptr++] = midic; + if (devc->m_ptr == 2) + { + devc->m_state = ST_INIT; + devc->m_ptr = 0; + timer_ext_event(devc, TMR_SPP, + ((devc->m_buf[1] & 0x7f) << 7) | + (devc->m_buf[0] & 0x7f)); + } + break; + + case ST_DATABYTE: + BUFTEST(devc); + devc->m_buf[devc->m_ptr++] = midic; + if ((--devc->m_left) <= 0) + { + devc->m_state = ST_INIT; + do_midi_msg(devc->synthno, devc->m_buf, devc->m_ptr); + devc->m_ptr = 0; + } + break; + + default: + printk("Bad state %d ", devc->m_state); + devc->m_state = ST_INIT; + } + return 1; +} + +static void mpu401_input_loop(struct mpu_config *devc) +{ + unsigned long flags; + int busy; + int n; + + spin_lock_irqsave(&devc->lock,flags); + busy = devc->m_busy; + devc->m_busy = 1; + spin_unlock_irqrestore(&devc->lock,flags); + + if (busy) /* Already inside the scanner */ + return; + + n = 50; + + while (input_avail(devc) && n-- > 0) + { + unsigned char c = read_data(devc); + + if (devc->mode == MODE_SYNTH) + { + mpu_input_scanner(devc, c); + } + else if (devc->opened & OPEN_READ && devc->inputintr != NULL) + devc->inputintr(devc->devno, c); + } + devc->m_busy = 0; +} + +static irqreturn_t mpuintr(int irq, void *dev_id) +{ + struct mpu_config *devc; + int dev = (int)(unsigned long) dev_id; + int handled = 0; + + devc = &dev_conf[dev]; + + if (input_avail(devc)) + { + handled = 1; + if (devc->base != 0 && (devc->opened & OPEN_READ || devc->mode == MODE_SYNTH)) + mpu401_input_loop(devc); + else + { + /* Dummy read (just to acknowledge the interrupt) */ + read_data(devc); + } + } + return IRQ_RETVAL(handled); +} + +static int mpu401_open(int dev, int mode, + void (*input) (int dev, unsigned char data), + void (*output) (int dev) +) +{ + int err; + struct mpu_config *devc; + struct coproc_operations *coprocessor; + + if (dev < 0 || dev >= num_midis || midi_devs[dev] == NULL) + return -ENXIO; + + devc = &dev_conf[dev]; + + if (devc->opened) + return -EBUSY; + /* + * Verify that the device is really running. + * Some devices (such as Ensoniq SoundScape don't + * work before the on board processor (OBP) is initialized + * by downloading its microcode. + */ + + if (!devc->initialized) + { + if (mpu401_status(devc) == 0xff) /* Bus float */ + { + printk(KERN_ERR "mpu401: Device not initialized properly\n"); + return -EIO; + } + reset_mpu401(devc); + } + + if ( (coprocessor = midi_devs[dev]->coproc) != NULL ) + { + if (!try_module_get(coprocessor->owner)) { + mpu401_close(dev); + return -ENODEV; + } + + if ((err = coprocessor->open(coprocessor->devc, COPR_MIDI)) < 0) + { + printk(KERN_WARNING "MPU-401: Can't access coprocessor device\n"); + mpu401_close(dev); + return err; + } + } + + set_uart_mode(dev, devc, 1); + devc->mode = MODE_MIDI; + devc->synthno = 0; + + mpu401_input_loop(devc); + + devc->inputintr = input; + devc->opened = mode; + + return 0; +} + +static void mpu401_close(int dev) +{ + struct mpu_config *devc; + struct coproc_operations *coprocessor; + + devc = &dev_conf[dev]; + if (devc->uart_mode) + reset_mpu401(devc); /* + * This disables the UART mode + */ + devc->mode = 0; + devc->inputintr = NULL; + + coprocessor = midi_devs[dev]->coproc; + if (coprocessor) { + coprocessor->close(coprocessor->devc, COPR_MIDI); + module_put(coprocessor->owner); + } + devc->opened = 0; +} + +static int mpu401_out(int dev, unsigned char midi_byte) +{ + int timeout; + unsigned long flags; + + struct mpu_config *devc; + + devc = &dev_conf[dev]; + + /* + * Sometimes it takes about 30000 loops before the output becomes ready + * (After reset). Normally it takes just about 10 loops. + */ + + for (timeout = 30000; timeout > 0 && !output_ready(devc); timeout--); + + spin_lock_irqsave(&devc->lock,flags); + if (!output_ready(devc)) + { + printk(KERN_WARNING "mpu401: Send data timeout\n"); + spin_unlock_irqrestore(&devc->lock,flags); + return 0; + } + write_data(devc, midi_byte); + spin_unlock_irqrestore(&devc->lock,flags); + return 1; +} + +static int mpu401_command(int dev, mpu_command_rec * cmd) +{ + int i, timeout, ok; + int ret = 0; + unsigned long flags; + struct mpu_config *devc; + + devc = &dev_conf[dev]; + + if (devc->uart_mode) /* + * Not possible in UART mode + */ + { + printk(KERN_WARNING "mpu401: commands not possible in the UART mode\n"); + return -EINVAL; + } + /* + * Test for input since pending input seems to block the output. + */ + if (input_avail(devc)) + mpu401_input_loop(devc); + + /* + * Sometimes it takes about 50000 loops before the output becomes ready + * (After reset). Normally it takes just about 10 loops. + */ + + timeout = 50000; +retry: + if (timeout-- <= 0) + { + printk(KERN_WARNING "mpu401: Command (0x%x) timeout\n", (int) cmd->cmd); + return -EIO; + } + spin_lock_irqsave(&devc->lock,flags); + + if (!output_ready(devc)) + { + spin_unlock_irqrestore(&devc->lock,flags); + goto retry; + } + write_command(devc, cmd->cmd); + + ok = 0; + for (timeout = 50000; timeout > 0 && !ok; timeout--) + { + if (input_avail(devc)) + { + if (devc->opened && devc->mode == MODE_SYNTH) + { + if (mpu_input_scanner(devc, read_data(devc)) == MPU_ACK) + ok = 1; + } + else + { + /* Device is not currently open. Use simpler method */ + if (read_data(devc) == MPU_ACK) + ok = 1; + } + } + } + if (!ok) + { + spin_unlock_irqrestore(&devc->lock,flags); + return -EIO; + } + if (cmd->nr_args) + { + for (i = 0; i < cmd->nr_args; i++) + { + for (timeout = 3000; timeout > 0 && !output_ready(devc); timeout--); + + if (!mpu401_out(dev, cmd->data[i])) + { + spin_unlock_irqrestore(&devc->lock,flags); + printk(KERN_WARNING "mpu401: Command (0x%x), parm send failed.\n", (int) cmd->cmd); + return -EIO; + } + } + } + ret = 0; + cmd->data[0] = 0; + + if (cmd->nr_returns) + { + for (i = 0; i < cmd->nr_returns; i++) + { + ok = 0; + for (timeout = 5000; timeout > 0 && !ok; timeout--) + if (input_avail(devc)) + { + cmd->data[i] = read_data(devc); + ok = 1; + } + if (!ok) + { + spin_unlock_irqrestore(&devc->lock,flags); + return -EIO; + } + } + } + spin_unlock_irqrestore(&devc->lock,flags); + return ret; +} + +static int mpu_cmd(int dev, int cmd, int data) +{ + int ret; + + static mpu_command_rec rec; + + rec.cmd = cmd & 0xff; + rec.nr_args = ((cmd & 0xf0) == 0xE0); + rec.nr_returns = ((cmd & 0xf0) == 0xA0); + rec.data[0] = data & 0xff; + + if ((ret = mpu401_command(dev, &rec)) < 0) + return ret; + return (unsigned char) rec.data[0]; +} + +static int mpu401_prefix_cmd(int dev, unsigned char status) +{ + struct mpu_config *devc = &dev_conf[dev]; + + if (devc->uart_mode) + return 1; + + if (status < 0xf0) + { + if (mpu_cmd(dev, 0xD0, 0) < 0) + return 0; + return 1; + } + switch (status) + { + case 0xF0: + if (mpu_cmd(dev, 0xDF, 0) < 0) + return 0; + return 1; + + default: + return 0; + } +} + +static int mpu401_start_read(int dev) +{ + return 0; +} + +static int mpu401_end_read(int dev) +{ + return 0; +} + +static int mpu401_ioctl(int dev, unsigned cmd, void __user *arg) +{ + struct mpu_config *devc; + mpu_command_rec rec; + int val, ret; + + devc = &dev_conf[dev]; + switch (cmd) + { + case SNDCTL_MIDI_MPUMODE: + if (!(devc->capabilities & MPU_CAP_INTLG)) { /* No intelligent mode */ + printk(KERN_WARNING "mpu401: Intelligent mode not supported by the HW\n"); + return -EINVAL; + } + if (get_user(val, (int __user *)arg)) + return -EFAULT; + set_uart_mode(dev, devc, !val); + return 0; + + case SNDCTL_MIDI_MPUCMD: + if (copy_from_user(&rec, arg, sizeof(rec))) + return -EFAULT; + if ((ret = mpu401_command(dev, &rec)) < 0) + return ret; + if (copy_to_user(arg, &rec, sizeof(rec))) + return -EFAULT; + return 0; + + default: + return -EINVAL; + } +} + +static void mpu401_kick(int dev) +{ +} + +static int mpu401_buffer_status(int dev) +{ + return 0; /* + * No data in buffers + */ +} + +static int mpu_synth_ioctl(int dev, unsigned int cmd, void __user *arg) +{ + int midi_dev; + struct mpu_config *devc; + + midi_dev = synth_devs[dev]->midi_dev; + + if (midi_dev < 0 || midi_dev >= num_midis || midi_devs[midi_dev] == NULL) + return -ENXIO; + + devc = &dev_conf[midi_dev]; + + switch (cmd) + { + + case SNDCTL_SYNTH_INFO: + if (copy_to_user(arg, &mpu_synth_info[midi_dev], + sizeof(struct synth_info))) + return -EFAULT; + return 0; + + case SNDCTL_SYNTH_MEMAVL: + return 0x7fffffff; + + default: + return -EINVAL; + } +} + +static int mpu_synth_open(int dev, int mode) +{ + int midi_dev, err; + struct mpu_config *devc; + struct coproc_operations *coprocessor; + + midi_dev = synth_devs[dev]->midi_dev; + + if (midi_dev < 0 || midi_dev > num_midis || midi_devs[midi_dev] == NULL) + return -ENXIO; + + devc = &dev_conf[midi_dev]; + + /* + * Verify that the device is really running. + * Some devices (such as Ensoniq SoundScape don't + * work before the on board processor (OBP) is initialized + * by downloading its microcode. + */ + + if (!devc->initialized) + { + if (mpu401_status(devc) == 0xff) /* Bus float */ + { + printk(KERN_ERR "mpu401: Device not initialized properly\n"); + return -EIO; + } + reset_mpu401(devc); + } + if (devc->opened) + return -EBUSY; + devc->mode = MODE_SYNTH; + devc->synthno = dev; + + devc->inputintr = NULL; + + coprocessor = midi_devs[midi_dev]->coproc; + if (coprocessor) { + if (!try_module_get(coprocessor->owner)) + return -ENODEV; + + if ((err = coprocessor->open(coprocessor->devc, COPR_MIDI)) < 0) + { + printk(KERN_WARNING "mpu401: Can't access coprocessor device\n"); + return err; + } + } + devc->opened = mode; + reset_mpu401(devc); + + if (mode & OPEN_READ) + { + mpu_cmd(midi_dev, 0x8B, 0); /* Enable data in stop mode */ + mpu_cmd(midi_dev, 0x34, 0); /* Return timing bytes in stop mode */ + mpu_cmd(midi_dev, 0x87, 0); /* Enable pitch & controller */ + } + return 0; +} + +static void mpu_synth_close(int dev) +{ + int midi_dev; + struct mpu_config *devc; + struct coproc_operations *coprocessor; + + midi_dev = synth_devs[dev]->midi_dev; + + devc = &dev_conf[midi_dev]; + mpu_cmd(midi_dev, 0x15, 0); /* Stop recording, playback and MIDI */ + mpu_cmd(midi_dev, 0x8a, 0); /* Disable data in stopped mode */ + + devc->inputintr = NULL; + + coprocessor = midi_devs[midi_dev]->coproc; + if (coprocessor) { + coprocessor->close(coprocessor->devc, COPR_MIDI); + module_put(coprocessor->owner); + } + devc->opened = 0; + devc->mode = 0; +} + +#define MIDI_SYNTH_NAME "MPU-401 UART Midi" +#define MIDI_SYNTH_CAPS SYNTH_CAP_INPUT +#include "midi_synth.h" + +static struct synth_operations mpu401_synth_proto = +{ + .owner = THIS_MODULE, + .id = "MPU401", + .info = NULL, + .midi_dev = 0, + .synth_type = SYNTH_TYPE_MIDI, + .synth_subtype = 0, + .open = mpu_synth_open, + .close = mpu_synth_close, + .ioctl = mpu_synth_ioctl, + .kill_note = midi_synth_kill_note, + .start_note = midi_synth_start_note, + .set_instr = midi_synth_set_instr, + .reset = midi_synth_reset, + .hw_control = midi_synth_hw_control, + .load_patch = midi_synth_load_patch, + .aftertouch = midi_synth_aftertouch, + .controller = midi_synth_controller, + .panning = midi_synth_panning, + .bender = midi_synth_bender, + .setup_voice = midi_synth_setup_voice, + .send_sysex = midi_synth_send_sysex +}; + +static struct synth_operations *mpu401_synth_operations[MAX_MIDI_DEV]; + +static struct midi_operations mpu401_midi_proto = +{ + .owner = THIS_MODULE, + .info = {"MPU-401 Midi", 0, MIDI_CAP_MPU401, SNDCARD_MPU401}, + .in_info = {0}, + .open = mpu401_open, + .close = mpu401_close, + .ioctl = mpu401_ioctl, + .outputc = mpu401_out, + .start_read = mpu401_start_read, + .end_read = mpu401_end_read, + .kick = mpu401_kick, + .buffer_status = mpu401_buffer_status, + .prefix_cmd = mpu401_prefix_cmd +}; + +static struct midi_operations mpu401_midi_operations[MAX_MIDI_DEV]; + +static void mpu401_chk_version(int n, struct mpu_config *devc) +{ + int tmp; + + devc->version = devc->revision = 0; + + tmp = mpu_cmd(n, 0xAC, 0); + if (tmp < 0) + return; + if ((tmp & 0xf0) > 0x20) /* Why it's larger than 2.x ??? */ + return; + devc->version = tmp; + + if ((tmp = mpu_cmd(n, 0xAD, 0)) < 0) { + devc->version = 0; + return; + } + devc->revision = tmp; +} + +int attach_mpu401(struct address_info *hw_config, struct module *owner) +{ + unsigned long flags; + char revision_char; + + int m, ret; + struct mpu_config *devc; + + hw_config->slots[1] = -1; + m = sound_alloc_mididev(); + if (m == -1) + { + printk(KERN_WARNING "MPU-401: Too many midi devices detected\n"); + ret = -ENOMEM; + goto out_err; + } + devc = &dev_conf[m]; + devc->base = hw_config->io_base; + devc->osp = hw_config->osp; + devc->irq = hw_config->irq; + devc->opened = 0; + devc->uart_mode = 0; + devc->initialized = 0; + devc->version = 0; + devc->revision = 0; + devc->capabilities = 0; + devc->timer_flag = 0; + devc->m_busy = 0; + devc->m_state = ST_INIT; + devc->shared_irq = hw_config->always_detect; + devc->irq = hw_config->irq; + spin_lock_init(&devc->lock); + + if (devc->irq < 0) + { + devc->irq *= -1; + devc->shared_irq = 1; + } + + if (!hw_config->always_detect) + { + /* Verify the hardware again */ + if (!reset_mpu401(devc)) + { + printk(KERN_WARNING "mpu401: Device didn't respond\n"); + ret = -ENODEV; + goto out_mididev; + } + if (!devc->shared_irq) + { + if (request_irq(devc->irq, mpuintr, 0, "mpu401", + hw_config) < 0) + { + printk(KERN_WARNING "mpu401: Failed to allocate IRQ%d\n", devc->irq); + ret = -ENOMEM; + goto out_mididev; + } + } + spin_lock_irqsave(&devc->lock,flags); + mpu401_chk_version(m, devc); + if (devc->version == 0) + mpu401_chk_version(m, devc); + spin_unlock_irqrestore(&devc->lock, flags); + } + + if (devc->version != 0) + if (mpu_cmd(m, 0xC5, 0) >= 0) /* Set timebase OK */ + if (mpu_cmd(m, 0xE0, 120) >= 0) /* Set tempo OK */ + devc->capabilities |= MPU_CAP_INTLG; /* Supports intelligent mode */ + + + mpu401_synth_operations[m] = kmalloc(sizeof(struct synth_operations), GFP_KERNEL); + + if (mpu401_synth_operations[m] == NULL) + { + printk(KERN_ERR "mpu401: Can't allocate memory\n"); + ret = -ENOMEM; + goto out_irq; + } + if (!(devc->capabilities & MPU_CAP_INTLG)) /* No intelligent mode */ + { + memcpy((char *) mpu401_synth_operations[m], + (char *) &std_midi_synth, + sizeof(struct synth_operations)); + } + else + { + memcpy((char *) mpu401_synth_operations[m], + (char *) &mpu401_synth_proto, + sizeof(struct synth_operations)); + } + if (owner) + mpu401_synth_operations[m]->owner = owner; + + memcpy((char *) &mpu401_midi_operations[m], + (char *) &mpu401_midi_proto, + sizeof(struct midi_operations)); + + mpu401_midi_operations[m].converter = mpu401_synth_operations[m]; + + memcpy((char *) &mpu_synth_info[m], + (char *) &mpu_synth_info_proto, + sizeof(struct synth_info)); + + n_mpu_devs++; + + if (devc->version == 0x20 && devc->revision >= 0x07) /* MusicQuest interface */ + { + int ports = (devc->revision & 0x08) ? 32 : 16; + + devc->capabilities |= MPU_CAP_SYNC | MPU_CAP_SMPTE | + MPU_CAP_CLS | MPU_CAP_2PORT; + + revision_char = (devc->revision == 0x7f) ? 'M' : ' '; + sprintf(mpu_synth_info[m].name, "MQX-%d%c MIDI Interface #%d", + ports, + revision_char, + n_mpu_devs); + } + else + { + revision_char = devc->revision ? devc->revision + '@' : ' '; + if ((int) devc->revision > ('Z' - '@')) + revision_char = '+'; + + devc->capabilities |= MPU_CAP_SYNC | MPU_CAP_FSK; + + if (hw_config->name) + sprintf(mpu_synth_info[m].name, "%s (MPU401)", hw_config->name); + else + sprintf(mpu_synth_info[m].name, + "MPU-401 %d.%d%c MIDI #%d", + (int) (devc->version & 0xf0) >> 4, + devc->version & 0x0f, + revision_char, + n_mpu_devs); + } + + strcpy(mpu401_midi_operations[m].info.name, + mpu_synth_info[m].name); + + conf_printf(mpu_synth_info[m].name, hw_config); + + mpu401_synth_operations[m]->midi_dev = devc->devno = m; + mpu401_synth_operations[devc->devno]->info = &mpu_synth_info[devc->devno]; + + if (devc->capabilities & MPU_CAP_INTLG) /* Intelligent mode */ + hw_config->slots[2] = mpu_timer_init(m); + + midi_devs[m] = &mpu401_midi_operations[devc->devno]; + + if (owner) + midi_devs[m]->owner = owner; + + hw_config->slots[1] = m; + sequencer_init(); + + return 0; + +out_irq: + free_irq(devc->irq, hw_config); +out_mididev: + sound_unload_mididev(m); +out_err: + release_region(hw_config->io_base, 2); + return ret; +} + +static int reset_mpu401(struct mpu_config *devc) +{ + unsigned long flags; + int ok, timeout, n; + int timeout_limit; + + /* + * Send the RESET command. Try again if no success at the first time. + * (If the device is in the UART mode, it will not ack the reset cmd). + */ + + ok = 0; + + timeout_limit = devc->initialized ? 30000 : 100000; + devc->initialized = 1; + + for (n = 0; n < 2 && !ok; n++) + { + for (timeout = timeout_limit; timeout > 0 && !ok; timeout--) + ok = output_ready(devc); + + write_command(devc, MPU_RESET); /* + * Send MPU-401 RESET Command + */ + + /* + * Wait at least 25 msec. This method is not accurate so let's make the + * loop bit longer. Cannot sleep since this is called during boot. + */ + + for (timeout = timeout_limit * 2; timeout > 0 && !ok; timeout--) + { + spin_lock_irqsave(&devc->lock,flags); + if (input_avail(devc)) + if (read_data(devc) == MPU_ACK) + ok = 1; + spin_unlock_irqrestore(&devc->lock,flags); + } + + } + + devc->m_state = ST_INIT; + devc->m_ptr = 0; + devc->m_left = 0; + devc->last_status = 0; + devc->uart_mode = 0; + + return ok; +} + +static void set_uart_mode(int dev, struct mpu_config *devc, int arg) +{ + if (!arg && (devc->capabilities & MPU_CAP_INTLG)) + return; + if ((devc->uart_mode == 0) == (arg == 0)) + return; /* Already set */ + reset_mpu401(devc); /* This exits the uart mode */ + + if (arg) + { + if (mpu_cmd(dev, UART_MODE_ON, 0) < 0) + { + printk(KERN_ERR "mpu401: Can't enter UART mode\n"); + devc->uart_mode = 0; + return; + } + } + devc->uart_mode = arg; + +} + +int probe_mpu401(struct address_info *hw_config, struct resource *ports) +{ + int ok = 0; + struct mpu_config tmp_devc; + + tmp_devc.base = hw_config->io_base; + tmp_devc.irq = hw_config->irq; + tmp_devc.initialized = 0; + tmp_devc.opened = 0; + tmp_devc.osp = hw_config->osp; + + if (hw_config->always_detect) + return 1; + + if (inb(hw_config->io_base + 1) == 0xff) + { + DDB(printk("MPU401: Port %x looks dead.\n", hw_config->io_base)); + return 0; /* Just bus float? */ + } + ok = reset_mpu401(&tmp_devc); + + if (!ok) + { + DDB(printk("MPU401: Reset failed on port %x\n", hw_config->io_base)); + } + return ok; +} + +void unload_mpu401(struct address_info *hw_config) +{ + void *p; + int n=hw_config->slots[1]; + + if (n != -1) { + release_region(hw_config->io_base, 2); + if (hw_config->always_detect == 0 && hw_config->irq > 0) + free_irq(hw_config->irq, hw_config); + p=mpu401_synth_operations[n]; + sound_unload_mididev(n); + sound_unload_timerdev(hw_config->slots[2]); + kfree(p); + } +} + +/***************************************************** + * Timer stuff + ****************************************************/ + +static volatile int timer_initialized = 0, timer_open = 0, tmr_running = 0; +static volatile int curr_tempo, curr_timebase, hw_timebase; +static int max_timebase = 8; /* 8*24=192 ppqn */ +static volatile unsigned long next_event_time; +static volatile unsigned long curr_ticks, curr_clocks; +static unsigned long prev_event_time; +static int metronome_mode; + +static unsigned long clocks2ticks(unsigned long clocks) +{ + /* + * The MPU-401 supports just a limited set of possible timebase values. + * Since the applications require more choices, the driver has to + * program the HW to do its best and to convert between the HW and + * actual timebases. + */ + return ((clocks * curr_timebase) + (hw_timebase / 2)) / hw_timebase; +} + +static void set_timebase(int midi_dev, int val) +{ + int hw_val; + + if (val < 48) + val = 48; + if (val > 1000) + val = 1000; + + hw_val = val; + hw_val = (hw_val + 12) / 24; + if (hw_val > max_timebase) + hw_val = max_timebase; + + if (mpu_cmd(midi_dev, 0xC0 | (hw_val & 0x0f), 0) < 0) + { + printk(KERN_WARNING "mpu401: Can't set HW timebase to %d\n", hw_val * 24); + return; + } + hw_timebase = hw_val * 24; + curr_timebase = val; + +} + +static void tmr_reset(struct mpu_config *devc) +{ + unsigned long flags; + + spin_lock_irqsave(&devc->lock,flags); + next_event_time = (unsigned long) -1; + prev_event_time = 0; + curr_ticks = curr_clocks = 0; + spin_unlock_irqrestore(&devc->lock,flags); +} + +static void set_timer_mode(int midi_dev) +{ + if (timer_mode & TMR_MODE_CLS) + mpu_cmd(midi_dev, 0x3c, 0); /* Use CLS sync */ + else if (timer_mode & TMR_MODE_SMPTE) + mpu_cmd(midi_dev, 0x3d, 0); /* Use SMPTE sync */ + + if (timer_mode & TMR_INTERNAL) + { + mpu_cmd(midi_dev, 0x80, 0); /* Use MIDI sync */ + } + else + { + if (timer_mode & (TMR_MODE_MIDI | TMR_MODE_CLS)) + { + mpu_cmd(midi_dev, 0x82, 0); /* Use MIDI sync */ + mpu_cmd(midi_dev, 0x91, 0); /* Enable ext MIDI ctrl */ + } + else if (timer_mode & TMR_MODE_FSK) + mpu_cmd(midi_dev, 0x81, 0); /* Use FSK sync */ + } +} + +static void stop_metronome(int midi_dev) +{ + mpu_cmd(midi_dev, 0x84, 0); /* Disable metronome */ +} + +static void setup_metronome(int midi_dev) +{ + int numerator, denominator; + int clks_per_click, num_32nds_per_beat; + int beats_per_measure; + + numerator = ((unsigned) metronome_mode >> 24) & 0xff; + denominator = ((unsigned) metronome_mode >> 16) & 0xff; + clks_per_click = ((unsigned) metronome_mode >> 8) & 0xff; + num_32nds_per_beat = (unsigned) metronome_mode & 0xff; + beats_per_measure = (numerator * 4) >> denominator; + + if (!metronome_mode) + mpu_cmd(midi_dev, 0x84, 0); /* Disable metronome */ + else + { + mpu_cmd(midi_dev, 0xE4, clks_per_click); + mpu_cmd(midi_dev, 0xE6, beats_per_measure); + mpu_cmd(midi_dev, 0x83, 0); /* Enable metronome without accents */ + } +} + +static int mpu_start_timer(int midi_dev) +{ + struct mpu_config *devc= &dev_conf[midi_dev]; + + tmr_reset(devc); + set_timer_mode(midi_dev); + + if (tmr_running) + return TIMER_NOT_ARMED; /* Already running */ + + if (timer_mode & TMR_INTERNAL) + { + mpu_cmd(midi_dev, 0x02, 0); /* Send MIDI start */ + tmr_running = 1; + return TIMER_NOT_ARMED; + } + else + { + mpu_cmd(midi_dev, 0x35, 0); /* Enable mode messages to PC */ + mpu_cmd(midi_dev, 0x38, 0); /* Enable sys common messages to PC */ + mpu_cmd(midi_dev, 0x39, 0); /* Enable real time messages to PC */ + mpu_cmd(midi_dev, 0x97, 0); /* Enable system exclusive messages to PC */ + } + return TIMER_ARMED; +} + +static int mpu_timer_open(int dev, int mode) +{ + int midi_dev = sound_timer_devs[dev]->devlink; + struct mpu_config *devc= &dev_conf[midi_dev]; + + if (timer_open) + return -EBUSY; + + tmr_reset(devc); + curr_tempo = 50; + mpu_cmd(midi_dev, 0xE0, 50); + curr_timebase = hw_timebase = 120; + set_timebase(midi_dev, 120); + timer_open = 1; + metronome_mode = 0; + set_timer_mode(midi_dev); + + mpu_cmd(midi_dev, 0xe7, 0x04); /* Send all clocks to host */ + mpu_cmd(midi_dev, 0x95, 0); /* Enable clock to host */ + + return 0; +} + +static void mpu_timer_close(int dev) +{ + int midi_dev = sound_timer_devs[dev]->devlink; + + timer_open = tmr_running = 0; + mpu_cmd(midi_dev, 0x15, 0); /* Stop all */ + mpu_cmd(midi_dev, 0x94, 0); /* Disable clock to host */ + mpu_cmd(midi_dev, 0x8c, 0); /* Disable measure end messages to host */ + stop_metronome(midi_dev); +} + +static int mpu_timer_event(int dev, unsigned char *event) +{ + unsigned char command = event[1]; + unsigned long parm = *(unsigned int *) &event[4]; + int midi_dev = sound_timer_devs[dev]->devlink; + + switch (command) + { + case TMR_WAIT_REL: + parm += prev_event_time; + case TMR_WAIT_ABS: + if (parm > 0) + { + long time; + + if (parm <= curr_ticks) /* It's the time */ + return TIMER_NOT_ARMED; + time = parm; + next_event_time = prev_event_time = time; + + return TIMER_ARMED; + } + break; + + case TMR_START: + if (tmr_running) + break; + return mpu_start_timer(midi_dev); + + case TMR_STOP: + mpu_cmd(midi_dev, 0x01, 0); /* Send MIDI stop */ + stop_metronome(midi_dev); + tmr_running = 0; + break; + + case TMR_CONTINUE: + if (tmr_running) + break; + mpu_cmd(midi_dev, 0x03, 0); /* Send MIDI continue */ + setup_metronome(midi_dev); + tmr_running = 1; + break; + + case TMR_TEMPO: + if (parm) + { + if (parm < 8) + parm = 8; + if (parm > 250) + parm = 250; + if (mpu_cmd(midi_dev, 0xE0, parm) < 0) + printk(KERN_WARNING "mpu401: Can't set tempo to %d\n", (int) parm); + curr_tempo = parm; + } + break; + + case TMR_ECHO: + seq_copy_to_input(event, 8); + break; + + case TMR_TIMESIG: + if (metronome_mode) /* Metronome enabled */ + { + metronome_mode = parm; + setup_metronome(midi_dev); + } + break; + + default:; + } + return TIMER_NOT_ARMED; +} + +static unsigned long mpu_timer_get_time(int dev) +{ + if (!timer_open) + return 0; + + return curr_ticks; +} + +static int mpu_timer_ioctl(int dev, unsigned int command, void __user *arg) +{ + int midi_dev = sound_timer_devs[dev]->devlink; + int __user *p = (int __user *)arg; + + switch (command) + { + case SNDCTL_TMR_SOURCE: + { + int parm; + + if (get_user(parm, p)) + return -EFAULT; + parm &= timer_caps; + + if (parm != 0) + { + timer_mode = parm; + + if (timer_mode & TMR_MODE_CLS) + mpu_cmd(midi_dev, 0x3c, 0); /* Use CLS sync */ + else if (timer_mode & TMR_MODE_SMPTE) + mpu_cmd(midi_dev, 0x3d, 0); /* Use SMPTE sync */ + } + if (put_user(timer_mode, p)) + return -EFAULT; + return timer_mode; + } + break; + + case SNDCTL_TMR_START: + mpu_start_timer(midi_dev); + return 0; + + case SNDCTL_TMR_STOP: + tmr_running = 0; + mpu_cmd(midi_dev, 0x01, 0); /* Send MIDI stop */ + stop_metronome(midi_dev); + return 0; + + case SNDCTL_TMR_CONTINUE: + if (tmr_running) + return 0; + tmr_running = 1; + mpu_cmd(midi_dev, 0x03, 0); /* Send MIDI continue */ + return 0; + + case SNDCTL_TMR_TIMEBASE: + { + int val; + if (get_user(val, p)) + return -EFAULT; + if (val) + set_timebase(midi_dev, val); + if (put_user(curr_timebase, p)) + return -EFAULT; + return curr_timebase; + } + break; + + case SNDCTL_TMR_TEMPO: + { + int val; + int ret; + + if (get_user(val, p)) + return -EFAULT; + + if (val) + { + if (val < 8) + val = 8; + if (val > 250) + val = 250; + if ((ret = mpu_cmd(midi_dev, 0xE0, val)) < 0) + { + printk(KERN_WARNING "mpu401: Can't set tempo to %d\n", (int) val); + return ret; + } + curr_tempo = val; + } + if (put_user(curr_tempo, p)) + return -EFAULT; + return curr_tempo; + } + break; + + case SNDCTL_SEQ_CTRLRATE: + { + int val; + if (get_user(val, p)) + return -EFAULT; + + if (val != 0) /* Can't change */ + return -EINVAL; + val = ((curr_tempo * curr_timebase) + 30)/60; + if (put_user(val, p)) + return -EFAULT; + return val; + } + break; + + case SNDCTL_SEQ_GETTIME: + if (put_user(curr_ticks, p)) + return -EFAULT; + return curr_ticks; + + case SNDCTL_TMR_METRONOME: + if (get_user(metronome_mode, p)) + return -EFAULT; + setup_metronome(midi_dev); + return 0; + + default:; + } + return -EINVAL; +} + +static void mpu_timer_arm(int dev, long time) +{ + if (time < 0) + time = curr_ticks + 1; + else if (time <= curr_ticks) /* It's the time */ + return; + next_event_time = prev_event_time = time; + return; +} + +static struct sound_timer_operations mpu_timer = +{ + .owner = THIS_MODULE, + .info = {"MPU-401 Timer", 0}, + .priority = 10, /* Priority */ + .devlink = 0, /* Local device link */ + .open = mpu_timer_open, + .close = mpu_timer_close, + .event = mpu_timer_event, + .get_time = mpu_timer_get_time, + .ioctl = mpu_timer_ioctl, + .arm_timer = mpu_timer_arm +}; + +static void mpu_timer_interrupt(void) +{ + if (!timer_open) + return; + + if (!tmr_running) + return; + + curr_clocks++; + curr_ticks = clocks2ticks(curr_clocks); + + if (curr_ticks >= next_event_time) + { + next_event_time = (unsigned long) -1; + sequencer_timer(0); + } +} + +static void timer_ext_event(struct mpu_config *devc, int event, int parm) +{ + int midi_dev = devc->devno; + + if (!devc->timer_flag) + return; + + switch (event) + { + case TMR_CLOCK: + printk("<MIDI clk>"); + break; + + case TMR_START: + printk("Ext MIDI start\n"); + if (!tmr_running) + { + if (timer_mode & TMR_EXTERNAL) + { + tmr_running = 1; + setup_metronome(midi_dev); + next_event_time = 0; + STORE(SEQ_START_TIMER()); + } + } + break; + + case TMR_STOP: + printk("Ext MIDI stop\n"); + if (timer_mode & TMR_EXTERNAL) + { + tmr_running = 0; + stop_metronome(midi_dev); + STORE(SEQ_STOP_TIMER()); + } + break; + + case TMR_CONTINUE: + printk("Ext MIDI continue\n"); + if (timer_mode & TMR_EXTERNAL) + { + tmr_running = 1; + setup_metronome(midi_dev); + STORE(SEQ_CONTINUE_TIMER()); + } + break; + + case TMR_SPP: + printk("Songpos: %d\n", parm); + if (timer_mode & TMR_EXTERNAL) + { + STORE(SEQ_SONGPOS(parm)); + } + break; + } +} + +static int mpu_timer_init(int midi_dev) +{ + struct mpu_config *devc; + int n; + + devc = &dev_conf[midi_dev]; + + if (timer_initialized) + return -1; /* There is already a similar timer */ + + timer_initialized = 1; + + mpu_timer.devlink = midi_dev; + dev_conf[midi_dev].timer_flag = 1; + + n = sound_alloc_timerdev(); + if (n == -1) + n = 0; + sound_timer_devs[n] = &mpu_timer; + + if (devc->version < 0x20) /* Original MPU-401 */ + timer_caps = TMR_INTERNAL | TMR_EXTERNAL | TMR_MODE_FSK | TMR_MODE_MIDI; + else + { + /* + * The version number 2.0 is used (at least) by the + * MusicQuest cards and the Roland Super-MPU. + * + * MusicQuest has given a special meaning to the bits of the + * revision number. The Super-MPU returns 0. + */ + + if (devc->revision) + timer_caps |= TMR_EXTERNAL | TMR_MODE_MIDI; + + if (devc->revision & 0x02) + timer_caps |= TMR_MODE_CLS; + + + if (devc->revision & 0x40) + max_timebase = 10; /* Has the 216 and 240 ppqn modes */ + } + + timer_mode = (TMR_INTERNAL | TMR_MODE_MIDI) & timer_caps; + return n; + +} + +EXPORT_SYMBOL(probe_mpu401); +EXPORT_SYMBOL(attach_mpu401); +EXPORT_SYMBOL(unload_mpu401); + +static struct address_info cfg; + +static int io = -1; +static int irq = -1; + +module_param(irq, int, 0); +module_param(io, int, 0); + +static int __init init_mpu401(void) +{ + int ret; + /* Can be loaded either for module use or to provide functions + to others */ + if (io != -1 && irq != -1) { + struct resource *ports; + cfg.irq = irq; + cfg.io_base = io; + ports = request_region(io, 2, "mpu401"); + if (!ports) + return -EBUSY; + if (probe_mpu401(&cfg, ports) == 0) { + release_region(io, 2); + return -ENODEV; + } + if ((ret = attach_mpu401(&cfg, THIS_MODULE))) + return ret; + } + + return 0; +} + +static void __exit cleanup_mpu401(void) +{ + if (io != -1 && irq != -1) { + /* Check for use by, for example, sscape driver */ + unload_mpu401(&cfg); + } +} + +module_init(init_mpu401); +module_exit(cleanup_mpu401); + +#ifndef MODULE +static int __init setup_mpu401(char *str) +{ + /* io, irq */ + int ints[3]; + + str = get_options(str, ARRAY_SIZE(ints), ints); + + io = ints[1]; + irq = ints[2]; + + return 1; +} + +__setup("mpu401=", setup_mpu401); +#endif +MODULE_LICENSE("GPL"); diff --git a/sound/oss/mpu401.h b/sound/oss/mpu401.h new file mode 100644 index 00000000..0ad1e9ee --- /dev/null +++ b/sound/oss/mpu401.h @@ -0,0 +1,11 @@ + +/* From uart401.c */ +int probe_uart401 (struct address_info *hw_config, struct module *owner); +void unload_uart401 (struct address_info *hw_config); + +irqreturn_t uart401intr (int irq, void *dev_id); + +/* From mpu401.c */ +int probe_mpu401(struct address_info *hw_config, struct resource *ports); +int attach_mpu401(struct address_info * hw_config, struct module *owner); +void unload_mpu401(struct address_info *hw_info); diff --git a/sound/oss/msnd.c b/sound/oss/msnd.c new file mode 100644 index 00000000..c0cc951b --- /dev/null +++ b/sound/oss/msnd.c @@ -0,0 +1,413 @@ +/********************************************************************* + * + * msnd.c - Driver Base + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Copyright (C) 1998 Andrew Veliath + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/vmalloc.h> +#include <linux/types.h> +#include <linux/delay.h> +#include <linux/mm.h> +#include <linux/init.h> +#include <linux/interrupt.h> + +#include <asm/io.h> +#include <asm/uaccess.h> +#include <linux/spinlock.h> +#include <asm/irq.h> +#include "msnd.h" + +#define LOGNAME "msnd" + +#define MSND_MAX_DEVS 4 + +static multisound_dev_t *devs[MSND_MAX_DEVS]; +static int num_devs; + +int msnd_register(multisound_dev_t *dev) +{ + int i; + + for (i = 0; i < MSND_MAX_DEVS; ++i) + if (devs[i] == NULL) + break; + + if (i == MSND_MAX_DEVS) + return -ENOMEM; + + devs[i] = dev; + ++num_devs; + return 0; +} + +void msnd_unregister(multisound_dev_t *dev) +{ + int i; + + for (i = 0; i < MSND_MAX_DEVS; ++i) + if (devs[i] == dev) + break; + + if (i == MSND_MAX_DEVS) { + printk(KERN_WARNING LOGNAME ": Unregistering unknown device\n"); + return; + } + + devs[i] = NULL; + --num_devs; +} + +void msnd_init_queue(void __iomem *base, int start, int size) +{ + writew(PCTODSP_BASED(start), base + JQS_wStart); + writew(PCTODSP_OFFSET(size) - 1, base + JQS_wSize); + writew(0, base + JQS_wHead); + writew(0, base + JQS_wTail); +} + +void msnd_fifo_init(msnd_fifo *f) +{ + f->data = NULL; +} + +void msnd_fifo_free(msnd_fifo *f) +{ + vfree(f->data); + f->data = NULL; +} + +int msnd_fifo_alloc(msnd_fifo *f, size_t n) +{ + msnd_fifo_free(f); + f->data = vmalloc(n); + f->n = n; + f->tail = 0; + f->head = 0; + f->len = 0; + + if (!f->data) + return -ENOMEM; + + return 0; +} + +void msnd_fifo_make_empty(msnd_fifo *f) +{ + f->len = f->tail = f->head = 0; +} + +int msnd_fifo_write_io(msnd_fifo *f, char __iomem *buf, size_t len) +{ + int count = 0; + + while ((count < len) && (f->len != f->n)) { + + int nwritten; + + if (f->head <= f->tail) { + nwritten = len - count; + if (nwritten > f->n - f->tail) + nwritten = f->n - f->tail; + } + else { + nwritten = f->head - f->tail; + if (nwritten > len - count) + nwritten = len - count; + } + + memcpy_fromio(f->data + f->tail, buf, nwritten); + + count += nwritten; + buf += nwritten; + f->len += nwritten; + f->tail += nwritten; + f->tail %= f->n; + } + + return count; +} + +int msnd_fifo_write(msnd_fifo *f, const char *buf, size_t len) +{ + int count = 0; + + while ((count < len) && (f->len != f->n)) { + + int nwritten; + + if (f->head <= f->tail) { + nwritten = len - count; + if (nwritten > f->n - f->tail) + nwritten = f->n - f->tail; + } + else { + nwritten = f->head - f->tail; + if (nwritten > len - count) + nwritten = len - count; + } + + memcpy(f->data + f->tail, buf, nwritten); + + count += nwritten; + buf += nwritten; + f->len += nwritten; + f->tail += nwritten; + f->tail %= f->n; + } + + return count; +} + +int msnd_fifo_read_io(msnd_fifo *f, char __iomem *buf, size_t len) +{ + int count = 0; + + while ((count < len) && (f->len > 0)) { + + int nread; + + if (f->tail <= f->head) { + nread = len - count; + if (nread > f->n - f->head) + nread = f->n - f->head; + } + else { + nread = f->tail - f->head; + if (nread > len - count) + nread = len - count; + } + + memcpy_toio(buf, f->data + f->head, nread); + + count += nread; + buf += nread; + f->len -= nread; + f->head += nread; + f->head %= f->n; + } + + return count; +} + +int msnd_fifo_read(msnd_fifo *f, char *buf, size_t len) +{ + int count = 0; + + while ((count < len) && (f->len > 0)) { + + int nread; + + if (f->tail <= f->head) { + nread = len - count; + if (nread > f->n - f->head) + nread = f->n - f->head; + } + else { + nread = f->tail - f->head; + if (nread > len - count) + nread = len - count; + } + + memcpy(buf, f->data + f->head, nread); + + count += nread; + buf += nread; + f->len -= nread; + f->head += nread; + f->head %= f->n; + } + + return count; +} + +static int msnd_wait_TXDE(multisound_dev_t *dev) +{ + register unsigned int io = dev->io; + register int timeout = 1000; + + while(timeout-- > 0) + if (msnd_inb(io + HP_ISR) & HPISR_TXDE) + return 0; + + return -EIO; +} + +static int msnd_wait_HC0(multisound_dev_t *dev) +{ + register unsigned int io = dev->io; + register int timeout = 1000; + + while(timeout-- > 0) + if (!(msnd_inb(io + HP_CVR) & HPCVR_HC)) + return 0; + + return -EIO; +} + +int msnd_send_dsp_cmd(multisound_dev_t *dev, BYTE cmd) +{ + unsigned long flags; + + spin_lock_irqsave(&dev->lock, flags); + if (msnd_wait_HC0(dev) == 0) { + msnd_outb(cmd, dev->io + HP_CVR); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + printk(KERN_DEBUG LOGNAME ": Send DSP command timeout\n"); + + return -EIO; +} + +int msnd_send_word(multisound_dev_t *dev, unsigned char high, + unsigned char mid, unsigned char low) +{ + register unsigned int io = dev->io; + + if (msnd_wait_TXDE(dev) == 0) { + msnd_outb(high, io + HP_TXH); + msnd_outb(mid, io + HP_TXM); + msnd_outb(low, io + HP_TXL); + return 0; + } + + printk(KERN_DEBUG LOGNAME ": Send host word timeout\n"); + + return -EIO; +} + +int msnd_upload_host(multisound_dev_t *dev, char *bin, int len) +{ + int i; + + if (len % 3 != 0) { + printk(KERN_WARNING LOGNAME ": Upload host data not multiple of 3!\n"); + return -EINVAL; + } + + for (i = 0; i < len; i += 3) + if (msnd_send_word(dev, bin[i], bin[i + 1], bin[i + 2]) != 0) + return -EIO; + + msnd_inb(dev->io + HP_RXL); + msnd_inb(dev->io + HP_CVR); + + return 0; +} + +int msnd_enable_irq(multisound_dev_t *dev) +{ + unsigned long flags; + + if (dev->irq_ref++) + return 0; + + printk(KERN_DEBUG LOGNAME ": Enabling IRQ\n"); + + spin_lock_irqsave(&dev->lock, flags); + if (msnd_wait_TXDE(dev) == 0) { + msnd_outb(msnd_inb(dev->io + HP_ICR) | HPICR_TREQ, dev->io + HP_ICR); + if (dev->type == msndClassic) + msnd_outb(dev->irqid, dev->io + HP_IRQM); + msnd_outb(msnd_inb(dev->io + HP_ICR) & ~HPICR_TREQ, dev->io + HP_ICR); + msnd_outb(msnd_inb(dev->io + HP_ICR) | HPICR_RREQ, dev->io + HP_ICR); + enable_irq(dev->irq); + msnd_init_queue(dev->DSPQ, dev->dspq_data_buff, dev->dspq_buff_size); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + printk(KERN_DEBUG LOGNAME ": Enable IRQ failed\n"); + + return -EIO; +} + +int msnd_disable_irq(multisound_dev_t *dev) +{ + unsigned long flags; + + if (--dev->irq_ref > 0) + return 0; + + if (dev->irq_ref < 0) + printk(KERN_DEBUG LOGNAME ": IRQ ref count is %d\n", dev->irq_ref); + + printk(KERN_DEBUG LOGNAME ": Disabling IRQ\n"); + + spin_lock_irqsave(&dev->lock, flags); + if (msnd_wait_TXDE(dev) == 0) { + msnd_outb(msnd_inb(dev->io + HP_ICR) & ~HPICR_RREQ, dev->io + HP_ICR); + if (dev->type == msndClassic) + msnd_outb(HPIRQ_NONE, dev->io + HP_IRQM); + disable_irq(dev->irq); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + printk(KERN_DEBUG LOGNAME ": Disable IRQ failed\n"); + + return -EIO; +} + +#ifndef LINUX20 +EXPORT_SYMBOL(msnd_register); +EXPORT_SYMBOL(msnd_unregister); + +EXPORT_SYMBOL(msnd_init_queue); + +EXPORT_SYMBOL(msnd_fifo_init); +EXPORT_SYMBOL(msnd_fifo_free); +EXPORT_SYMBOL(msnd_fifo_alloc); +EXPORT_SYMBOL(msnd_fifo_make_empty); +EXPORT_SYMBOL(msnd_fifo_write_io); +EXPORT_SYMBOL(msnd_fifo_read_io); +EXPORT_SYMBOL(msnd_fifo_write); +EXPORT_SYMBOL(msnd_fifo_read); + +EXPORT_SYMBOL(msnd_send_dsp_cmd); +EXPORT_SYMBOL(msnd_send_word); +EXPORT_SYMBOL(msnd_upload_host); + +EXPORT_SYMBOL(msnd_enable_irq); +EXPORT_SYMBOL(msnd_disable_irq); +#endif + +#ifdef MODULE +MODULE_AUTHOR ("Andrew Veliath <andrewtv@usa.net>"); +MODULE_DESCRIPTION ("Turtle Beach MultiSound Driver Base"); +MODULE_LICENSE("GPL"); + + +int init_module(void) +{ + return 0; +} + +void cleanup_module(void) +{ +} +#endif diff --git a/sound/oss/msnd.h b/sound/oss/msnd.h new file mode 100644 index 00000000..c8be47ec --- /dev/null +++ b/sound/oss/msnd.h @@ -0,0 +1,278 @@ +/********************************************************************* + * + * msnd.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_H +#define __MSND_H + +#define VERSION "0.8.3.1" + +#define DEFSAMPLERATE DSP_DEFAULT_SPEED +#define DEFSAMPLESIZE AFMT_U8 +#define DEFCHANNELS 1 + +#define DEFFIFOSIZE 128 + +#define SNDCARD_MSND 38 + +#define SRAM_BANK_SIZE 0x8000 +#define SRAM_CNTL_START 0x7F00 + +#define DSP_BASE_ADDR 0x4000 +#define DSP_BANK_BASE 0x4000 + +#define HP_ICR 0x00 +#define HP_CVR 0x01 +#define HP_ISR 0x02 +#define HP_IVR 0x03 +#define HP_NU 0x04 +#define HP_INFO 0x04 +#define HP_TXH 0x05 +#define HP_RXH 0x05 +#define HP_TXM 0x06 +#define HP_RXM 0x06 +#define HP_TXL 0x07 +#define HP_RXL 0x07 + +#define HP_ICR_DEF 0x00 +#define HP_CVR_DEF 0x12 +#define HP_ISR_DEF 0x06 +#define HP_IVR_DEF 0x0f +#define HP_NU_DEF 0x00 + +#define HP_IRQM 0x09 + +#define HPR_BLRC 0x08 +#define HPR_SPR1 0x09 +#define HPR_SPR2 0x0A +#define HPR_TCL0 0x0B +#define HPR_TCL1 0x0C +#define HPR_TCL2 0x0D +#define HPR_TCL3 0x0E +#define HPR_TCL4 0x0F + +#define HPICR_INIT 0x80 +#define HPICR_HM1 0x40 +#define HPICR_HM0 0x20 +#define HPICR_HF1 0x10 +#define HPICR_HF0 0x08 +#define HPICR_TREQ 0x02 +#define HPICR_RREQ 0x01 + +#define HPCVR_HC 0x80 + +#define HPISR_HREQ 0x80 +#define HPISR_DMA 0x40 +#define HPISR_HF3 0x10 +#define HPISR_HF2 0x08 +#define HPISR_TRDY 0x04 +#define HPISR_TXDE 0x02 +#define HPISR_RXDF 0x01 + +#define HPIO_290 0 +#define HPIO_260 1 +#define HPIO_250 2 +#define HPIO_240 3 +#define HPIO_230 4 +#define HPIO_220 5 +#define HPIO_210 6 +#define HPIO_3E0 7 + +#define HPMEM_NONE 0 +#define HPMEM_B000 1 +#define HPMEM_C800 2 +#define HPMEM_D000 3 +#define HPMEM_D400 4 +#define HPMEM_D800 5 +#define HPMEM_E000 6 +#define HPMEM_E800 7 + +#define HPIRQ_NONE 0 +#define HPIRQ_5 1 +#define HPIRQ_7 2 +#define HPIRQ_9 3 +#define HPIRQ_10 4 +#define HPIRQ_11 5 +#define HPIRQ_12 6 +#define HPIRQ_15 7 + +#define HIMT_PLAY_DONE 0x00 +#define HIMT_RECORD_DONE 0x01 +#define HIMT_MIDI_EOS 0x02 +#define HIMT_MIDI_OUT 0x03 + +#define HIMT_MIDI_IN_UCHAR 0x0E +#define HIMT_DSP 0x0F + +#define HDEX_BASE 0x92 +#define HDEX_PLAY_START (0 + HDEX_BASE) +#define HDEX_PLAY_STOP (1 + HDEX_BASE) +#define HDEX_PLAY_PAUSE (2 + HDEX_BASE) +#define HDEX_PLAY_RESUME (3 + HDEX_BASE) +#define HDEX_RECORD_START (4 + HDEX_BASE) +#define HDEX_RECORD_STOP (5 + HDEX_BASE) +#define HDEX_MIDI_IN_START (6 + HDEX_BASE) +#define HDEX_MIDI_IN_STOP (7 + HDEX_BASE) +#define HDEX_MIDI_OUT_START (8 + HDEX_BASE) +#define HDEX_MIDI_OUT_STOP (9 + HDEX_BASE) +#define HDEX_AUX_REQ (10 + HDEX_BASE) + +#define HIWORD(l) ((WORD)((((DWORD)(l)) >> 16) & 0xFFFF)) +#define LOWORD(l) ((WORD)(DWORD)(l)) +#define HIBYTE(w) ((BYTE)(((WORD)(w) >> 8) & 0xFF)) +#define LOBYTE(w) ((BYTE)(w)) +#define MAKELONG(low,hi) ((long)(((WORD)(low))|(((DWORD)((WORD)(hi)))<<16))) +#define MAKEWORD(low,hi) ((WORD)(((BYTE)(low))|(((WORD)((BYTE)(hi)))<<8))) + +#define PCTODSP_OFFSET(w) (USHORT)((w)/2) +#define PCTODSP_BASED(w) (USHORT)(((w)/2) + DSP_BASE_ADDR) +#define DSPTOPC_BASED(w) (((w) - DSP_BASE_ADDR) * 2) + +#ifdef SLOWIO +#define msnd_outb outb_p +#define msnd_inb inb_p +#else +#define msnd_outb outb +#define msnd_inb inb +#endif + +/* JobQueueStruct */ +#define JQS_wStart 0x00 +#define JQS_wSize 0x02 +#define JQS_wHead 0x04 +#define JQS_wTail 0x06 +#define JQS__size 0x08 + +/* DAQueueDataStruct */ +#define DAQDS_wStart 0x00 +#define DAQDS_wSize 0x02 +#define DAQDS_wFormat 0x04 +#define DAQDS_wSampleSize 0x06 +#define DAQDS_wChannels 0x08 +#define DAQDS_wSampleRate 0x0A +#define DAQDS_wIntMsg 0x0C +#define DAQDS_wFlags 0x0E +#define DAQDS__size 0x10 + +typedef u8 BYTE; +typedef u16 USHORT; +typedef u16 WORD; +typedef u32 DWORD; +typedef void __iomem * LPDAQD; + +/* Generic FIFO */ +typedef struct { + size_t n, len; + char *data; + int head, tail; +} msnd_fifo; + +typedef struct multisound_dev { + /* Linux device info */ + char *name; + int dsp_minor, mixer_minor; + int ext_midi_dev, hdr_midi_dev; + + /* Hardware resources */ + int io, numio; + int memid, irqid; + int irq, irq_ref; + unsigned char info; + void __iomem *base; + + /* Motorola 56k DSP SMA */ + void __iomem *SMA; + void __iomem *DAPQ, *DARQ, *MODQ, *MIDQ, *DSPQ; + void __iomem *pwDSPQData, *pwMIDQData, *pwMODQData; + int dspq_data_buff, dspq_buff_size; + + /* State variables */ + enum { msndClassic, msndPinnacle } type; + fmode_t mode; + unsigned long flags; +#define F_RESETTING 0 +#define F_HAVEDIGITAL 1 +#define F_AUDIO_WRITE_INUSE 2 +#define F_WRITING 3 +#define F_WRITEBLOCK 4 +#define F_WRITEFLUSH 5 +#define F_AUDIO_READ_INUSE 6 +#define F_READING 7 +#define F_READBLOCK 8 +#define F_EXT_MIDI_INUSE 9 +#define F_HDR_MIDI_INUSE 10 +#define F_DISABLE_WRITE_NDELAY 11 + wait_queue_head_t writeblock; + wait_queue_head_t readblock; + wait_queue_head_t writeflush; + spinlock_t lock; + int nresets; + unsigned long recsrc; + int left_levels[32]; + int right_levels[32]; + int mixer_mod_count; + int calibrate_signal; + int play_sample_size, play_sample_rate, play_channels; + int play_ndelay; + int rec_sample_size, rec_sample_rate, rec_channels; + int rec_ndelay; + BYTE bCurrentMidiPatch; + + /* Digital audio FIFOs */ + msnd_fifo DAPF, DARF; + int fifosize; + int last_playbank, last_recbank; + + /* MIDI in callback */ + void (*midi_in_interrupt)(struct multisound_dev *); +} multisound_dev_t; + +#ifndef mdelay +# define mdelay(a) udelay((a) * 1000) +#endif + +int msnd_register(multisound_dev_t *dev); +void msnd_unregister(multisound_dev_t *dev); + +void msnd_init_queue(void __iomem *, int start, int size); + +void msnd_fifo_init(msnd_fifo *f); +void msnd_fifo_free(msnd_fifo *f); +int msnd_fifo_alloc(msnd_fifo *f, size_t n); +void msnd_fifo_make_empty(msnd_fifo *f); +int msnd_fifo_write_io(msnd_fifo *f, char __iomem *buf, size_t len); +int msnd_fifo_read_io(msnd_fifo *f, char __iomem *buf, size_t len); +int msnd_fifo_write(msnd_fifo *f, const char *buf, size_t len); +int msnd_fifo_read(msnd_fifo *f, char *buf, size_t len); + +int msnd_send_dsp_cmd(multisound_dev_t *dev, BYTE cmd); +int msnd_send_word(multisound_dev_t *dev, unsigned char high, + unsigned char mid, unsigned char low); +int msnd_upload_host(multisound_dev_t *dev, char *bin, int len); +int msnd_enable_irq(multisound_dev_t *dev); +int msnd_disable_irq(multisound_dev_t *dev); + +#endif /* __MSND_H */ diff --git a/sound/oss/msnd_classic.c b/sound/oss/msnd_classic.c new file mode 100644 index 00000000..3b23a096 --- /dev/null +++ b/sound/oss/msnd_classic.c @@ -0,0 +1,3 @@ +/* The work is in msnd_pinnacle.c, just define MSND_CLASSIC before it. */ +#define MSND_CLASSIC +#include "msnd_pinnacle.c" diff --git a/sound/oss/msnd_classic.h b/sound/oss/msnd_classic.h new file mode 100644 index 00000000..1a17dde2 --- /dev/null +++ b/sound/oss/msnd_classic.h @@ -0,0 +1,185 @@ +/********************************************************************* + * + * msnd_classic.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_CLASSIC_H +#define __MSND_CLASSIC_H + + +#define DSP_NUMIO 0x10 + +#define HP_MEMM 0x08 + +#define HP_BITM 0x0E +#define HP_WAIT 0x0D +#define HP_DSPR 0x0A +#define HP_PROR 0x0B +#define HP_BLKS 0x0C + +#define HPPRORESET_OFF 0 +#define HPPRORESET_ON 1 + +#define HPDSPRESET_OFF 0 +#define HPDSPRESET_ON 1 + +#define HPBLKSEL_0 0 +#define HPBLKSEL_1 1 + +#define HPWAITSTATE_0 0 +#define HPWAITSTATE_1 1 + +#define HPBITMODE_16 0 +#define HPBITMODE_8 1 + +#define HIDSP_INT_PLAY_UNDER 0x00 +#define HIDSP_INT_RECORD_OVER 0x01 +#define HIDSP_INPUT_CLIPPING 0x02 +#define HIDSP_MIDI_IN_OVER 0x10 +#define HIDSP_MIDI_OVERRUN_ERR 0x13 + +#define HDEXAR_CLEAR_PEAKS 1 +#define HDEXAR_IN_SET_POTS 2 +#define HDEXAR_AUX_SET_POTS 3 +#define HDEXAR_CAL_A_TO_D 4 +#define HDEXAR_RD_EXT_DSP_BITS 5 + +#define TIME_PRO_RESET_DONE 0x028A +#define TIME_PRO_SYSEX 0x0040 +#define TIME_PRO_RESET 0x0032 + +#define AGND 0x01 +#define SIGNAL 0x02 + +#define EXT_DSP_BIT_DCAL 0x0001 +#define EXT_DSP_BIT_MIDI_CON 0x0002 + +#define BUFFSIZE 0x8000 +#define HOSTQ_SIZE 0x40 + +#define SRAM_CNTL_START 0x7F00 +#define SMA_STRUCT_START 0x7F40 + +#define DAP_BUFF_SIZE 0x2400 +#define DAR_BUFF_SIZE 0x2000 + +#define DAPQ_STRUCT_SIZE 0x10 +#define DARQ_STRUCT_SIZE 0x10 +#define DAPQ_BUFF_SIZE (3 * 0x10) +#define DARQ_BUFF_SIZE (3 * 0x10) +#define MODQ_BUFF_SIZE 0x400 +#define MIDQ_BUFF_SIZE 0x200 +#define DSPQ_BUFF_SIZE 0x40 + +#define DAPQ_DATA_BUFF 0x6C00 +#define DARQ_DATA_BUFF 0x6C30 +#define MODQ_DATA_BUFF 0x6C60 +#define MIDQ_DATA_BUFF 0x7060 +#define DSPQ_DATA_BUFF 0x7260 + +#define DAPQ_OFFSET SRAM_CNTL_START +#define DARQ_OFFSET (SRAM_CNTL_START + 0x08) +#define MODQ_OFFSET (SRAM_CNTL_START + 0x10) +#define MIDQ_OFFSET (SRAM_CNTL_START + 0x18) +#define DSPQ_OFFSET (SRAM_CNTL_START + 0x20) + +#define MOP_SYNTH 0x10 +#define MOP_EXTOUT 0x32 +#define MOP_EXTTHRU 0x02 +#define MOP_OUTMASK 0x01 + +#define MIP_EXTIN 0x01 +#define MIP_SYNTH 0x00 +#define MIP_INMASK 0x32 + +/* Classic SMA Common Data */ +#define SMA_wCurrPlayBytes 0x0000 +#define SMA_wCurrRecordBytes 0x0002 +#define SMA_wCurrPlayVolLeft 0x0004 +#define SMA_wCurrPlayVolRight 0x0006 +#define SMA_wCurrInVolLeft 0x0008 +#define SMA_wCurrInVolRight 0x000a +#define SMA_wUser_3 0x000c +#define SMA_wUser_4 0x000e +#define SMA_dwUser_5 0x0010 +#define SMA_dwUser_6 0x0014 +#define SMA_wUser_7 0x0018 +#define SMA_wReserved_A 0x001a +#define SMA_wReserved_B 0x001c +#define SMA_wReserved_C 0x001e +#define SMA_wReserved_D 0x0020 +#define SMA_wReserved_E 0x0022 +#define SMA_wReserved_F 0x0024 +#define SMA_wReserved_G 0x0026 +#define SMA_wReserved_H 0x0028 +#define SMA_wCurrDSPStatusFlags 0x002a +#define SMA_wCurrHostStatusFlags 0x002c +#define SMA_wCurrInputTagBits 0x002e +#define SMA_wCurrLeftPeak 0x0030 +#define SMA_wCurrRightPeak 0x0032 +#define SMA_wExtDSPbits 0x0034 +#define SMA_bExtHostbits 0x0036 +#define SMA_bBoardLevel 0x0037 +#define SMA_bInPotPosRight 0x0038 +#define SMA_bInPotPosLeft 0x0039 +#define SMA_bAuxPotPosRight 0x003a +#define SMA_bAuxPotPosLeft 0x003b +#define SMA_wCurrMastVolLeft 0x003c +#define SMA_wCurrMastVolRight 0x003e +#define SMA_bUser_12 0x0040 +#define SMA_bUser_13 0x0041 +#define SMA_wUser_14 0x0042 +#define SMA_wUser_15 0x0044 +#define SMA_wCalFreqAtoD 0x0046 +#define SMA_wUser_16 0x0048 +#define SMA_wUser_17 0x004a +#define SMA__size 0x004c + +#ifdef HAVE_DSPCODEH +# include "msndperm.c" +# include "msndinit.c" +# define PERMCODE msndperm +# define INITCODE msndinit +# define PERMCODESIZE sizeof(msndperm) +# define INITCODESIZE sizeof(msndinit) +#else +# ifndef CONFIG_MSNDCLAS_INIT_FILE +# define CONFIG_MSNDCLAS_INIT_FILE \ + "/etc/sound/msndinit.bin" +# endif +# ifndef CONFIG_MSNDCLAS_PERM_FILE +# define CONFIG_MSNDCLAS_PERM_FILE \ + "/etc/sound/msndperm.bin" +# endif +# define PERMCODEFILE CONFIG_MSNDCLAS_PERM_FILE +# define INITCODEFILE CONFIG_MSNDCLAS_INIT_FILE +# define PERMCODE dspini +# define INITCODE permini +# define PERMCODESIZE sizeof_dspini +# define INITCODESIZE sizeof_permini +#endif +#define LONGNAME "MultiSound (Classic/Monterey/Tahiti)" + +#endif /* __MSND_CLASSIC_H */ diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c new file mode 100644 index 00000000..7b5c77b3 --- /dev/null +++ b/sound/oss/msnd_pinnacle.c @@ -0,0 +1,1931 @@ +/********************************************************************* + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * Linux 2.0/2.2 Version + * + * msnd_pinnacle.c / msnd_classic.c + * + * -- If MSND_CLASSIC is defined: + * + * -> driver for Turtle Beach Classic/Monterey/Tahiti + * + * -- Else + * + * -> driver for Turtle Beach Pinnacle/Fiji + * + * Copyright (C) 1998 Andrew Veliath + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + * 12-3-2000 Modified IO port validation Steve Sycamore + * + ********************************************************************/ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/types.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/mutex.h> +#include <linux/gfp.h> +#include <asm/irq.h> +#include <asm/io.h> +#include "sound_config.h" +#include "sound_firmware.h" +#ifdef MSND_CLASSIC +# ifndef __alpha__ +# define SLOWIO +# endif +#endif +#include "msnd.h" +#ifdef MSND_CLASSIC +# ifdef CONFIG_MSNDCLAS_HAVE_BOOT +# define HAVE_DSPCODEH +# endif +# include "msnd_classic.h" +# define LOGNAME "msnd_classic" +#else +# ifdef CONFIG_MSNDPIN_HAVE_BOOT +# define HAVE_DSPCODEH +# endif +# include "msnd_pinnacle.h" +# define LOGNAME "msnd_pinnacle" +#endif + +#ifndef CONFIG_MSND_WRITE_NDELAY +# define CONFIG_MSND_WRITE_NDELAY 1 +#endif + +#define get_play_delay_jiffies(size) ((size) * HZ * \ + dev.play_sample_size / 8 / \ + dev.play_sample_rate / \ + dev.play_channels) + +#define get_rec_delay_jiffies(size) ((size) * HZ * \ + dev.rec_sample_size / 8 / \ + dev.rec_sample_rate / \ + dev.rec_channels) + +static DEFINE_MUTEX(msnd_pinnacle_mutex); +static multisound_dev_t dev; + +#ifndef HAVE_DSPCODEH +static char *dspini, *permini; +static int sizeof_dspini, sizeof_permini; +#endif + +static int dsp_full_reset(void); +static void dsp_write_flush(void); + +static __inline__ int chk_send_dsp_cmd(multisound_dev_t *dev, register BYTE cmd) +{ + if (msnd_send_dsp_cmd(dev, cmd) == 0) + return 0; + dsp_full_reset(); + return msnd_send_dsp_cmd(dev, cmd); +} + +static void reset_play_queue(void) +{ + int n; + LPDAQD lpDAQ; + + dev.last_playbank = -1; + writew(PCTODSP_OFFSET(0 * DAQDS__size), dev.DAPQ + JQS_wHead); + writew(PCTODSP_OFFSET(0 * DAQDS__size), dev.DAPQ + JQS_wTail); + + for (n = 0, lpDAQ = dev.base + DAPQ_DATA_BUFF; n < 3; ++n, lpDAQ += DAQDS__size) { + writew(PCTODSP_BASED((DWORD)(DAP_BUFF_SIZE * n)), lpDAQ + DAQDS_wStart); + writew(0, lpDAQ + DAQDS_wSize); + writew(1, lpDAQ + DAQDS_wFormat); + writew(dev.play_sample_size, lpDAQ + DAQDS_wSampleSize); + writew(dev.play_channels, lpDAQ + DAQDS_wChannels); + writew(dev.play_sample_rate, lpDAQ + DAQDS_wSampleRate); + writew(HIMT_PLAY_DONE * 0x100 + n, lpDAQ + DAQDS_wIntMsg); + writew(n, lpDAQ + DAQDS_wFlags); + } +} + +static void reset_record_queue(void) +{ + int n; + LPDAQD lpDAQ; + unsigned long flags; + + dev.last_recbank = 2; + writew(PCTODSP_OFFSET(0 * DAQDS__size), dev.DARQ + JQS_wHead); + writew(PCTODSP_OFFSET(dev.last_recbank * DAQDS__size), dev.DARQ + JQS_wTail); + + /* Critical section: bank 1 access */ + spin_lock_irqsave(&dev.lock, flags); + msnd_outb(HPBLKSEL_1, dev.io + HP_BLKS); + memset_io(dev.base, 0, DAR_BUFF_SIZE * 3); + msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS); + spin_unlock_irqrestore(&dev.lock, flags); + + for (n = 0, lpDAQ = dev.base + DARQ_DATA_BUFF; n < 3; ++n, lpDAQ += DAQDS__size) { + writew(PCTODSP_BASED((DWORD)(DAR_BUFF_SIZE * n)) + 0x4000, lpDAQ + DAQDS_wStart); + writew(DAR_BUFF_SIZE, lpDAQ + DAQDS_wSize); + writew(1, lpDAQ + DAQDS_wFormat); + writew(dev.rec_sample_size, lpDAQ + DAQDS_wSampleSize); + writew(dev.rec_channels, lpDAQ + DAQDS_wChannels); + writew(dev.rec_sample_rate, lpDAQ + DAQDS_wSampleRate); + writew(HIMT_RECORD_DONE * 0x100 + n, lpDAQ + DAQDS_wIntMsg); + writew(n, lpDAQ + DAQDS_wFlags); + } +} + +static void reset_queues(void) +{ + if (dev.mode & FMODE_WRITE) { + msnd_fifo_make_empty(&dev.DAPF); + reset_play_queue(); + } + if (dev.mode & FMODE_READ) { + msnd_fifo_make_empty(&dev.DARF); + reset_record_queue(); + } +} + +static int dsp_set_format(struct file *file, int val) +{ + int data, i; + LPDAQD lpDAQ, lpDARQ; + + lpDAQ = dev.base + DAPQ_DATA_BUFF; + lpDARQ = dev.base + DARQ_DATA_BUFF; + + switch (val) { + case AFMT_U8: + case AFMT_S16_LE: + data = val; + break; + default: + data = DEFSAMPLESIZE; + break; + } + + for (i = 0; i < 3; ++i, lpDAQ += DAQDS__size, lpDARQ += DAQDS__size) { + if (file->f_mode & FMODE_WRITE) + writew(data, lpDAQ + DAQDS_wSampleSize); + if (file->f_mode & FMODE_READ) + writew(data, lpDARQ + DAQDS_wSampleSize); + } + if (file->f_mode & FMODE_WRITE) + dev.play_sample_size = data; + if (file->f_mode & FMODE_READ) + dev.rec_sample_size = data; + + return data; +} + +static int dsp_get_frag_size(void) +{ + int size; + size = dev.fifosize / 4; + if (size > 32 * 1024) + size = 32 * 1024; + return size; +} + +static int dsp_ioctl(struct file *file, unsigned int cmd, unsigned long arg) +{ + int val, i, data, tmp; + LPDAQD lpDAQ, lpDARQ; + audio_buf_info abinfo; + unsigned long flags; + int __user *p = (int __user *)arg; + + lpDAQ = dev.base + DAPQ_DATA_BUFF; + lpDARQ = dev.base + DARQ_DATA_BUFF; + + switch (cmd) { + case SNDCTL_DSP_SUBDIVIDE: + case SNDCTL_DSP_SETFRAGMENT: + case SNDCTL_DSP_SETDUPLEX: + case SNDCTL_DSP_POST: + return 0; + + case SNDCTL_DSP_GETIPTR: + case SNDCTL_DSP_GETOPTR: + case SNDCTL_DSP_MAPINBUF: + case SNDCTL_DSP_MAPOUTBUF: + return -EINVAL; + + case SNDCTL_DSP_GETOSPACE: + if (!(file->f_mode & FMODE_WRITE)) + return -EINVAL; + spin_lock_irqsave(&dev.lock, flags); + abinfo.fragsize = dsp_get_frag_size(); + abinfo.bytes = dev.DAPF.n - dev.DAPF.len; + abinfo.fragstotal = dev.DAPF.n / abinfo.fragsize; + abinfo.fragments = abinfo.bytes / abinfo.fragsize; + spin_unlock_irqrestore(&dev.lock, flags); + return copy_to_user((void __user *)arg, &abinfo, sizeof(abinfo)) ? -EFAULT : 0; + + case SNDCTL_DSP_GETISPACE: + if (!(file->f_mode & FMODE_READ)) + return -EINVAL; + spin_lock_irqsave(&dev.lock, flags); + abinfo.fragsize = dsp_get_frag_size(); + abinfo.bytes = dev.DARF.n - dev.DARF.len; + abinfo.fragstotal = dev.DARF.n / abinfo.fragsize; + abinfo.fragments = abinfo.bytes / abinfo.fragsize; + spin_unlock_irqrestore(&dev.lock, flags); + return copy_to_user((void __user *)arg, &abinfo, sizeof(abinfo)) ? -EFAULT : 0; + + case SNDCTL_DSP_RESET: + dev.nresets = 0; + reset_queues(); + return 0; + + case SNDCTL_DSP_SYNC: + dsp_write_flush(); + return 0; + + case SNDCTL_DSP_GETBLKSIZE: + tmp = dsp_get_frag_size(); + if (put_user(tmp, p)) + return -EFAULT; + return 0; + + case SNDCTL_DSP_GETFMTS: + val = AFMT_S16_LE | AFMT_U8; + if (put_user(val, p)) + return -EFAULT; + return 0; + + case SNDCTL_DSP_SETFMT: + if (get_user(val, p)) + return -EFAULT; + + if (file->f_mode & FMODE_WRITE) + data = val == AFMT_QUERY + ? dev.play_sample_size + : dsp_set_format(file, val); + else + data = val == AFMT_QUERY + ? dev.rec_sample_size + : dsp_set_format(file, val); + + if (put_user(data, p)) + return -EFAULT; + return 0; + + case SNDCTL_DSP_NONBLOCK: + if (!test_bit(F_DISABLE_WRITE_NDELAY, &dev.flags) && + file->f_mode & FMODE_WRITE) + dev.play_ndelay = 1; + if (file->f_mode & FMODE_READ) + dev.rec_ndelay = 1; + return 0; + + case SNDCTL_DSP_GETCAPS: + val = DSP_CAP_DUPLEX | DSP_CAP_BATCH; + if (put_user(val, p)) + return -EFAULT; + return 0; + + case SNDCTL_DSP_SPEED: + if (get_user(val, p)) + return -EFAULT; + + if (val < 8000) + val = 8000; + + if (val > 48000) + val = 48000; + + data = val; + + for (i = 0; i < 3; ++i, lpDAQ += DAQDS__size, lpDARQ += DAQDS__size) { + if (file->f_mode & FMODE_WRITE) + writew(data, lpDAQ + DAQDS_wSampleRate); + if (file->f_mode & FMODE_READ) + writew(data, lpDARQ + DAQDS_wSampleRate); + } + if (file->f_mode & FMODE_WRITE) + dev.play_sample_rate = data; + if (file->f_mode & FMODE_READ) + dev.rec_sample_rate = data; + + if (put_user(data, p)) + return -EFAULT; + return 0; + + case SNDCTL_DSP_CHANNELS: + case SNDCTL_DSP_STEREO: + if (get_user(val, p)) + return -EFAULT; + + if (cmd == SNDCTL_DSP_CHANNELS) { + switch (val) { + case 1: + case 2: + data = val; + break; + default: + val = data = 2; + break; + } + } else { + switch (val) { + case 0: + data = 1; + break; + default: + val = 1; + case 1: + data = 2; + break; + } + } + + for (i = 0; i < 3; ++i, lpDAQ += DAQDS__size, lpDARQ += DAQDS__size) { + if (file->f_mode & FMODE_WRITE) + writew(data, lpDAQ + DAQDS_wChannels); + if (file->f_mode & FMODE_READ) + writew(data, lpDARQ + DAQDS_wChannels); + } + if (file->f_mode & FMODE_WRITE) + dev.play_channels = data; + if (file->f_mode & FMODE_READ) + dev.rec_channels = data; + + if (put_user(val, p)) + return -EFAULT; + return 0; + } + + return -EINVAL; +} + +static int mixer_get(int d) +{ + if (d > 31) + return -EINVAL; + + switch (d) { + case SOUND_MIXER_VOLUME: + case SOUND_MIXER_PCM: + case SOUND_MIXER_LINE: + case SOUND_MIXER_IMIX: + case SOUND_MIXER_LINE1: +#ifndef MSND_CLASSIC + case SOUND_MIXER_MIC: + case SOUND_MIXER_SYNTH: +#endif + return (dev.left_levels[d] >> 8) * 100 / 0xff | + (((dev.right_levels[d] >> 8) * 100 / 0xff) << 8); + default: + return 0; + } +} + +#define update_volm(a,b) \ + writew((dev.left_levels[a] >> 1) * \ + readw(dev.SMA + SMA_wCurrMastVolLeft) / 0xffff, \ + dev.SMA + SMA_##b##Left); \ + writew((dev.right_levels[a] >> 1) * \ + readw(dev.SMA + SMA_wCurrMastVolRight) / 0xffff, \ + dev.SMA + SMA_##b##Right); + +#define update_potm(d,s,ar) \ + writeb((dev.left_levels[d] >> 8) * \ + readw(dev.SMA + SMA_wCurrMastVolLeft) / 0xffff, \ + dev.SMA + SMA_##s##Left); \ + writeb((dev.right_levels[d] >> 8) * \ + readw(dev.SMA + SMA_wCurrMastVolRight) / 0xffff, \ + dev.SMA + SMA_##s##Right); \ + if (msnd_send_word(&dev, 0, 0, ar) == 0) \ + chk_send_dsp_cmd(&dev, HDEX_AUX_REQ); + +#define update_pot(d,s,ar) \ + writeb(dev.left_levels[d] >> 8, \ + dev.SMA + SMA_##s##Left); \ + writeb(dev.right_levels[d] >> 8, \ + dev.SMA + SMA_##s##Right); \ + if (msnd_send_word(&dev, 0, 0, ar) == 0) \ + chk_send_dsp_cmd(&dev, HDEX_AUX_REQ); + +static int mixer_set(int d, int value) +{ + int left = value & 0x000000ff; + int right = (value & 0x0000ff00) >> 8; + int bLeft, bRight; + int wLeft, wRight; + int updatemaster = 0; + + if (d > 31) + return -EINVAL; + + bLeft = left * 0xff / 100; + wLeft = left * 0xffff / 100; + + bRight = right * 0xff / 100; + wRight = right * 0xffff / 100; + + dev.left_levels[d] = wLeft; + dev.right_levels[d] = wRight; + + switch (d) { + /* master volume unscaled controls */ + case SOUND_MIXER_LINE: /* line pot control */ + /* scaled by IMIX in digital mix */ + writeb(bLeft, dev.SMA + SMA_bInPotPosLeft); + writeb(bRight, dev.SMA + SMA_bInPotPosRight); + if (msnd_send_word(&dev, 0, 0, HDEXAR_IN_SET_POTS) == 0) + chk_send_dsp_cmd(&dev, HDEX_AUX_REQ); + break; +#ifndef MSND_CLASSIC + case SOUND_MIXER_MIC: /* mic pot control */ + /* scaled by IMIX in digital mix */ + writeb(bLeft, dev.SMA + SMA_bMicPotPosLeft); + writeb(bRight, dev.SMA + SMA_bMicPotPosRight); + if (msnd_send_word(&dev, 0, 0, HDEXAR_MIC_SET_POTS) == 0) + chk_send_dsp_cmd(&dev, HDEX_AUX_REQ); + break; +#endif + case SOUND_MIXER_VOLUME: /* master volume */ + writew(wLeft, dev.SMA + SMA_wCurrMastVolLeft); + writew(wRight, dev.SMA + SMA_wCurrMastVolRight); + /* fall through */ + + case SOUND_MIXER_LINE1: /* aux pot control */ + /* scaled by master volume */ + /* fall through */ + + /* digital controls */ + case SOUND_MIXER_SYNTH: /* synth vol (dsp mix) */ + case SOUND_MIXER_PCM: /* pcm vol (dsp mix) */ + case SOUND_MIXER_IMIX: /* input monitor (dsp mix) */ + /* scaled by master volume */ + updatemaster = 1; + break; + + default: + return 0; + } + + if (updatemaster) { + /* update master volume scaled controls */ + update_volm(SOUND_MIXER_PCM, wCurrPlayVol); + update_volm(SOUND_MIXER_IMIX, wCurrInVol); +#ifndef MSND_CLASSIC + update_volm(SOUND_MIXER_SYNTH, wCurrMHdrVol); +#endif + update_potm(SOUND_MIXER_LINE1, bAuxPotPos, HDEXAR_AUX_SET_POTS); + } + + return mixer_get(d); +} + +static void mixer_setup(void) +{ + update_pot(SOUND_MIXER_LINE, bInPotPos, HDEXAR_IN_SET_POTS); + update_potm(SOUND_MIXER_LINE1, bAuxPotPos, HDEXAR_AUX_SET_POTS); + update_volm(SOUND_MIXER_PCM, wCurrPlayVol); + update_volm(SOUND_MIXER_IMIX, wCurrInVol); +#ifndef MSND_CLASSIC + update_pot(SOUND_MIXER_MIC, bMicPotPos, HDEXAR_MIC_SET_POTS); + update_volm(SOUND_MIXER_SYNTH, wCurrMHdrVol); +#endif +} + +static unsigned long set_recsrc(unsigned long recsrc) +{ + if (dev.recsrc == recsrc) + return dev.recsrc; +#ifdef HAVE_NORECSRC + else if (recsrc == 0) + dev.recsrc = 0; +#endif + else + dev.recsrc ^= recsrc; + +#ifndef MSND_CLASSIC + if (dev.recsrc & SOUND_MASK_IMIX) { + if (msnd_send_word(&dev, 0, 0, HDEXAR_SET_ANA_IN) == 0) + chk_send_dsp_cmd(&dev, HDEX_AUX_REQ); + } + else if (dev.recsrc & SOUND_MASK_SYNTH) { + if (msnd_send_word(&dev, 0, 0, HDEXAR_SET_SYNTH_IN) == 0) + chk_send_dsp_cmd(&dev, HDEX_AUX_REQ); + } + else if ((dev.recsrc & SOUND_MASK_DIGITAL1) && test_bit(F_HAVEDIGITAL, &dev.flags)) { + if (msnd_send_word(&dev, 0, 0, HDEXAR_SET_DAT_IN) == 0) + chk_send_dsp_cmd(&dev, HDEX_AUX_REQ); + } + else { +#ifdef HAVE_NORECSRC + /* Select no input (?) */ + dev.recsrc = 0; +#else + dev.recsrc = SOUND_MASK_IMIX; + if (msnd_send_word(&dev, 0, 0, HDEXAR_SET_ANA_IN) == 0) + chk_send_dsp_cmd(&dev, HDEX_AUX_REQ); +#endif + } +#endif /* MSND_CLASSIC */ + + return dev.recsrc; +} + +static unsigned long force_recsrc(unsigned long recsrc) +{ + dev.recsrc = 0; + return set_recsrc(recsrc); +} + +#define set_mixer_info() \ + memset(&info, 0, sizeof(info)); \ + strlcpy(info.id, "MSNDMIXER", sizeof(info.id)); \ + strlcpy(info.name, "MultiSound Mixer", sizeof(info.name)); + +static int mixer_ioctl(unsigned int cmd, unsigned long arg) +{ + if (cmd == SOUND_MIXER_INFO) { + mixer_info info; + set_mixer_info(); + info.modify_counter = dev.mixer_mod_count; + if (copy_to_user((void __user *)arg, &info, sizeof(info))) + return -EFAULT; + return 0; + } else if (cmd == SOUND_OLD_MIXER_INFO) { + _old_mixer_info info; + set_mixer_info(); + if (copy_to_user((void __user *)arg, &info, sizeof(info))) + return -EFAULT; + return 0; + } else if (cmd == SOUND_MIXER_PRIVATE1) { + dev.nresets = 0; + dsp_full_reset(); + return 0; + } else if (((cmd >> 8) & 0xff) == 'M') { + int val = 0; + + if (_SIOC_DIR(cmd) & _SIOC_WRITE) { + switch (cmd & 0xff) { + case SOUND_MIXER_RECSRC: + if (get_user(val, (int __user *)arg)) + return -EFAULT; + val = set_recsrc(val); + break; + + default: + if (get_user(val, (int __user *)arg)) + return -EFAULT; + val = mixer_set(cmd & 0xff, val); + break; + } + ++dev.mixer_mod_count; + return put_user(val, (int __user *)arg); + } else { + switch (cmd & 0xff) { + case SOUND_MIXER_RECSRC: + val = dev.recsrc; + break; + + case SOUND_MIXER_DEVMASK: + case SOUND_MIXER_STEREODEVS: + val = SOUND_MASK_PCM | + SOUND_MASK_LINE | + SOUND_MASK_IMIX | + SOUND_MASK_LINE1 | +#ifndef MSND_CLASSIC + SOUND_MASK_MIC | + SOUND_MASK_SYNTH | +#endif + SOUND_MASK_VOLUME; + break; + + case SOUND_MIXER_RECMASK: +#ifdef MSND_CLASSIC + val = 0; +#else + val = SOUND_MASK_IMIX | + SOUND_MASK_SYNTH; + if (test_bit(F_HAVEDIGITAL, &dev.flags)) + val |= SOUND_MASK_DIGITAL1; +#endif + break; + + case SOUND_MIXER_CAPS: + val = SOUND_CAP_EXCL_INPUT; + break; + + default: + if ((val = mixer_get(cmd & 0xff)) < 0) + return -EINVAL; + break; + } + } + + return put_user(val, (int __user *)arg); + } + + return -EINVAL; +} + +static long dev_ioctl(struct file *file, unsigned int cmd, unsigned long arg) +{ + int minor = iminor(file->f_path.dentry->d_inode); + int ret; + + if (cmd == OSS_GETVERSION) { + int sound_version = SOUND_VERSION; + return put_user(sound_version, (int __user *)arg); + } + + ret = -EINVAL; + + mutex_lock(&msnd_pinnacle_mutex); + if (minor == dev.dsp_minor) + ret = dsp_ioctl(file, cmd, arg); + else if (minor == dev.mixer_minor) + ret = mixer_ioctl(cmd, arg); + mutex_unlock(&msnd_pinnacle_mutex); + + return ret; +} + +static void dsp_write_flush(void) +{ + if (!(dev.mode & FMODE_WRITE) || !test_bit(F_WRITING, &dev.flags)) + return; + set_bit(F_WRITEFLUSH, &dev.flags); + interruptible_sleep_on_timeout( + &dev.writeflush, + get_play_delay_jiffies(dev.DAPF.len)); + clear_bit(F_WRITEFLUSH, &dev.flags); + if (!signal_pending(current)) { + current->state = TASK_INTERRUPTIBLE; + schedule_timeout(get_play_delay_jiffies(DAP_BUFF_SIZE)); + } + clear_bit(F_WRITING, &dev.flags); +} + +static void dsp_halt(struct file *file) +{ + if ((file ? file->f_mode : dev.mode) & FMODE_READ) { + clear_bit(F_READING, &dev.flags); + chk_send_dsp_cmd(&dev, HDEX_RECORD_STOP); + msnd_disable_irq(&dev); + if (file) { + printk(KERN_DEBUG LOGNAME ": Stopping read for %p\n", file); + dev.mode &= ~FMODE_READ; + } + clear_bit(F_AUDIO_READ_INUSE, &dev.flags); + } + if ((file ? file->f_mode : dev.mode) & FMODE_WRITE) { + if (test_bit(F_WRITING, &dev.flags)) { + dsp_write_flush(); + chk_send_dsp_cmd(&dev, HDEX_PLAY_STOP); + } + msnd_disable_irq(&dev); + if (file) { + printk(KERN_DEBUG LOGNAME ": Stopping write for %p\n", file); + dev.mode &= ~FMODE_WRITE; + } + clear_bit(F_AUDIO_WRITE_INUSE, &dev.flags); + } +} + +static int dsp_release(struct file *file) +{ + dsp_halt(file); + return 0; +} + +static int dsp_open(struct file *file) +{ + if ((file ? file->f_mode : dev.mode) & FMODE_WRITE) { + set_bit(F_AUDIO_WRITE_INUSE, &dev.flags); + clear_bit(F_WRITING, &dev.flags); + msnd_fifo_make_empty(&dev.DAPF); + reset_play_queue(); + if (file) { + printk(KERN_DEBUG LOGNAME ": Starting write for %p\n", file); + dev.mode |= FMODE_WRITE; + } + msnd_enable_irq(&dev); + } + if ((file ? file->f_mode : dev.mode) & FMODE_READ) { + set_bit(F_AUDIO_READ_INUSE, &dev.flags); + clear_bit(F_READING, &dev.flags); + msnd_fifo_make_empty(&dev.DARF); + reset_record_queue(); + if (file) { + printk(KERN_DEBUG LOGNAME ": Starting read for %p\n", file); + dev.mode |= FMODE_READ; + } + msnd_enable_irq(&dev); + } + return 0; +} + +static void set_default_play_audio_parameters(void) +{ + dev.play_sample_size = DEFSAMPLESIZE; + dev.play_sample_rate = DEFSAMPLERATE; + dev.play_channels = DEFCHANNELS; +} + +static void set_default_rec_audio_parameters(void) +{ + dev.rec_sample_size = DEFSAMPLESIZE; + dev.rec_sample_rate = DEFSAMPLERATE; + dev.rec_channels = DEFCHANNELS; +} + +static void set_default_audio_parameters(void) +{ + set_default_play_audio_parameters(); + set_default_rec_audio_parameters(); +} + +static int dev_open(struct inode *inode, struct file *file) +{ + int minor = iminor(inode); + int err = 0; + + mutex_lock(&msnd_pinnacle_mutex); + if (minor == dev.dsp_minor) { + if ((file->f_mode & FMODE_WRITE && + test_bit(F_AUDIO_WRITE_INUSE, &dev.flags)) || + (file->f_mode & FMODE_READ && + test_bit(F_AUDIO_READ_INUSE, &dev.flags))) { + err = -EBUSY; + goto out; + } + + if ((err = dsp_open(file)) >= 0) { + dev.nresets = 0; + if (file->f_mode & FMODE_WRITE) { + set_default_play_audio_parameters(); + if (!test_bit(F_DISABLE_WRITE_NDELAY, &dev.flags)) + dev.play_ndelay = (file->f_flags & O_NDELAY) ? 1 : 0; + else + dev.play_ndelay = 0; + } + if (file->f_mode & FMODE_READ) { + set_default_rec_audio_parameters(); + dev.rec_ndelay = (file->f_flags & O_NDELAY) ? 1 : 0; + } + } + } + else if (minor == dev.mixer_minor) { + /* nothing */ + } else + err = -EINVAL; +out: + mutex_unlock(&msnd_pinnacle_mutex); + return err; +} + +static int dev_release(struct inode *inode, struct file *file) +{ + int minor = iminor(inode); + int err = 0; + + mutex_lock(&msnd_pinnacle_mutex); + if (minor == dev.dsp_minor) + err = dsp_release(file); + else if (minor == dev.mixer_minor) { + /* nothing */ + } else + err = -EINVAL; + mutex_unlock(&msnd_pinnacle_mutex); + return err; +} + +static __inline__ int pack_DARQ_to_DARF(register int bank) +{ + register int size, timeout = 3; + register WORD wTmp; + LPDAQD DAQD; + + /* Increment the tail and check for queue wrap */ + wTmp = readw(dev.DARQ + JQS_wTail) + PCTODSP_OFFSET(DAQDS__size); + if (wTmp > readw(dev.DARQ + JQS_wSize)) + wTmp = 0; + while (wTmp == readw(dev.DARQ + JQS_wHead) && timeout--) + udelay(1); + writew(wTmp, dev.DARQ + JQS_wTail); + + /* Get our digital audio queue struct */ + DAQD = bank * DAQDS__size + dev.base + DARQ_DATA_BUFF; + + /* Get length of data */ + size = readw(DAQD + DAQDS_wSize); + + /* Read data from the head (unprotected bank 1 access okay + since this is only called inside an interrupt) */ + msnd_outb(HPBLKSEL_1, dev.io + HP_BLKS); + msnd_fifo_write_io( + &dev.DARF, + dev.base + bank * DAR_BUFF_SIZE, + size); + msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS); + + return 1; +} + +static __inline__ int pack_DAPF_to_DAPQ(register int start) +{ + register WORD DAPQ_tail; + register int protect = start, nbanks = 0; + LPDAQD DAQD; + + DAPQ_tail = readw(dev.DAPQ + JQS_wTail); + while (DAPQ_tail != readw(dev.DAPQ + JQS_wHead) || start) { + register int bank_num = DAPQ_tail / PCTODSP_OFFSET(DAQDS__size); + register int n; + unsigned long flags; + + /* Write the data to the new tail */ + if (protect) { + /* Critical section: protect fifo in non-interrupt */ + spin_lock_irqsave(&dev.lock, flags); + n = msnd_fifo_read_io( + &dev.DAPF, + dev.base + bank_num * DAP_BUFF_SIZE, + DAP_BUFF_SIZE); + spin_unlock_irqrestore(&dev.lock, flags); + } else { + n = msnd_fifo_read_io( + &dev.DAPF, + dev.base + bank_num * DAP_BUFF_SIZE, + DAP_BUFF_SIZE); + } + if (!n) + break; + + if (start) + start = 0; + + /* Get our digital audio queue struct */ + DAQD = bank_num * DAQDS__size + dev.base + DAPQ_DATA_BUFF; + + /* Write size of this bank */ + writew(n, DAQD + DAQDS_wSize); + ++nbanks; + + /* Then advance the tail */ + DAPQ_tail = (++bank_num % 3) * PCTODSP_OFFSET(DAQDS__size); + writew(DAPQ_tail, dev.DAPQ + JQS_wTail); + /* Tell the DSP to play the bank */ + msnd_send_dsp_cmd(&dev, HDEX_PLAY_START); + } + return nbanks; +} + +static int dsp_read(char __user *buf, size_t len) +{ + int count = len; + char *page = (char *)__get_free_page(GFP_KERNEL); + + if (!page) + return -ENOMEM; + + while (count > 0) { + int n, k; + unsigned long flags; + + k = PAGE_SIZE; + if (k > count) + k = count; + + /* Critical section: protect fifo in non-interrupt */ + spin_lock_irqsave(&dev.lock, flags); + n = msnd_fifo_read(&dev.DARF, page, k); + spin_unlock_irqrestore(&dev.lock, flags); + if (copy_to_user(buf, page, n)) { + free_page((unsigned long)page); + return -EFAULT; + } + buf += n; + count -= n; + + if (n == k && count) + continue; + + if (!test_bit(F_READING, &dev.flags) && dev.mode & FMODE_READ) { + dev.last_recbank = -1; + if (chk_send_dsp_cmd(&dev, HDEX_RECORD_START) == 0) + set_bit(F_READING, &dev.flags); + } + + if (dev.rec_ndelay) { + free_page((unsigned long)page); + return count == len ? -EAGAIN : len - count; + } + + if (count > 0) { + set_bit(F_READBLOCK, &dev.flags); + if (!interruptible_sleep_on_timeout( + &dev.readblock, + get_rec_delay_jiffies(DAR_BUFF_SIZE))) + clear_bit(F_READING, &dev.flags); + clear_bit(F_READBLOCK, &dev.flags); + if (signal_pending(current)) { + free_page((unsigned long)page); + return -EINTR; + } + } + } + free_page((unsigned long)page); + return len - count; +} + +static int dsp_write(const char __user *buf, size_t len) +{ + int count = len; + char *page = (char *)__get_free_page(GFP_KERNEL); + + if (!page) + return -ENOMEM; + + while (count > 0) { + int n, k; + unsigned long flags; + + k = PAGE_SIZE; + if (k > count) + k = count; + + if (copy_from_user(page, buf, k)) { + free_page((unsigned long)page); + return -EFAULT; + } + + /* Critical section: protect fifo in non-interrupt */ + spin_lock_irqsave(&dev.lock, flags); + n = msnd_fifo_write(&dev.DAPF, page, k); + spin_unlock_irqrestore(&dev.lock, flags); + buf += n; + count -= n; + + if (count && n == k) + continue; + + if (!test_bit(F_WRITING, &dev.flags) && (dev.mode & FMODE_WRITE)) { + dev.last_playbank = -1; + if (pack_DAPF_to_DAPQ(1) > 0) + set_bit(F_WRITING, &dev.flags); + } + + if (dev.play_ndelay) { + free_page((unsigned long)page); + return count == len ? -EAGAIN : len - count; + } + + if (count > 0) { + set_bit(F_WRITEBLOCK, &dev.flags); + interruptible_sleep_on_timeout( + &dev.writeblock, + get_play_delay_jiffies(DAP_BUFF_SIZE)); + clear_bit(F_WRITEBLOCK, &dev.flags); + if (signal_pending(current)) { + free_page((unsigned long)page); + return -EINTR; + } + } + } + + free_page((unsigned long)page); + return len - count; +} + +static ssize_t dev_read(struct file *file, char __user *buf, size_t count, loff_t *off) +{ + int minor = iminor(file->f_path.dentry->d_inode); + if (minor == dev.dsp_minor) + return dsp_read(buf, count); + else + return -EINVAL; +} + +static ssize_t dev_write(struct file *file, const char __user *buf, size_t count, loff_t *off) +{ + int minor = iminor(file->f_path.dentry->d_inode); + if (minor == dev.dsp_minor) + return dsp_write(buf, count); + else + return -EINVAL; +} + +static __inline__ void eval_dsp_msg(register WORD wMessage) +{ + switch (HIBYTE(wMessage)) { + case HIMT_PLAY_DONE: + if (dev.last_playbank == LOBYTE(wMessage) || !test_bit(F_WRITING, &dev.flags)) + break; + dev.last_playbank = LOBYTE(wMessage); + + if (pack_DAPF_to_DAPQ(0) <= 0) { + if (!test_bit(F_WRITEBLOCK, &dev.flags)) { + if (test_and_clear_bit(F_WRITEFLUSH, &dev.flags)) + wake_up_interruptible(&dev.writeflush); + } + clear_bit(F_WRITING, &dev.flags); + } + + if (test_bit(F_WRITEBLOCK, &dev.flags)) + wake_up_interruptible(&dev.writeblock); + break; + + case HIMT_RECORD_DONE: + if (dev.last_recbank == LOBYTE(wMessage)) + break; + dev.last_recbank = LOBYTE(wMessage); + + pack_DARQ_to_DARF(dev.last_recbank); + + if (test_bit(F_READBLOCK, &dev.flags)) + wake_up_interruptible(&dev.readblock); + break; + + case HIMT_DSP: + switch (LOBYTE(wMessage)) { +#ifndef MSND_CLASSIC + case HIDSP_PLAY_UNDER: +#endif + case HIDSP_INT_PLAY_UNDER: +/* printk(KERN_DEBUG LOGNAME ": Play underflow\n"); */ + clear_bit(F_WRITING, &dev.flags); + break; + + case HIDSP_INT_RECORD_OVER: +/* printk(KERN_DEBUG LOGNAME ": Record overflow\n"); */ + clear_bit(F_READING, &dev.flags); + break; + + default: +/* printk(KERN_DEBUG LOGNAME ": DSP message %d 0x%02x\n", + LOBYTE(wMessage), LOBYTE(wMessage)); */ + break; + } + break; + + case HIMT_MIDI_IN_UCHAR: + if (dev.midi_in_interrupt) + (*dev.midi_in_interrupt)(&dev); + break; + + default: +/* printk(KERN_DEBUG LOGNAME ": HIMT message %d 0x%02x\n", HIBYTE(wMessage), HIBYTE(wMessage)); */ + break; + } +} + +static irqreturn_t intr(int irq, void *dev_id) +{ + /* Send ack to DSP */ + msnd_inb(dev.io + HP_RXL); + + /* Evaluate queued DSP messages */ + while (readw(dev.DSPQ + JQS_wTail) != readw(dev.DSPQ + JQS_wHead)) { + register WORD wTmp; + + eval_dsp_msg(readw(dev.pwDSPQData + 2*readw(dev.DSPQ + JQS_wHead))); + + if ((wTmp = readw(dev.DSPQ + JQS_wHead) + 1) > readw(dev.DSPQ + JQS_wSize)) + writew(0, dev.DSPQ + JQS_wHead); + else + writew(wTmp, dev.DSPQ + JQS_wHead); + } + return IRQ_HANDLED; +} + +static const struct file_operations dev_fileops = { + .owner = THIS_MODULE, + .read = dev_read, + .write = dev_write, + .unlocked_ioctl = dev_ioctl, + .open = dev_open, + .release = dev_release, + .llseek = noop_llseek, +}; + +static int reset_dsp(void) +{ + int timeout = 100; + + msnd_outb(HPDSPRESET_ON, dev.io + HP_DSPR); + mdelay(1); +#ifndef MSND_CLASSIC + dev.info = msnd_inb(dev.io + HP_INFO); +#endif + msnd_outb(HPDSPRESET_OFF, dev.io + HP_DSPR); + mdelay(1); + while (timeout-- > 0) { + if (msnd_inb(dev.io + HP_CVR) == HP_CVR_DEF) + return 0; + mdelay(1); + } + printk(KERN_ERR LOGNAME ": Cannot reset DSP\n"); + + return -EIO; +} + +static int __init probe_multisound(void) +{ +#ifndef MSND_CLASSIC + char *xv, *rev = NULL; + char *pin = "Pinnacle", *fiji = "Fiji"; + char *pinfiji = "Pinnacle/Fiji"; +#endif + + if (!request_region(dev.io, dev.numio, "probing")) { + printk(KERN_ERR LOGNAME ": I/O port conflict\n"); + return -ENODEV; + } + + if (reset_dsp() < 0) { + release_region(dev.io, dev.numio); + return -ENODEV; + } + +#ifdef MSND_CLASSIC + dev.name = "Classic/Tahiti/Monterey"; + printk(KERN_INFO LOGNAME ": %s, " +#else + switch (dev.info >> 4) { + case 0xf: xv = "<= 1.15"; break; + case 0x1: xv = "1.18/1.2"; break; + case 0x2: xv = "1.3"; break; + case 0x3: xv = "1.4"; break; + default: xv = "unknown"; break; + } + + switch (dev.info & 0x7) { + case 0x0: rev = "I"; dev.name = pin; break; + case 0x1: rev = "F"; dev.name = pin; break; + case 0x2: rev = "G"; dev.name = pin; break; + case 0x3: rev = "H"; dev.name = pin; break; + case 0x4: rev = "E"; dev.name = fiji; break; + case 0x5: rev = "C"; dev.name = fiji; break; + case 0x6: rev = "D"; dev.name = fiji; break; + case 0x7: + rev = "A-B (Fiji) or A-E (Pinnacle)"; + dev.name = pinfiji; + break; + } + printk(KERN_INFO LOGNAME ": %s revision %s, Xilinx version %s, " +#endif /* MSND_CLASSIC */ + "I/O 0x%x-0x%x, IRQ %d, memory mapped to %p-%p\n", + dev.name, +#ifndef MSND_CLASSIC + rev, xv, +#endif + dev.io, dev.io + dev.numio - 1, + dev.irq, + dev.base, dev.base + 0x7fff); + + release_region(dev.io, dev.numio); + return 0; +} + +static int init_sma(void) +{ + static int initted; + WORD mastVolLeft, mastVolRight; + unsigned long flags; + +#ifdef MSND_CLASSIC + msnd_outb(dev.memid, dev.io + HP_MEMM); +#endif + msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS); + if (initted) { + mastVolLeft = readw(dev.SMA + SMA_wCurrMastVolLeft); + mastVolRight = readw(dev.SMA + SMA_wCurrMastVolRight); + } else + mastVolLeft = mastVolRight = 0; + memset_io(dev.base, 0, 0x8000); + + /* Critical section: bank 1 access */ + spin_lock_irqsave(&dev.lock, flags); + msnd_outb(HPBLKSEL_1, dev.io + HP_BLKS); + memset_io(dev.base, 0, 0x8000); + msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS); + spin_unlock_irqrestore(&dev.lock, flags); + + dev.pwDSPQData = (dev.base + DSPQ_DATA_BUFF); + dev.pwMODQData = (dev.base + MODQ_DATA_BUFF); + dev.pwMIDQData = (dev.base + MIDQ_DATA_BUFF); + + /* Motorola 56k shared memory base */ + dev.SMA = dev.base + SMA_STRUCT_START; + + /* Digital audio play queue */ + dev.DAPQ = dev.base + DAPQ_OFFSET; + msnd_init_queue(dev.DAPQ, DAPQ_DATA_BUFF, DAPQ_BUFF_SIZE); + + /* Digital audio record queue */ + dev.DARQ = dev.base + DARQ_OFFSET; + msnd_init_queue(dev.DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE); + + /* MIDI out queue */ + dev.MODQ = dev.base + MODQ_OFFSET; + msnd_init_queue(dev.MODQ, MODQ_DATA_BUFF, MODQ_BUFF_SIZE); + + /* MIDI in queue */ + dev.MIDQ = dev.base + MIDQ_OFFSET; + msnd_init_queue(dev.MIDQ, MIDQ_DATA_BUFF, MIDQ_BUFF_SIZE); + + /* DSP -> host message queue */ + dev.DSPQ = dev.base + DSPQ_OFFSET; + msnd_init_queue(dev.DSPQ, DSPQ_DATA_BUFF, DSPQ_BUFF_SIZE); + + /* Setup some DSP values */ +#ifndef MSND_CLASSIC + writew(1, dev.SMA + SMA_wCurrPlayFormat); + writew(dev.play_sample_size, dev.SMA + SMA_wCurrPlaySampleSize); + writew(dev.play_channels, dev.SMA + SMA_wCurrPlayChannels); + writew(dev.play_sample_rate, dev.SMA + SMA_wCurrPlaySampleRate); +#endif + writew(dev.play_sample_rate, dev.SMA + SMA_wCalFreqAtoD); + writew(mastVolLeft, dev.SMA + SMA_wCurrMastVolLeft); + writew(mastVolRight, dev.SMA + SMA_wCurrMastVolRight); +#ifndef MSND_CLASSIC + writel(0x00010000, dev.SMA + SMA_dwCurrPlayPitch); + writel(0x00000001, dev.SMA + SMA_dwCurrPlayRate); +#endif + writew(0x303, dev.SMA + SMA_wCurrInputTagBits); + + initted = 1; + + return 0; +} + +static int __init calibrate_adc(WORD srate) +{ + writew(srate, dev.SMA + SMA_wCalFreqAtoD); + if (dev.calibrate_signal == 0) + writew(readw(dev.SMA + SMA_wCurrHostStatusFlags) + | 0x0001, dev.SMA + SMA_wCurrHostStatusFlags); + else + writew(readw(dev.SMA + SMA_wCurrHostStatusFlags) + & ~0x0001, dev.SMA + SMA_wCurrHostStatusFlags); + if (msnd_send_word(&dev, 0, 0, HDEXAR_CAL_A_TO_D) == 0 && + chk_send_dsp_cmd(&dev, HDEX_AUX_REQ) == 0) { + current->state = TASK_INTERRUPTIBLE; + schedule_timeout(HZ / 3); + return 0; + } + printk(KERN_WARNING LOGNAME ": ADC calibration failed\n"); + + return -EIO; +} + +static int upload_dsp_code(void) +{ + msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS); +#ifndef HAVE_DSPCODEH + INITCODESIZE = mod_firmware_load(INITCODEFILE, &INITCODE); + if (!INITCODE) { + printk(KERN_ERR LOGNAME ": Error loading " INITCODEFILE); + return -EBUSY; + } + + PERMCODESIZE = mod_firmware_load(PERMCODEFILE, &PERMCODE); + if (!PERMCODE) { + printk(KERN_ERR LOGNAME ": Error loading " PERMCODEFILE); + vfree(INITCODE); + return -EBUSY; + } +#endif + memcpy_toio(dev.base, PERMCODE, PERMCODESIZE); + if (msnd_upload_host(&dev, INITCODE, INITCODESIZE) < 0) { + printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n"); + return -ENODEV; + } +#ifdef HAVE_DSPCODEH + printk(KERN_INFO LOGNAME ": DSP firmware uploaded (resident)\n"); +#else + printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n"); +#endif + +#ifndef HAVE_DSPCODEH + vfree(INITCODE); + vfree(PERMCODE); +#endif + + return 0; +} + +#ifdef MSND_CLASSIC +static void reset_proteus(void) +{ + msnd_outb(HPPRORESET_ON, dev.io + HP_PROR); + mdelay(TIME_PRO_RESET); + msnd_outb(HPPRORESET_OFF, dev.io + HP_PROR); + mdelay(TIME_PRO_RESET_DONE); +} +#endif + +static int initialize(void) +{ + int err, timeout; + +#ifdef MSND_CLASSIC + msnd_outb(HPWAITSTATE_0, dev.io + HP_WAIT); + msnd_outb(HPBITMODE_16, dev.io + HP_BITM); + + reset_proteus(); +#endif + if ((err = init_sma()) < 0) { + printk(KERN_WARNING LOGNAME ": Cannot initialize SMA\n"); + return err; + } + + if ((err = reset_dsp()) < 0) + return err; + + if ((err = upload_dsp_code()) < 0) { + printk(KERN_WARNING LOGNAME ": Cannot upload DSP code\n"); + return err; + } + + timeout = 200; + while (readw(dev.base)) { + mdelay(1); + if (!timeout--) { + printk(KERN_DEBUG LOGNAME ": DSP reset timeout\n"); + return -EIO; + } + } + + mixer_setup(); + + return 0; +} + +static int dsp_full_reset(void) +{ + int rv; + + if (test_bit(F_RESETTING, &dev.flags) || ++dev.nresets > 10) + return 0; + + set_bit(F_RESETTING, &dev.flags); + printk(KERN_INFO LOGNAME ": DSP reset\n"); + dsp_halt(NULL); /* Unconditionally halt */ + if ((rv = initialize())) + printk(KERN_WARNING LOGNAME ": DSP reset failed\n"); + force_recsrc(dev.recsrc); + dsp_open(NULL); + clear_bit(F_RESETTING, &dev.flags); + + return rv; +} + +static int __init attach_multisound(void) +{ + int err; + + if ((err = request_irq(dev.irq, intr, 0, dev.name, &dev)) < 0) { + printk(KERN_ERR LOGNAME ": Couldn't grab IRQ %d\n", dev.irq); + return err; + } + if (request_region(dev.io, dev.numio, dev.name) == NULL) { + free_irq(dev.irq, &dev); + return -EBUSY; + } + + err = dsp_full_reset(); + if (err < 0) { + release_region(dev.io, dev.numio); + free_irq(dev.irq, &dev); + return err; + } + + if ((err = msnd_register(&dev)) < 0) { + printk(KERN_ERR LOGNAME ": Unable to register MultiSound\n"); + release_region(dev.io, dev.numio); + free_irq(dev.irq, &dev); + return err; + } + + if ((dev.dsp_minor = register_sound_dsp(&dev_fileops, -1)) < 0) { + printk(KERN_ERR LOGNAME ": Unable to register DSP operations\n"); + msnd_unregister(&dev); + release_region(dev.io, dev.numio); + free_irq(dev.irq, &dev); + return dev.dsp_minor; + } + + if ((dev.mixer_minor = register_sound_mixer(&dev_fileops, -1)) < 0) { + printk(KERN_ERR LOGNAME ": Unable to register mixer operations\n"); + unregister_sound_mixer(dev.mixer_minor); + msnd_unregister(&dev); + release_region(dev.io, dev.numio); + free_irq(dev.irq, &dev); + return dev.mixer_minor; + } + + dev.ext_midi_dev = dev.hdr_midi_dev = -1; + + disable_irq(dev.irq); + calibrate_adc(dev.play_sample_rate); +#ifndef MSND_CLASSIC + force_recsrc(SOUND_MASK_IMIX); +#endif + + return 0; +} + +static void __exit unload_multisound(void) +{ + release_region(dev.io, dev.numio); + free_irq(dev.irq, &dev); + unregister_sound_mixer(dev.mixer_minor); + unregister_sound_dsp(dev.dsp_minor); + msnd_unregister(&dev); +} + +#ifndef MSND_CLASSIC + +/* Pinnacle/Fiji Logical Device Configuration */ + +static int __init msnd_write_cfg(int cfg, int reg, int value) +{ + msnd_outb(reg, cfg); + msnd_outb(value, cfg + 1); + if (value != msnd_inb(cfg + 1)) { + printk(KERN_ERR LOGNAME ": msnd_write_cfg: I/O error\n"); + return -EIO; + } + return 0; +} + +static int __init msnd_write_cfg_io0(int cfg, int num, WORD io) +{ + if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io))) + return -EIO; + if (msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io))) + return -EIO; + return 0; +} + +static int __init msnd_write_cfg_io1(int cfg, int num, WORD io) +{ + if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io))) + return -EIO; + if (msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io))) + return -EIO; + return 0; +} + +static int __init msnd_write_cfg_irq(int cfg, int num, WORD irq) +{ + if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq))) + return -EIO; + if (msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE)) + return -EIO; + return 0; +} + +static int __init msnd_write_cfg_mem(int cfg, int num, int mem) +{ + WORD wmem; + + mem >>= 8; + mem &= 0xfff; + wmem = (WORD)mem; + if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem))) + return -EIO; + if (msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem))) + return -EIO; + if (wmem && msnd_write_cfg(cfg, IREG_MEMCONTROL, (MEMTYPE_HIADDR | MEMTYPE_16BIT))) + return -EIO; + return 0; +} + +static int __init msnd_activate_logical(int cfg, int num) +{ + if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE)) + return -EIO; + return 0; +} + +static int __init msnd_write_cfg_logical(int cfg, int num, WORD io0, WORD io1, WORD irq, int mem) +{ + if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (msnd_write_cfg_io0(cfg, num, io0)) + return -EIO; + if (msnd_write_cfg_io1(cfg, num, io1)) + return -EIO; + if (msnd_write_cfg_irq(cfg, num, irq)) + return -EIO; + if (msnd_write_cfg_mem(cfg, num, mem)) + return -EIO; + if (msnd_activate_logical(cfg, num)) + return -EIO; + return 0; +} + +typedef struct msnd_pinnacle_cfg_device { + WORD io0, io1, irq; + int mem; +} msnd_pinnacle_cfg_t[4]; + +static int __init msnd_pinnacle_cfg_devices(int cfg, int reset, msnd_pinnacle_cfg_t device) +{ + int i; + + /* Reset devices if told to */ + if (reset) { + printk(KERN_INFO LOGNAME ": Resetting all devices\n"); + for (i = 0; i < 4; ++i) + if (msnd_write_cfg_logical(cfg, i, 0, 0, 0, 0)) + return -EIO; + } + + /* Configure specified devices */ + for (i = 0; i < 4; ++i) { + + switch (i) { + case 0: /* DSP */ + if (!(device[i].io0 && device[i].irq && device[i].mem)) + continue; + break; + case 1: /* MPU */ + if (!(device[i].io0 && device[i].irq)) + continue; + printk(KERN_INFO LOGNAME + ": Configuring MPU to I/O 0x%x IRQ %d\n", + device[i].io0, device[i].irq); + break; + case 2: /* IDE */ + if (!(device[i].io0 && device[i].io1 && device[i].irq)) + continue; + printk(KERN_INFO LOGNAME + ": Configuring IDE to I/O 0x%x, 0x%x IRQ %d\n", + device[i].io0, device[i].io1, device[i].irq); + break; + case 3: /* Joystick */ + if (!(device[i].io0)) + continue; + printk(KERN_INFO LOGNAME + ": Configuring joystick to I/O 0x%x\n", + device[i].io0); + break; + } + + /* Configure the device */ + if (msnd_write_cfg_logical(cfg, i, device[i].io0, device[i].io1, device[i].irq, device[i].mem)) + return -EIO; + } + + return 0; +} +#endif + +#ifdef MODULE +MODULE_AUTHOR ("Andrew Veliath <andrewtv@usa.net>"); +MODULE_DESCRIPTION ("Turtle Beach " LONGNAME " Linux Driver"); +MODULE_LICENSE("GPL"); + +static int io __initdata = -1; +static int irq __initdata = -1; +static int mem __initdata = -1; +static int write_ndelay __initdata = -1; + +#ifndef MSND_CLASSIC +/* Pinnacle/Fiji non-PnP Config Port */ +static int cfg __initdata = -1; + +/* Extra Peripheral Configuration */ +static int reset __initdata = 0; +static int mpu_io __initdata = 0; +static int mpu_irq __initdata = 0; +static int ide_io0 __initdata = 0; +static int ide_io1 __initdata = 0; +static int ide_irq __initdata = 0; +static int joystick_io __initdata = 0; + +/* If we have the digital daugherboard... */ +static int digital __initdata = 0; +#endif + +static int fifosize __initdata = DEFFIFOSIZE; +static int calibrate_signal __initdata = 0; + +#else /* not a module */ + +static int write_ndelay __initdata = -1; + +#ifdef MSND_CLASSIC +static int io __initdata = CONFIG_MSNDCLAS_IO; +static int irq __initdata = CONFIG_MSNDCLAS_IRQ; +static int mem __initdata = CONFIG_MSNDCLAS_MEM; +#else /* Pinnacle/Fiji */ + +static int io __initdata = CONFIG_MSNDPIN_IO; +static int irq __initdata = CONFIG_MSNDPIN_IRQ; +static int mem __initdata = CONFIG_MSNDPIN_MEM; + +/* Pinnacle/Fiji non-PnP Config Port */ +#ifdef CONFIG_MSNDPIN_NONPNP +# ifndef CONFIG_MSNDPIN_CFG +# define CONFIG_MSNDPIN_CFG 0x250 +# endif +#else +# ifdef CONFIG_MSNDPIN_CFG +# undef CONFIG_MSNDPIN_CFG +# endif +# define CONFIG_MSNDPIN_CFG -1 +#endif +static int cfg __initdata = CONFIG_MSNDPIN_CFG; +/* If not a module, we don't need to bother with reset=1 */ +static int reset; + +/* Extra Peripheral Configuration (Default: Disable) */ +#ifndef CONFIG_MSNDPIN_MPU_IO +# define CONFIG_MSNDPIN_MPU_IO 0 +#endif +static int mpu_io __initdata = CONFIG_MSNDPIN_MPU_IO; + +#ifndef CONFIG_MSNDPIN_MPU_IRQ +# define CONFIG_MSNDPIN_MPU_IRQ 0 +#endif +static int mpu_irq __initdata = CONFIG_MSNDPIN_MPU_IRQ; + +#ifndef CONFIG_MSNDPIN_IDE_IO0 +# define CONFIG_MSNDPIN_IDE_IO0 0 +#endif +static int ide_io0 __initdata = CONFIG_MSNDPIN_IDE_IO0; + +#ifndef CONFIG_MSNDPIN_IDE_IO1 +# define CONFIG_MSNDPIN_IDE_IO1 0 +#endif +static int ide_io1 __initdata = CONFIG_MSNDPIN_IDE_IO1; + +#ifndef CONFIG_MSNDPIN_IDE_IRQ +# define CONFIG_MSNDPIN_IDE_IRQ 0 +#endif +static int ide_irq __initdata = CONFIG_MSNDPIN_IDE_IRQ; + +#ifndef CONFIG_MSNDPIN_JOYSTICK_IO +# define CONFIG_MSNDPIN_JOYSTICK_IO 0 +#endif +static int joystick_io __initdata = CONFIG_MSNDPIN_JOYSTICK_IO; + +/* Have SPDIF (Digital) Daughterboard */ +#ifndef CONFIG_MSNDPIN_DIGITAL +# define CONFIG_MSNDPIN_DIGITAL 0 +#endif +static int digital __initdata = CONFIG_MSNDPIN_DIGITAL; + +#endif /* MSND_CLASSIC */ + +#ifndef CONFIG_MSND_FIFOSIZE +# define CONFIG_MSND_FIFOSIZE DEFFIFOSIZE +#endif +static int fifosize __initdata = CONFIG_MSND_FIFOSIZE; + +#ifndef CONFIG_MSND_CALSIGNAL +# define CONFIG_MSND_CALSIGNAL 0 +#endif +static int +calibrate_signal __initdata = CONFIG_MSND_CALSIGNAL; +#endif /* MODULE */ + +module_param (io, int, 0); +module_param (irq, int, 0); +module_param (mem, int, 0); +module_param (write_ndelay, int, 0); +module_param (fifosize, int, 0); +module_param (calibrate_signal, int, 0); +#ifndef MSND_CLASSIC +module_param (digital, bool, 0); +module_param (cfg, int, 0); +module_param (reset, int, 0); +module_param (mpu_io, int, 0); +module_param (mpu_irq, int, 0); +module_param (ide_io0, int, 0); +module_param (ide_io1, int, 0); +module_param (ide_irq, int, 0); +module_param (joystick_io, int, 0); +#endif + +static int __init msnd_init(void) +{ + int err; +#ifndef MSND_CLASSIC + static msnd_pinnacle_cfg_t pinnacle_devs; +#endif /* MSND_CLASSIC */ + + printk(KERN_INFO LOGNAME ": Turtle Beach " LONGNAME " Linux Driver Version " + VERSION ", Copyright (C) 1998 Andrew Veliath\n"); + + if (io == -1 || irq == -1 || mem == -1) + printk(KERN_WARNING LOGNAME ": io, irq and mem must be set\n"); + +#ifdef MSND_CLASSIC + if (io == -1 || + !(io == 0x290 || + io == 0x260 || + io == 0x250 || + io == 0x240 || + io == 0x230 || + io == 0x220 || + io == 0x210 || + io == 0x3e0)) { + printk(KERN_ERR LOGNAME ": \"io\" - DSP I/O base must be set to 0x210, 0x220, 0x230, 0x240, 0x250, 0x260, 0x290, or 0x3E0\n"); + return -EINVAL; + } +#else + if (io == -1 || + io < 0x100 || + io > 0x3e0 || + (io % 0x10) != 0) { + printk(KERN_ERR LOGNAME ": \"io\" - DSP I/O base must within the range 0x100 to 0x3E0 and must be evenly divisible by 0x10\n"); + return -EINVAL; + } +#endif /* MSND_CLASSIC */ + + if (irq == -1 || + !(irq == 5 || + irq == 7 || + irq == 9 || + irq == 10 || + irq == 11 || + irq == 12)) { + printk(KERN_ERR LOGNAME ": \"irq\" - must be set to 5, 7, 9, 10, 11 or 12\n"); + return -EINVAL; + } + + if (mem == -1 || + !(mem == 0xb0000 || + mem == 0xc8000 || + mem == 0xd0000 || + mem == 0xd8000 || + mem == 0xe0000 || + mem == 0xe8000)) { + printk(KERN_ERR LOGNAME ": \"mem\" - must be set to " + "0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000\n"); + return -EINVAL; + } + +#ifdef MSND_CLASSIC + switch (irq) { + case 5: dev.irqid = HPIRQ_5; break; + case 7: dev.irqid = HPIRQ_7; break; + case 9: dev.irqid = HPIRQ_9; break; + case 10: dev.irqid = HPIRQ_10; break; + case 11: dev.irqid = HPIRQ_11; break; + case 12: dev.irqid = HPIRQ_12; break; + } + + switch (mem) { + case 0xb0000: dev.memid = HPMEM_B000; break; + case 0xc8000: dev.memid = HPMEM_C800; break; + case 0xd0000: dev.memid = HPMEM_D000; break; + case 0xd8000: dev.memid = HPMEM_D800; break; + case 0xe0000: dev.memid = HPMEM_E000; break; + case 0xe8000: dev.memid = HPMEM_E800; break; + } +#else + if (cfg == -1) { + printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); + } else if (cfg != 0x250 && cfg != 0x260 && cfg != 0x270) { + printk(KERN_INFO LOGNAME ": Config port must be 0x250, 0x260 or 0x270 (or unspecified for PnP mode)\n"); + return -EINVAL; + } else { + printk(KERN_INFO LOGNAME ": Non-PnP mode: configuring at port 0x%x\n", cfg); + + /* DSP */ + pinnacle_devs[0].io0 = io; + pinnacle_devs[0].irq = irq; + pinnacle_devs[0].mem = mem; + + /* The following are Pinnacle specific */ + + /* MPU */ + pinnacle_devs[1].io0 = mpu_io; + pinnacle_devs[1].irq = mpu_irq; + + /* IDE */ + pinnacle_devs[2].io0 = ide_io0; + pinnacle_devs[2].io1 = ide_io1; + pinnacle_devs[2].irq = ide_irq; + + /* Joystick */ + pinnacle_devs[3].io0 = joystick_io; + + if (!request_region(cfg, 2, "Pinnacle/Fiji Config")) { + printk(KERN_ERR LOGNAME ": Config port 0x%x conflict\n", cfg); + return -EIO; + } + + if (msnd_pinnacle_cfg_devices(cfg, reset, pinnacle_devs)) { + printk(KERN_ERR LOGNAME ": Device configuration error\n"); + release_region(cfg, 2); + return -EIO; + } + release_region(cfg, 2); + } +#endif /* MSND_CLASSIC */ + + if (fifosize < 16) + fifosize = 16; + + if (fifosize > 1024) + fifosize = 1024; + + set_default_audio_parameters(); +#ifdef MSND_CLASSIC + dev.type = msndClassic; +#else + dev.type = msndPinnacle; +#endif + dev.io = io; + dev.numio = DSP_NUMIO; + dev.irq = irq; + dev.base = ioremap(mem, 0x8000); + dev.fifosize = fifosize * 1024; + dev.calibrate_signal = calibrate_signal ? 1 : 0; + dev.recsrc = 0; + dev.dspq_data_buff = DSPQ_DATA_BUFF; + dev.dspq_buff_size = DSPQ_BUFF_SIZE; + if (write_ndelay == -1) + write_ndelay = CONFIG_MSND_WRITE_NDELAY; + if (write_ndelay) + clear_bit(F_DISABLE_WRITE_NDELAY, &dev.flags); + else + set_bit(F_DISABLE_WRITE_NDELAY, &dev.flags); +#ifndef MSND_CLASSIC + if (digital) + set_bit(F_HAVEDIGITAL, &dev.flags); +#endif + init_waitqueue_head(&dev.writeblock); + init_waitqueue_head(&dev.readblock); + init_waitqueue_head(&dev.writeflush); + msnd_fifo_init(&dev.DAPF); + msnd_fifo_init(&dev.DARF); + spin_lock_init(&dev.lock); + printk(KERN_INFO LOGNAME ": %u byte audio FIFOs (x2)\n", dev.fifosize); + if ((err = msnd_fifo_alloc(&dev.DAPF, dev.fifosize)) < 0) { + printk(KERN_ERR LOGNAME ": Couldn't allocate write FIFO\n"); + return err; + } + + if ((err = msnd_fifo_alloc(&dev.DARF, dev.fifosize)) < 0) { + printk(KERN_ERR LOGNAME ": Couldn't allocate read FIFO\n"); + msnd_fifo_free(&dev.DAPF); + return err; + } + + if ((err = probe_multisound()) < 0) { + printk(KERN_ERR LOGNAME ": Probe failed\n"); + msnd_fifo_free(&dev.DAPF); + msnd_fifo_free(&dev.DARF); + return err; + } + + if ((err = attach_multisound()) < 0) { + printk(KERN_ERR LOGNAME ": Attach failed\n"); + msnd_fifo_free(&dev.DAPF); + msnd_fifo_free(&dev.DARF); + return err; + } + + return 0; +} + +static void __exit msdn_cleanup(void) +{ + unload_multisound(); + msnd_fifo_free(&dev.DAPF); + msnd_fifo_free(&dev.DARF); +} + +module_init(msnd_init); +module_exit(msdn_cleanup); diff --git a/sound/oss/msnd_pinnacle.h b/sound/oss/msnd_pinnacle.h new file mode 100644 index 00000000..c18d66cb --- /dev/null +++ b/sound/oss/msnd_pinnacle.h @@ -0,0 +1,246 @@ +/********************************************************************* + * + * msnd_pinnacle.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_PINNACLE_H +#define __MSND_PINNACLE_H + + +#define DSP_NUMIO 0x08 + +#define IREG_LOGDEVICE 0x07 +#define IREG_ACTIVATE 0x30 +#define LD_ACTIVATE 0x01 +#define LD_DISACTIVATE 0x00 +#define IREG_EECONTROL 0x3F +#define IREG_MEMBASEHI 0x40 +#define IREG_MEMBASELO 0x41 +#define IREG_MEMCONTROL 0x42 +#define IREG_MEMRANGEHI 0x43 +#define IREG_MEMRANGELO 0x44 +#define MEMTYPE_8BIT 0x00 +#define MEMTYPE_16BIT 0x02 +#define MEMTYPE_RANGE 0x00 +#define MEMTYPE_HIADDR 0x01 +#define IREG_IO0_BASEHI 0x60 +#define IREG_IO0_BASELO 0x61 +#define IREG_IO1_BASEHI 0x62 +#define IREG_IO1_BASELO 0x63 +#define IREG_IRQ_NUMBER 0x70 +#define IREG_IRQ_TYPE 0x71 +#define IRQTYPE_HIGH 0x02 +#define IRQTYPE_LOW 0x00 +#define IRQTYPE_LEVEL 0x01 +#define IRQTYPE_EDGE 0x00 + +#define HP_DSPR 0x04 +#define HP_BLKS 0x04 + +#define HPDSPRESET_OFF 2 +#define HPDSPRESET_ON 0 + +#define HPBLKSEL_0 2 +#define HPBLKSEL_1 3 + +#define HIMT_DAT_OFF 0x03 + +#define HIDSP_PLAY_UNDER 0x00 +#define HIDSP_INT_PLAY_UNDER 0x01 +#define HIDSP_SSI_TX_UNDER 0x02 +#define HIDSP_RECQ_OVERFLOW 0x08 +#define HIDSP_INT_RECORD_OVER 0x09 +#define HIDSP_SSI_RX_OVERFLOW 0x0a + +#define HIDSP_MIDI_IN_OVER 0x10 + +#define HIDSP_MIDI_FRAME_ERR 0x11 +#define HIDSP_MIDI_PARITY_ERR 0x12 +#define HIDSP_MIDI_OVERRUN_ERR 0x13 + +#define HIDSP_INPUT_CLIPPING 0x20 +#define HIDSP_MIX_CLIPPING 0x30 +#define HIDSP_DAT_IN_OFF 0x21 + +#define HDEXAR_SET_ANA_IN 0 +#define HDEXAR_CLEAR_PEAKS 1 +#define HDEXAR_IN_SET_POTS 2 +#define HDEXAR_AUX_SET_POTS 3 +#define HDEXAR_CAL_A_TO_D 4 +#define HDEXAR_RD_EXT_DSP_BITS 5 + +#define HDEXAR_SET_SYNTH_IN 4 +#define HDEXAR_READ_DAT_IN 5 +#define HDEXAR_MIC_SET_POTS 6 +#define HDEXAR_SET_DAT_IN 7 + +#define HDEXAR_SET_SYNTH_48 8 +#define HDEXAR_SET_SYNTH_44 9 + +#define TIME_PRO_RESET_DONE 0x028A +#define TIME_PRO_SYSEX 0x001E +#define TIME_PRO_RESET 0x0032 + +#define AGND 0x01 +#define SIGNAL 0x02 + +#define EXT_DSP_BIT_DCAL 0x0001 +#define EXT_DSP_BIT_MIDI_CON 0x0002 + +#define BUFFSIZE 0x8000 +#define HOSTQ_SIZE 0x40 + +#define SRAM_CNTL_START 0x7F00 +#define SMA_STRUCT_START 0x7F40 + +#define DAP_BUFF_SIZE 0x2400 +#define DAR_BUFF_SIZE 0x2000 + +#define DAPQ_STRUCT_SIZE 0x10 +#define DARQ_STRUCT_SIZE 0x10 +#define DAPQ_BUFF_SIZE (3 * 0x10) +#define DARQ_BUFF_SIZE (3 * 0x10) +#define MODQ_BUFF_SIZE 0x400 +#define MIDQ_BUFF_SIZE 0x800 +#define DSPQ_BUFF_SIZE 0x5A0 + +#define DAPQ_DATA_BUFF 0x6C00 +#define DARQ_DATA_BUFF 0x6C30 +#define MODQ_DATA_BUFF 0x6C60 +#define MIDQ_DATA_BUFF 0x7060 +#define DSPQ_DATA_BUFF 0x7860 + +#define DAPQ_OFFSET SRAM_CNTL_START +#define DARQ_OFFSET (SRAM_CNTL_START + 0x08) +#define MODQ_OFFSET (SRAM_CNTL_START + 0x10) +#define MIDQ_OFFSET (SRAM_CNTL_START + 0x18) +#define DSPQ_OFFSET (SRAM_CNTL_START + 0x20) + +#define MOP_WAVEHDR 0 +#define MOP_EXTOUT 1 +#define MOP_HWINIT 0xfe +#define MOP_NONE 0xff +#define MOP_MAX 1 + +#define MIP_EXTIN 0 +#define MIP_WAVEHDR 1 +#define MIP_HWINIT 0xfe +#define MIP_MAX 1 + +/* Pinnacle/Fiji SMA Common Data */ +#define SMA_wCurrPlayBytes 0x0000 +#define SMA_wCurrRecordBytes 0x0002 +#define SMA_wCurrPlayVolLeft 0x0004 +#define SMA_wCurrPlayVolRight 0x0006 +#define SMA_wCurrInVolLeft 0x0008 +#define SMA_wCurrInVolRight 0x000a +#define SMA_wCurrMHdrVolLeft 0x000c +#define SMA_wCurrMHdrVolRight 0x000e +#define SMA_dwCurrPlayPitch 0x0010 +#define SMA_dwCurrPlayRate 0x0014 +#define SMA_wCurrMIDIIOPatch 0x0018 +#define SMA_wCurrPlayFormat 0x001a +#define SMA_wCurrPlaySampleSize 0x001c +#define SMA_wCurrPlayChannels 0x001e +#define SMA_wCurrPlaySampleRate 0x0020 +#define SMA_wCurrRecordFormat 0x0022 +#define SMA_wCurrRecordSampleSize 0x0024 +#define SMA_wCurrRecordChannels 0x0026 +#define SMA_wCurrRecordSampleRate 0x0028 +#define SMA_wCurrDSPStatusFlags 0x002a +#define SMA_wCurrHostStatusFlags 0x002c +#define SMA_wCurrInputTagBits 0x002e +#define SMA_wCurrLeftPeak 0x0030 +#define SMA_wCurrRightPeak 0x0032 +#define SMA_bMicPotPosLeft 0x0034 +#define SMA_bMicPotPosRight 0x0035 +#define SMA_bMicPotMaxLeft 0x0036 +#define SMA_bMicPotMaxRight 0x0037 +#define SMA_bInPotPosLeft 0x0038 +#define SMA_bInPotPosRight 0x0039 +#define SMA_bAuxPotPosLeft 0x003a +#define SMA_bAuxPotPosRight 0x003b +#define SMA_bInPotMaxLeft 0x003c +#define SMA_bInPotMaxRight 0x003d +#define SMA_bAuxPotMaxLeft 0x003e +#define SMA_bAuxPotMaxRight 0x003f +#define SMA_bInPotMaxMethod 0x0040 +#define SMA_bAuxPotMaxMethod 0x0041 +#define SMA_wCurrMastVolLeft 0x0042 +#define SMA_wCurrMastVolRight 0x0044 +#define SMA_wCalFreqAtoD 0x0046 +#define SMA_wCurrAuxVolLeft 0x0048 +#define SMA_wCurrAuxVolRight 0x004a +#define SMA_wCurrPlay1VolLeft 0x004c +#define SMA_wCurrPlay1VolRight 0x004e +#define SMA_wCurrPlay2VolLeft 0x0050 +#define SMA_wCurrPlay2VolRight 0x0052 +#define SMA_wCurrPlay3VolLeft 0x0054 +#define SMA_wCurrPlay3VolRight 0x0056 +#define SMA_wCurrPlay4VolLeft 0x0058 +#define SMA_wCurrPlay4VolRight 0x005a +#define SMA_wCurrPlay1PeakLeft 0x005c +#define SMA_wCurrPlay1PeakRight 0x005e +#define SMA_wCurrPlay2PeakLeft 0x0060 +#define SMA_wCurrPlay2PeakRight 0x0062 +#define SMA_wCurrPlay3PeakLeft 0x0064 +#define SMA_wCurrPlay3PeakRight 0x0066 +#define SMA_wCurrPlay4PeakLeft 0x0068 +#define SMA_wCurrPlay4PeakRight 0x006a +#define SMA_wCurrPlayPeakLeft 0x006c +#define SMA_wCurrPlayPeakRight 0x006e +#define SMA_wCurrDATSR 0x0070 +#define SMA_wCurrDATRXCHNL 0x0072 +#define SMA_wCurrDATTXCHNL 0x0074 +#define SMA_wCurrDATRXRate 0x0076 +#define SMA_dwDSPPlayCount 0x0078 +#define SMA__size 0x007c + +#ifdef HAVE_DSPCODEH +# include "pndsperm.c" +# include "pndspini.c" +# define PERMCODE pndsperm +# define INITCODE pndspini +# define PERMCODESIZE sizeof(pndsperm) +# define INITCODESIZE sizeof(pndspini) +#else +# ifndef CONFIG_MSNDPIN_INIT_FILE +# define CONFIG_MSNDPIN_INIT_FILE \ + "/etc/sound/pndspini.bin" +# endif +# ifndef CONFIG_MSNDPIN_PERM_FILE +# define CONFIG_MSNDPIN_PERM_FILE \ + "/etc/sound/pndsperm.bin" +# endif +# define PERMCODEFILE CONFIG_MSNDPIN_PERM_FILE +# define INITCODEFILE CONFIG_MSNDPIN_INIT_FILE +# define PERMCODE dspini +# define INITCODE permini +# define PERMCODESIZE sizeof_dspini +# define INITCODESIZE sizeof_permini +#endif +#define LONGNAME "MultiSound (Pinnacle/Fiji)" + +#endif /* __MSND_PINNACLE_H */ diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c new file mode 100644 index 00000000..407cd677 --- /dev/null +++ b/sound/oss/opl3.c @@ -0,0 +1,1258 @@ +/* + * sound/oss/opl3.c + * + * A low level driver for Yamaha YM3812 and OPL-3 -chips + * + * + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * + * + * Changes + * Thomas Sailer ioctl code reworked (vmalloc/vfree removed) + * Alan Cox modularisation, fixed sound_mem allocs. + * Christoph Hellwig Adapted to module_init/module_exit + * Arnaldo C. de Melo get rid of check_region, use request_region for + * OPL4, release it on exit, some cleanups. + * + * Status + * Believed to work. Badly needs rewriting a bit to support multiple + * OPL3 devices. + */ + +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/delay.h> + +/* + * Major improvements to the FM handling 30AUG92 by Rob Hooft, + * hooft@chem.ruu.nl + */ + +#include "sound_config.h" + +#include "opl3_hw.h" + +#define MAX_VOICE 18 +#define OFFS_4OP 11 + +struct voice_info +{ + unsigned char keyon_byte; + long bender; + long bender_range; + unsigned long orig_freq; + unsigned long current_freq; + int volume; + int mode; + int panning; /* 0xffff means not set */ +}; + +typedef struct opl_devinfo +{ + int base; + int left_io, right_io; + int nr_voice; + int lv_map[MAX_VOICE]; + + struct voice_info voc[MAX_VOICE]; + struct voice_alloc_info *v_alloc; + struct channel_info *chn_info; + + struct sbi_instrument i_map[SBFM_MAXINSTR]; + struct sbi_instrument *act_i[MAX_VOICE]; + + struct synth_info fm_info; + + int busy; + int model; + unsigned char cmask; + + int is_opl4; +} opl_devinfo; + +static struct opl_devinfo *devc = NULL; + +static int detected_model; + +static int store_instr(int instr_no, struct sbi_instrument *instr); +static void freq_to_fnum(int freq, int *block, int *fnum); +static void opl3_command(int io_addr, unsigned int addr, unsigned int val); +static int opl3_kill_note(int dev, int voice, int note, int velocity); + +static void enter_4op_mode(void) +{ + int i; + static int v4op[MAX_VOICE] = { + 0, 1, 2, 9, 10, 11, 6, 7, 8, 15, 16, 17 + }; + + devc->cmask = 0x3f; /* Connect all possible 4 OP voice operators */ + opl3_command(devc->right_io, CONNECTION_SELECT_REGISTER, 0x3f); + + for (i = 0; i < 3; i++) + pv_map[i].voice_mode = 4; + for (i = 3; i < 6; i++) + pv_map[i].voice_mode = 0; + + for (i = 9; i < 12; i++) + pv_map[i].voice_mode = 4; + for (i = 12; i < 15; i++) + pv_map[i].voice_mode = 0; + + for (i = 0; i < 12; i++) + devc->lv_map[i] = v4op[i]; + devc->v_alloc->max_voice = devc->nr_voice = 12; +} + +static int opl3_ioctl(int dev, unsigned int cmd, void __user * arg) +{ + struct sbi_instrument ins; + + switch (cmd) { + case SNDCTL_FM_LOAD_INSTR: + printk(KERN_WARNING "Warning: Obsolete ioctl(SNDCTL_FM_LOAD_INSTR) used. Fix the program.\n"); + if (copy_from_user(&ins, arg, sizeof(ins))) + return -EFAULT; + if (ins.channel < 0 || ins.channel >= SBFM_MAXINSTR) { + printk(KERN_WARNING "FM Error: Invalid instrument number %d\n", ins.channel); + return -EINVAL; + } + return store_instr(ins.channel, &ins); + + case SNDCTL_SYNTH_INFO: + devc->fm_info.nr_voices = (devc->nr_voice == 12) ? 6 : devc->nr_voice; + if (copy_to_user(arg, &devc->fm_info, sizeof(devc->fm_info))) + return -EFAULT; + return 0; + + case SNDCTL_SYNTH_MEMAVL: + return 0x7fffffff; + + case SNDCTL_FM_4OP_ENABLE: + if (devc->model == 2) + enter_4op_mode(); + return 0; + + default: + return -EINVAL; + } +} + +static int opl3_detect(int ioaddr) +{ + /* + * This function returns 1 if the FM chip is present at the given I/O port + * The detection algorithm plays with the timer built in the FM chip and + * looks for a change in the status register. + * + * Note! The timers of the FM chip are not connected to AdLib (and compatible) + * boards. + * + * Note2! The chip is initialized if detected. + */ + + unsigned char stat1, signature; + int i; + + if (devc != NULL) + { + printk(KERN_ERR "opl3: Only one OPL3 supported.\n"); + return 0; + } + + devc = kzalloc(sizeof(*devc), GFP_KERNEL); + + if (devc == NULL) + { + printk(KERN_ERR "opl3: Can't allocate memory for the device control " + "structure \n "); + return 0; + } + + strcpy(devc->fm_info.name, "OPL2"); + + if (!request_region(ioaddr, 4, devc->fm_info.name)) { + printk(KERN_WARNING "opl3: I/O port 0x%x already in use\n", ioaddr); + goto cleanup_devc; + } + + devc->base = ioaddr; + + /* Reset timers 1 and 2 */ + opl3_command(ioaddr, TIMER_CONTROL_REGISTER, TIMER1_MASK | TIMER2_MASK); + + /* Reset the IRQ of the FM chip */ + opl3_command(ioaddr, TIMER_CONTROL_REGISTER, IRQ_RESET); + + signature = stat1 = inb(ioaddr); /* Status register */ + + if (signature != 0x00 && signature != 0x06 && signature != 0x02 && + signature != 0x0f) + { + MDB(printk(KERN_INFO "OPL3 not detected %x\n", signature)); + goto cleanup_region; + } + + if (signature == 0x06) /* OPL2 */ + { + detected_model = 2; + } + else if (signature == 0x00 || signature == 0x0f) /* OPL3 or OPL4 */ + { + unsigned char tmp; + + detected_model = 3; + + /* + * Detect availability of OPL4 (_experimental_). Works probably + * only after a cold boot. In addition the OPL4 port + * of the chip may not be connected to the PC bus at all. + */ + + opl3_command(ioaddr + 2, OPL3_MODE_REGISTER, 0x00); + opl3_command(ioaddr + 2, OPL3_MODE_REGISTER, OPL3_ENABLE | OPL4_ENABLE); + + if ((tmp = inb(ioaddr)) == 0x02) /* Have a OPL4 */ + { + detected_model = 4; + } + + if (request_region(ioaddr - 8, 2, "OPL4")) /* OPL4 port was free */ + { + int tmp; + + outb((0x02), ioaddr - 8); /* Select OPL4 ID register */ + udelay(10); + tmp = inb(ioaddr - 7); /* Read it */ + udelay(10); + + if (tmp == 0x20) /* OPL4 should return 0x20 here */ + { + detected_model = 4; + outb((0xF8), ioaddr - 8); /* Select OPL4 FM mixer control */ + udelay(10); + outb((0x1B), ioaddr - 7); /* Write value */ + udelay(10); + } + else + { /* release OPL4 port */ + release_region(ioaddr - 8, 2); + detected_model = 3; + } + } + opl3_command(ioaddr + 2, OPL3_MODE_REGISTER, 0); + } + for (i = 0; i < 9; i++) + opl3_command(ioaddr, KEYON_BLOCK + i, 0); /* + * Note off + */ + + opl3_command(ioaddr, TEST_REGISTER, ENABLE_WAVE_SELECT); + opl3_command(ioaddr, PERCOSSION_REGISTER, 0x00); /* + * Melodic mode. + */ + return 1; +cleanup_region: + release_region(ioaddr, 4); +cleanup_devc: + kfree(devc); + devc = NULL; + return 0; +} + +static int opl3_kill_note (int devno, int voice, int note, int velocity) +{ + struct physical_voice_info *map; + + if (voice < 0 || voice >= devc->nr_voice) + return 0; + + devc->v_alloc->map[voice] = 0; + + map = &pv_map[devc->lv_map[voice]]; + DEB(printk("Kill note %d\n", voice)); + + if (map->voice_mode == 0) + return 0; + + opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num, devc->voc[voice].keyon_byte & ~0x20); + devc->voc[voice].keyon_byte = 0; + devc->voc[voice].bender = 0; + devc->voc[voice].volume = 64; + devc->voc[voice].panning = 0xffff; /* Not set */ + devc->voc[voice].bender_range = 200; + devc->voc[voice].orig_freq = 0; + devc->voc[voice].current_freq = 0; + devc->voc[voice].mode = 0; + return 0; +} + +#define HIHAT 0 +#define CYMBAL 1 +#define TOMTOM 2 +#define SNARE 3 +#define BDRUM 4 +#define UNDEFINED TOMTOM +#define DEFAULT TOMTOM + +static int store_instr(int instr_no, struct sbi_instrument *instr) +{ + if (instr->key != FM_PATCH && (instr->key != OPL3_PATCH || devc->model != 2)) + printk(KERN_WARNING "FM warning: Invalid patch format field (key) 0x%x\n", instr->key); + memcpy((char *) &(devc->i_map[instr_no]), (char *) instr, sizeof(*instr)); + return 0; +} + +static int opl3_set_instr (int dev, int voice, int instr_no) +{ + if (voice < 0 || voice >= devc->nr_voice) + return 0; + if (instr_no < 0 || instr_no >= SBFM_MAXINSTR) + instr_no = 0; /* Acoustic piano (usually) */ + + devc->act_i[voice] = &devc->i_map[instr_no]; + return 0; +} + +/* + * The next table looks magical, but it certainly is not. Its values have + * been calculated as table[i]=8*log(i/64)/log(2) with an obvious exception + * for i=0. This log-table converts a linear volume-scaling (0..127) to a + * logarithmic scaling as present in the FM-synthesizer chips. so : Volume + * 64 = 0 db = relative volume 0 and: Volume 32 = -6 db = relative + * volume -8 it was implemented as a table because it is only 128 bytes and + * it saves a lot of log() calculations. (RH) + */ + +static char fm_volume_table[128] = +{ + -64, -48, -40, -35, -32, -29, -27, -26, + -24, -23, -21, -20, -19, -18, -18, -17, + -16, -15, -15, -14, -13, -13, -12, -12, + -11, -11, -10, -10, -10, -9, -9, -8, + -8, -8, -7, -7, -7, -6, -6, -6, + -5, -5, -5, -5, -4, -4, -4, -4, + -3, -3, -3, -3, -2, -2, -2, -2, + -2, -1, -1, -1, -1, 0, 0, 0, + 0, 0, 0, 1, 1, 1, 1, 1, + 1, 2, 2, 2, 2, 2, 2, 2, + 3, 3, 3, 3, 3, 3, 3, 4, + 4, 4, 4, 4, 4, 4, 4, 5, + 5, 5, 5, 5, 5, 5, 5, 5, + 6, 6, 6, 6, 6, 6, 6, 6, + 6, 7, 7, 7, 7, 7, 7, 7, + 7, 7, 7, 8, 8, 8, 8, 8 +}; + +static void calc_vol(unsigned char *regbyte, int volume, int main_vol) +{ + int level = (~*regbyte & 0x3f); + + if (main_vol > 127) + main_vol = 127; + volume = (volume * main_vol) / 127; + + if (level) + level += fm_volume_table[volume]; + + if (level > 0x3f) + level = 0x3f; + if (level < 0) + level = 0; + + *regbyte = (*regbyte & 0xc0) | (~level & 0x3f); +} + +static void set_voice_volume(int voice, int volume, int main_vol) +{ + unsigned char vol1, vol2, vol3, vol4; + struct sbi_instrument *instr; + struct physical_voice_info *map; + + if (voice < 0 || voice >= devc->nr_voice) + return; + + map = &pv_map[devc->lv_map[voice]]; + instr = devc->act_i[voice]; + + if (!instr) + instr = &devc->i_map[0]; + + if (instr->channel < 0) + return; + + if (devc->voc[voice].mode == 0) + return; + + if (devc->voc[voice].mode == 2) + { + vol1 = instr->operators[2]; + vol2 = instr->operators[3]; + if ((instr->operators[10] & 0x01)) + { + calc_vol(&vol1, volume, main_vol); + calc_vol(&vol2, volume, main_vol); + } + else + { + calc_vol(&vol2, volume, main_vol); + } + opl3_command(map->ioaddr, KSL_LEVEL + map->op[0], vol1); + opl3_command(map->ioaddr, KSL_LEVEL + map->op[1], vol2); + } + else + { /* + * 4 OP voice + */ + int connection; + + vol1 = instr->operators[2]; + vol2 = instr->operators[3]; + vol3 = instr->operators[OFFS_4OP + 2]; + vol4 = instr->operators[OFFS_4OP + 3]; + + /* + * The connection method for 4 OP devc->voc is defined by the rightmost + * bits at the offsets 10 and 10+OFFS_4OP + */ + + connection = ((instr->operators[10] & 0x01) << 1) | (instr->operators[10 + OFFS_4OP] & 0x01); + + switch (connection) + { + case 0: + calc_vol(&vol4, volume, main_vol); + break; + + case 1: + calc_vol(&vol2, volume, main_vol); + calc_vol(&vol4, volume, main_vol); + break; + + case 2: + calc_vol(&vol1, volume, main_vol); + calc_vol(&vol4, volume, main_vol); + break; + + case 3: + calc_vol(&vol1, volume, main_vol); + calc_vol(&vol3, volume, main_vol); + calc_vol(&vol4, volume, main_vol); + break; + + default: + ; + } + opl3_command(map->ioaddr, KSL_LEVEL + map->op[0], vol1); + opl3_command(map->ioaddr, KSL_LEVEL + map->op[1], vol2); + opl3_command(map->ioaddr, KSL_LEVEL + map->op[2], vol3); + opl3_command(map->ioaddr, KSL_LEVEL + map->op[3], vol4); + } +} + +static int opl3_start_note (int dev, int voice, int note, int volume) +{ + unsigned char data, fpc; + int block, fnum, freq, voice_mode, pan; + struct sbi_instrument *instr; + struct physical_voice_info *map; + + if (voice < 0 || voice >= devc->nr_voice) + return 0; + + map = &pv_map[devc->lv_map[voice]]; + pan = devc->voc[voice].panning; + + if (map->voice_mode == 0) + return 0; + + if (note == 255) /* + * Just change the volume + */ + { + set_voice_volume(voice, volume, devc->voc[voice].volume); + return 0; + } + + /* + * Kill previous note before playing + */ + + opl3_command(map->ioaddr, KSL_LEVEL + map->op[1], 0xff); /* + * Carrier + * volume to + * min + */ + opl3_command(map->ioaddr, KSL_LEVEL + map->op[0], 0xff); /* + * Modulator + * volume to + */ + + if (map->voice_mode == 4) + { + opl3_command(map->ioaddr, KSL_LEVEL + map->op[2], 0xff); + opl3_command(map->ioaddr, KSL_LEVEL + map->op[3], 0xff); + } + + opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num, 0x00); /* + * Note + * off + */ + + instr = devc->act_i[voice]; + + if (!instr) + instr = &devc->i_map[0]; + + if (instr->channel < 0) + { + printk(KERN_WARNING "opl3: Initializing voice %d with undefined instrument\n", voice); + return 0; + } + + if (map->voice_mode == 2 && instr->key == OPL3_PATCH) + return 0; /* + * Cannot play + */ + + voice_mode = map->voice_mode; + + if (voice_mode == 4) + { + int voice_shift; + + voice_shift = (map->ioaddr == devc->left_io) ? 0 : 3; + voice_shift += map->voice_num; + + if (instr->key != OPL3_PATCH) /* + * Just 2 OP patch + */ + { + voice_mode = 2; + devc->cmask &= ~(1 << voice_shift); + } + else + { + devc->cmask |= (1 << voice_shift); + } + + opl3_command(devc->right_io, CONNECTION_SELECT_REGISTER, devc->cmask); + } + + /* + * Set Sound Characteristics + */ + + opl3_command(map->ioaddr, AM_VIB + map->op[0], instr->operators[0]); + opl3_command(map->ioaddr, AM_VIB + map->op[1], instr->operators[1]); + + /* + * Set Attack/Decay + */ + + opl3_command(map->ioaddr, ATTACK_DECAY + map->op[0], instr->operators[4]); + opl3_command(map->ioaddr, ATTACK_DECAY + map->op[1], instr->operators[5]); + + /* + * Set Sustain/Release + */ + + opl3_command(map->ioaddr, SUSTAIN_RELEASE + map->op[0], instr->operators[6]); + opl3_command(map->ioaddr, SUSTAIN_RELEASE + map->op[1], instr->operators[7]); + + /* + * Set Wave Select + */ + + opl3_command(map->ioaddr, WAVE_SELECT + map->op[0], instr->operators[8]); + opl3_command(map->ioaddr, WAVE_SELECT + map->op[1], instr->operators[9]); + + /* + * Set Feedback/Connection + */ + + fpc = instr->operators[10]; + + if (pan != 0xffff) + { + fpc &= ~STEREO_BITS; + if (pan < -64) + fpc |= VOICE_TO_LEFT; + else + if (pan > 64) + fpc |= VOICE_TO_RIGHT; + else + fpc |= (VOICE_TO_LEFT | VOICE_TO_RIGHT); + } + + if (!(fpc & 0x30)) + fpc |= 0x30; /* + * Ensure that at least one chn is enabled + */ + opl3_command(map->ioaddr, FEEDBACK_CONNECTION + map->voice_num, fpc); + + /* + * If the voice is a 4 OP one, initialize the operators 3 and 4 also + */ + + if (voice_mode == 4) + { + /* + * Set Sound Characteristics + */ + + opl3_command(map->ioaddr, AM_VIB + map->op[2], instr->operators[OFFS_4OP + 0]); + opl3_command(map->ioaddr, AM_VIB + map->op[3], instr->operators[OFFS_4OP + 1]); + + /* + * Set Attack/Decay + */ + + opl3_command(map->ioaddr, ATTACK_DECAY + map->op[2], instr->operators[OFFS_4OP + 4]); + opl3_command(map->ioaddr, ATTACK_DECAY + map->op[3], instr->operators[OFFS_4OP + 5]); + + /* + * Set Sustain/Release + */ + + opl3_command(map->ioaddr, SUSTAIN_RELEASE + map->op[2], instr->operators[OFFS_4OP + 6]); + opl3_command(map->ioaddr, SUSTAIN_RELEASE + map->op[3], instr->operators[OFFS_4OP + 7]); + + /* + * Set Wave Select + */ + + opl3_command(map->ioaddr, WAVE_SELECT + map->op[2], instr->operators[OFFS_4OP + 8]); + opl3_command(map->ioaddr, WAVE_SELECT + map->op[3], instr->operators[OFFS_4OP + 9]); + + /* + * Set Feedback/Connection + */ + + fpc = instr->operators[OFFS_4OP + 10]; + if (!(fpc & 0x30)) + fpc |= 0x30; /* + * Ensure that at least one chn is enabled + */ + opl3_command(map->ioaddr, FEEDBACK_CONNECTION + map->voice_num + 3, fpc); + } + + devc->voc[voice].mode = voice_mode; + set_voice_volume(voice, volume, devc->voc[voice].volume); + + freq = devc->voc[voice].orig_freq = note_to_freq(note) / 1000; + + /* + * Since the pitch bender may have been set before playing the note, we + * have to calculate the bending now. + */ + + freq = compute_finetune(devc->voc[voice].orig_freq, devc->voc[voice].bender, devc->voc[voice].bender_range, 0); + devc->voc[voice].current_freq = freq; + + freq_to_fnum(freq, &block, &fnum); + + /* + * Play note + */ + + data = fnum & 0xff; /* + * Least significant bits of fnumber + */ + opl3_command(map->ioaddr, FNUM_LOW + map->voice_num, data); + + data = 0x20 | ((block & 0x7) << 2) | ((fnum >> 8) & 0x3); + devc->voc[voice].keyon_byte = data; + opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num, data); + if (voice_mode == 4) + opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num + 3, data); + + return 0; +} + +static void freq_to_fnum (int freq, int *block, int *fnum) +{ + int f, octave; + + /* + * Converts the note frequency to block and fnum values for the FM chip + */ + /* + * First try to compute the block -value (octave) where the note belongs + */ + + f = freq; + + octave = 5; + + if (f == 0) + octave = 0; + else if (f < 261) + { + while (f < 261) + { + octave--; + f <<= 1; + } + } + else if (f > 493) + { + while (f > 493) + { + octave++; + f >>= 1; + } + } + + if (octave > 7) + octave = 7; + + *fnum = freq * (1 << (20 - octave)) / 49716; + *block = octave; +} + +static void opl3_command (int io_addr, unsigned int addr, unsigned int val) +{ + int i; + + /* + * The original 2-OP synth requires a quite long delay after writing to a + * register. The OPL-3 survives with just two INBs + */ + + outb(((unsigned char) (addr & 0xff)), io_addr); + + if (devc->model != 2) + udelay(10); + else + for (i = 0; i < 2; i++) + inb(io_addr); + + outb(((unsigned char) (val & 0xff)), io_addr + 1); + + if (devc->model != 2) + udelay(30); + else + for (i = 0; i < 2; i++) + inb(io_addr); +} + +static void opl3_reset(int devno) +{ + int i; + + for (i = 0; i < 18; i++) + devc->lv_map[i] = i; + + for (i = 0; i < devc->nr_voice; i++) + { + opl3_command(pv_map[devc->lv_map[i]].ioaddr, + KSL_LEVEL + pv_map[devc->lv_map[i]].op[0], 0xff); + + opl3_command(pv_map[devc->lv_map[i]].ioaddr, + KSL_LEVEL + pv_map[devc->lv_map[i]].op[1], 0xff); + + if (pv_map[devc->lv_map[i]].voice_mode == 4) + { + opl3_command(pv_map[devc->lv_map[i]].ioaddr, + KSL_LEVEL + pv_map[devc->lv_map[i]].op[2], 0xff); + + opl3_command(pv_map[devc->lv_map[i]].ioaddr, + KSL_LEVEL + pv_map[devc->lv_map[i]].op[3], 0xff); + } + + opl3_kill_note(devno, i, 0, 64); + } + + if (devc->model == 2) + { + devc->v_alloc->max_voice = devc->nr_voice = 18; + + for (i = 0; i < 18; i++) + pv_map[i].voice_mode = 2; + + } +} + +static int opl3_open(int dev, int mode) +{ + int i; + + if (devc->busy) + return -EBUSY; + devc->busy = 1; + + devc->v_alloc->max_voice = devc->nr_voice = (devc->model == 2) ? 18 : 9; + devc->v_alloc->timestamp = 0; + + for (i = 0; i < 18; i++) + { + devc->v_alloc->map[i] = 0; + devc->v_alloc->alloc_times[i] = 0; + } + + devc->cmask = 0x00; /* + * Just 2 OP mode + */ + if (devc->model == 2) + opl3_command(devc->right_io, CONNECTION_SELECT_REGISTER, devc->cmask); + return 0; +} + +static void opl3_close(int dev) +{ + devc->busy = 0; + devc->v_alloc->max_voice = devc->nr_voice = (devc->model == 2) ? 18 : 9; + + devc->fm_info.nr_drums = 0; + devc->fm_info.perc_mode = 0; + + opl3_reset(dev); +} + +static void opl3_hw_control(int dev, unsigned char *event) +{ +} + +static int opl3_load_patch(int dev, int format, const char __user *addr, + int count, int pmgr_flag) +{ + struct sbi_instrument ins; + + if (count <sizeof(ins)) + { + printk(KERN_WARNING "FM Error: Patch record too short\n"); + return -EINVAL; + } + + if (copy_from_user(&ins, addr, sizeof(ins))) + return -EFAULT; + + if (ins.channel < 0 || ins.channel >= SBFM_MAXINSTR) + { + printk(KERN_WARNING "FM Error: Invalid instrument number %d\n", ins.channel); + return -EINVAL; + } + ins.key = format; + + return store_instr(ins.channel, &ins); +} + +static void opl3_panning(int dev, int voice, int value) +{ + + if (voice < 0 || voice >= devc->nr_voice) + return; + + devc->voc[voice].panning = value; +} + +static void opl3_volume_method(int dev, int mode) +{ +} + +#define SET_VIBRATO(cell) { \ + tmp = instr->operators[(cell-1)+(((cell-1)/2)*OFFS_4OP)]; \ + if (pressure > 110) \ + tmp |= 0x40; /* Vibrato on */ \ + opl3_command (map->ioaddr, AM_VIB + map->op[cell-1], tmp);} + +static void opl3_aftertouch(int dev, int voice, int pressure) +{ + int tmp; + struct sbi_instrument *instr; + struct physical_voice_info *map; + + if (voice < 0 || voice >= devc->nr_voice) + return; + + map = &pv_map[devc->lv_map[voice]]; + + DEB(printk("Aftertouch %d\n", voice)); + + if (map->voice_mode == 0) + return; + + /* + * Adjust the amount of vibrato depending the pressure + */ + + instr = devc->act_i[voice]; + + if (!instr) + instr = &devc->i_map[0]; + + if (devc->voc[voice].mode == 4) + { + int connection = ((instr->operators[10] & 0x01) << 1) | (instr->operators[10 + OFFS_4OP] & 0x01); + + switch (connection) + { + case 0: + SET_VIBRATO(4); + break; + + case 1: + SET_VIBRATO(2); + SET_VIBRATO(4); + break; + + case 2: + SET_VIBRATO(1); + SET_VIBRATO(4); + break; + + case 3: + SET_VIBRATO(1); + SET_VIBRATO(3); + SET_VIBRATO(4); + break; + + } + /* + * Not implemented yet + */ + } + else + { + SET_VIBRATO(1); + + if ((instr->operators[10] & 0x01)) /* + * Additive synthesis + */ + SET_VIBRATO(2); + } +} + +#undef SET_VIBRATO + +static void bend_pitch(int dev, int voice, int value) +{ + unsigned char data; + int block, fnum, freq; + struct physical_voice_info *map; + + map = &pv_map[devc->lv_map[voice]]; + + if (map->voice_mode == 0) + return; + + devc->voc[voice].bender = value; + if (!value) + return; + if (!(devc->voc[voice].keyon_byte & 0x20)) + return; /* + * Not keyed on + */ + + freq = compute_finetune(devc->voc[voice].orig_freq, devc->voc[voice].bender, devc->voc[voice].bender_range, 0); + devc->voc[voice].current_freq = freq; + + freq_to_fnum(freq, &block, &fnum); + + data = fnum & 0xff; /* + * Least significant bits of fnumber + */ + opl3_command(map->ioaddr, FNUM_LOW + map->voice_num, data); + + data = 0x20 | ((block & 0x7) << 2) | ((fnum >> 8) & 0x3); + devc->voc[voice].keyon_byte = data; + opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num, data); +} + +static void opl3_controller (int dev, int voice, int ctrl_num, int value) +{ + if (voice < 0 || voice >= devc->nr_voice) + return; + + switch (ctrl_num) + { + case CTRL_PITCH_BENDER: + bend_pitch(dev, voice, value); + break; + + case CTRL_PITCH_BENDER_RANGE: + devc->voc[voice].bender_range = value; + break; + + case CTL_MAIN_VOLUME: + devc->voc[voice].volume = value / 128; + break; + + case CTL_PAN: + devc->voc[voice].panning = (value * 2) - 128; + break; + } +} + +static void opl3_bender(int dev, int voice, int value) +{ + if (voice < 0 || voice >= devc->nr_voice) + return; + + bend_pitch(dev, voice, value - 8192); +} + +static int opl3_alloc_voice(int dev, int chn, int note, struct voice_alloc_info *alloc) +{ + int i, p, best, first, avail, best_time = 0x7fffffff; + struct sbi_instrument *instr; + int is4op; + int instr_no; + + if (chn < 0 || chn > 15) + instr_no = 0; + else + instr_no = devc->chn_info[chn].pgm_num; + + instr = &devc->i_map[instr_no]; + if (instr->channel < 0 || /* Instrument not loaded */ + devc->nr_voice != 12) /* Not in 4 OP mode */ + is4op = 0; + else if (devc->nr_voice == 12) /* 4 OP mode */ + is4op = (instr->key == OPL3_PATCH); + else + is4op = 0; + + if (is4op) + { + first = p = 0; + avail = 6; + } + else + { + if (devc->nr_voice == 12) /* 4 OP mode. Use the '2 OP only' operators first */ + first = p = 6; + else + first = p = 0; + avail = devc->nr_voice; + } + + /* + * Now try to find a free voice + */ + best = first; + + for (i = 0; i < avail; i++) + { + if (alloc->map[p] == 0) + { + return p; + } + if (alloc->alloc_times[p] < best_time) /* Find oldest playing note */ + { + best_time = alloc->alloc_times[p]; + best = p; + } + p = (p + 1) % avail; + } + + /* + * Insert some kind of priority mechanism here. + */ + + if (best < 0) + best = 0; + if (best > devc->nr_voice) + best -= devc->nr_voice; + + return best; /* All devc->voc in use. Select the first one. */ +} + +static void opl3_setup_voice(int dev, int voice, int chn) +{ + struct channel_info *info; + + if (voice < 0 || voice >= devc->nr_voice) + return; + + if (chn < 0 || chn > 15) + return; + + info = &synth_devs[dev]->chn_info[chn]; + + opl3_set_instr(dev, voice, info->pgm_num); + + devc->voc[voice].bender = 0; + devc->voc[voice].bender_range = info->bender_range; + devc->voc[voice].volume = info->controllers[CTL_MAIN_VOLUME]; + devc->voc[voice].panning = (info->controllers[CTL_PAN] * 2) - 128; +} + +static struct synth_operations opl3_operations = +{ + .owner = THIS_MODULE, + .id = "OPL", + .info = NULL, + .midi_dev = 0, + .synth_type = SYNTH_TYPE_FM, + .synth_subtype = FM_TYPE_ADLIB, + .open = opl3_open, + .close = opl3_close, + .ioctl = opl3_ioctl, + .kill_note = opl3_kill_note, + .start_note = opl3_start_note, + .set_instr = opl3_set_instr, + .reset = opl3_reset, + .hw_control = opl3_hw_control, + .load_patch = opl3_load_patch, + .aftertouch = opl3_aftertouch, + .controller = opl3_controller, + .panning = opl3_panning, + .volume_method = opl3_volume_method, + .bender = opl3_bender, + .alloc_voice = opl3_alloc_voice, + .setup_voice = opl3_setup_voice +}; + +static int opl3_init(int ioaddr, struct module *owner) +{ + int i; + int me; + + if (devc == NULL) + { + printk(KERN_ERR "opl3: Device control structure not initialized.\n"); + return -1; + } + + if ((me = sound_alloc_synthdev()) == -1) + { + printk(KERN_WARNING "opl3: Too many synthesizers\n"); + return -1; + } + + devc->nr_voice = 9; + + devc->fm_info.device = 0; + devc->fm_info.synth_type = SYNTH_TYPE_FM; + devc->fm_info.synth_subtype = FM_TYPE_ADLIB; + devc->fm_info.perc_mode = 0; + devc->fm_info.nr_voices = 9; + devc->fm_info.nr_drums = 0; + devc->fm_info.instr_bank_size = SBFM_MAXINSTR; + devc->fm_info.capabilities = 0; + devc->left_io = ioaddr; + devc->right_io = ioaddr + 2; + + if (detected_model <= 2) + devc->model = 1; + else + { + devc->model = 2; + if (detected_model == 4) + devc->is_opl4 = 1; + } + + opl3_operations.info = &devc->fm_info; + + synth_devs[me] = &opl3_operations; + + if (owner) + synth_devs[me]->owner = owner; + + sequencer_init(); + devc->v_alloc = &opl3_operations.alloc; + devc->chn_info = &opl3_operations.chn_info[0]; + + if (devc->model == 2) + { + if (devc->is_opl4) + strcpy(devc->fm_info.name, "Yamaha OPL4/OPL3 FM"); + else + strcpy(devc->fm_info.name, "Yamaha OPL3"); + + devc->v_alloc->max_voice = devc->nr_voice = 18; + devc->fm_info.nr_drums = 0; + devc->fm_info.synth_subtype = FM_TYPE_OPL3; + devc->fm_info.capabilities |= SYNTH_CAP_OPL3; + + for (i = 0; i < 18; i++) + { + if (pv_map[i].ioaddr == USE_LEFT) + pv_map[i].ioaddr = devc->left_io; + else + pv_map[i].ioaddr = devc->right_io; + } + opl3_command(devc->right_io, OPL3_MODE_REGISTER, OPL3_ENABLE); + opl3_command(devc->right_io, CONNECTION_SELECT_REGISTER, 0x00); + } + else + { + strcpy(devc->fm_info.name, "Yamaha OPL2"); + devc->v_alloc->max_voice = devc->nr_voice = 9; + devc->fm_info.nr_drums = 0; + + for (i = 0; i < 18; i++) + pv_map[i].ioaddr = devc->left_io; + }; + conf_printf2(devc->fm_info.name, ioaddr, 0, -1, -1); + + for (i = 0; i < SBFM_MAXINSTR; i++) + devc->i_map[i].channel = -1; + + return me; +} + +static int me; + +static int io = -1; + +module_param(io, int, 0); + +static int __init init_opl3 (void) +{ + printk(KERN_INFO "YM3812 and OPL-3 driver Copyright (C) by Hannu Savolainen, Rob Hooft 1993-1996\n"); + + if (io != -1) /* User loading pure OPL3 module */ + { + if (!opl3_detect(io)) + { + return -ENODEV; + } + + me = opl3_init(io, THIS_MODULE); + } + + return 0; +} + +static void __exit cleanup_opl3(void) +{ + if (devc && io != -1) + { + if (devc->base) { + release_region(devc->base,4); + if (devc->is_opl4) + release_region(devc->base - 8, 2); + } + kfree(devc); + devc = NULL; + sound_unload_synthdev(me); + } +} + +module_init(init_opl3); +module_exit(cleanup_opl3); + +#ifndef MODULE +static int __init setup_opl3(char *str) +{ + /* io */ + int ints[2]; + + str = get_options(str, ARRAY_SIZE(ints), ints); + + io = ints[1]; + + return 1; +} + +__setup("opl3=", setup_opl3); +#endif +MODULE_LICENSE("GPL"); diff --git a/sound/oss/opl3_hw.h b/sound/oss/opl3_hw.h new file mode 100644 index 00000000..8b11c893 --- /dev/null +++ b/sound/oss/opl3_hw.h @@ -0,0 +1,246 @@ +/* + * opl3_hw.h - Definitions of the OPL-3 registers + * + * + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * + * + * The OPL-3 mode is switched on by writing 0x01, to the offset 5 + * of the right side. + * + * Another special register at the right side is at offset 4. It contains + * a bit mask defining which voices are used as 4 OP voices. + * + * The percussive mode is implemented in the left side only. + * + * With the above exceptions the both sides can be operated independently. + * + * A 4 OP voice can be created by setting the corresponding + * bit at offset 4 of the right side. + * + * For example setting the rightmost bit (0x01) changes the + * first voice on the right side to the 4 OP mode. The fourth + * voice is made inaccessible. + * + * If a voice is set to the 2 OP mode, it works like 2 OP modes + * of the original YM3812 (AdLib). In addition the voice can + * be connected the left, right or both stereo channels. It can + * even be left unconnected. This works with 4 OP voices also. + * + * The stereo connection bits are located in the FEEDBACK_CONNECTION + * register of the voice (0xC0-0xC8). In 4 OP voices these bits are + * in the second half of the voice. + */ + +/* + * Register numbers for the global registers + */ + +#define TEST_REGISTER 0x01 +#define ENABLE_WAVE_SELECT 0x20 + +#define TIMER1_REGISTER 0x02 +#define TIMER2_REGISTER 0x03 +#define TIMER_CONTROL_REGISTER 0x04 /* Left side */ +#define IRQ_RESET 0x80 +#define TIMER1_MASK 0x40 +#define TIMER2_MASK 0x20 +#define TIMER1_START 0x01 +#define TIMER2_START 0x02 + +#define CONNECTION_SELECT_REGISTER 0x04 /* Right side */ +#define RIGHT_4OP_0 0x01 +#define RIGHT_4OP_1 0x02 +#define RIGHT_4OP_2 0x04 +#define LEFT_4OP_0 0x08 +#define LEFT_4OP_1 0x10 +#define LEFT_4OP_2 0x20 + +#define OPL3_MODE_REGISTER 0x05 /* Right side */ +#define OPL3_ENABLE 0x01 +#define OPL4_ENABLE 0x02 + +#define KBD_SPLIT_REGISTER 0x08 /* Left side */ +#define COMPOSITE_SINE_WAVE_MODE 0x80 /* Don't use with OPL-3? */ +#define KEYBOARD_SPLIT 0x40 + +#define PERCOSSION_REGISTER 0xbd /* Left side only */ +#define TREMOLO_DEPTH 0x80 +#define VIBRATO_DEPTH 0x40 +#define PERCOSSION_ENABLE 0x20 +#define BASSDRUM_ON 0x10 +#define SNAREDRUM_ON 0x08 +#define TOMTOM_ON 0x04 +#define CYMBAL_ON 0x02 +#define HIHAT_ON 0x01 + +/* + * Offsets to the register banks for operators. To get the + * register number just add the operator offset to the bank offset + * + * AM/VIB/EG/KSR/Multiple (0x20 to 0x35) + */ +#define AM_VIB 0x20 +#define TREMOLO_ON 0x80 +#define VIBRATO_ON 0x40 +#define SUSTAIN_ON 0x20 +#define KSR 0x10 /* Key scaling rate */ +#define MULTIPLE_MASK 0x0f /* Frequency multiplier */ + + /* + * KSL/Total level (0x40 to 0x55) + */ +#define KSL_LEVEL 0x40 +#define KSL_MASK 0xc0 /* Envelope scaling bits */ +#define TOTAL_LEVEL_MASK 0x3f /* Strength (volume) of OP */ + +/* + * Attack / Decay rate (0x60 to 0x75) + */ +#define ATTACK_DECAY 0x60 +#define ATTACK_MASK 0xf0 +#define DECAY_MASK 0x0f + +/* + * Sustain level / Release rate (0x80 to 0x95) + */ +#define SUSTAIN_RELEASE 0x80 +#define SUSTAIN_MASK 0xf0 +#define RELEASE_MASK 0x0f + +/* + * Wave select (0xE0 to 0xF5) + */ +#define WAVE_SELECT 0xe0 + +/* + * Offsets to the register banks for voices. Just add to the + * voice number to get the register number. + * + * F-Number low bits (0xA0 to 0xA8). + */ +#define FNUM_LOW 0xa0 + +/* + * F-number high bits / Key on / Block (octave) (0xB0 to 0xB8) + */ +#define KEYON_BLOCK 0xb0 +#define KEYON_BIT 0x20 +#define BLOCKNUM_MASK 0x1c +#define FNUM_HIGH_MASK 0x03 + +/* + * Feedback / Connection (0xc0 to 0xc8) + * + * These registers have two new bits when the OPL-3 mode + * is selected. These bits controls connecting the voice + * to the stereo channels. For 4 OP voices this bit is + * defined in the second half of the voice (add 3 to the + * register offset). + * + * For 4 OP voices the connection bit is used in the + * both halves (gives 4 ways to connect the operators). + */ +#define FEEDBACK_CONNECTION 0xc0 +#define FEEDBACK_MASK 0x0e /* Valid just for 1st OP of a voice */ +#define CONNECTION_BIT 0x01 +/* + * In the 4 OP mode there is four possible configurations how the + * operators can be connected together (in 2 OP modes there is just + * AM or FM). The 4 OP connection mode is defined by the rightmost + * bit of the FEEDBACK_CONNECTION (0xC0-0xC8) on the both halves. + * + * First half Second half Mode + * + * +---+ + * v | + * 0 0 >+-1-+--2--3--4--> + * + * + * + * +---+ + * | | + * 0 1 >+-1-+--2-+ + * |-> + * >--3----4-+ + * + * +---+ + * | | + * 1 0 >+-1-+-----+ + * |-> + * >--2--3--4-+ + * + * +---+ + * | | + * 1 1 >+-1-+--+ + * | + * >--2--3-+-> + * | + * >--4----+ + */ +#define STEREO_BITS 0x30 /* OPL-3 only */ +#define VOICE_TO_LEFT 0x10 +#define VOICE_TO_RIGHT 0x20 + +/* + * Definition table for the physical voices + */ + +struct physical_voice_info { + unsigned char voice_num; + unsigned char voice_mode; /* 0=unavailable, 2=2 OP, 4=4 OP */ + unsigned short ioaddr; /* I/O port (left or right side) */ + unsigned char op[4]; /* Operator offsets */ + }; + +/* + * There is 18 possible 2 OP voices + * (9 in the left and 9 in the right). + * The first OP is the modulator and 2nd is the carrier. + * + * The first three voices in the both sides may be connected + * with another voice to a 4 OP voice. For example voice 0 + * can be connected with voice 3. The operators of voice 3 are + * used as operators 3 and 4 of the new 4 OP voice. + * In this case the 2 OP voice number 0 is the 'first half' and + * voice 3 is the second. + */ + +#define USE_LEFT 0 +#define USE_RIGHT 1 + +static struct physical_voice_info pv_map[18] = +{ +/* No Mode Side OP1 OP2 OP3 OP4 */ +/* --------------------------------------------------- */ + { 0, 2, USE_LEFT, {0x00, 0x03, 0x08, 0x0b}}, + { 1, 2, USE_LEFT, {0x01, 0x04, 0x09, 0x0c}}, + { 2, 2, USE_LEFT, {0x02, 0x05, 0x0a, 0x0d}}, + + { 3, 2, USE_LEFT, {0x08, 0x0b, 0x00, 0x00}}, + { 4, 2, USE_LEFT, {0x09, 0x0c, 0x00, 0x00}}, + { 5, 2, USE_LEFT, {0x0a, 0x0d, 0x00, 0x00}}, + + { 6, 2, USE_LEFT, {0x10, 0x13, 0x00, 0x00}}, /* Used by percussive voices */ + { 7, 2, USE_LEFT, {0x11, 0x14, 0x00, 0x00}}, /* if the percussive mode */ + { 8, 2, USE_LEFT, {0x12, 0x15, 0x00, 0x00}}, /* is selected */ + + { 0, 2, USE_RIGHT, {0x00, 0x03, 0x08, 0x0b}}, + { 1, 2, USE_RIGHT, {0x01, 0x04, 0x09, 0x0c}}, + { 2, 2, USE_RIGHT, {0x02, 0x05, 0x0a, 0x0d}}, + + { 3, 2, USE_RIGHT, {0x08, 0x0b, 0x00, 0x00}}, + { 4, 2, USE_RIGHT, {0x09, 0x0c, 0x00, 0x00}}, + { 5, 2, USE_RIGHT, {0x0a, 0x0d, 0x00, 0x00}}, + + { 6, 2, USE_RIGHT, {0x10, 0x13, 0x00, 0x00}}, + { 7, 2, USE_RIGHT, {0x11, 0x14, 0x00, 0x00}}, + { 8, 2, USE_RIGHT, {0x12, 0x15, 0x00, 0x00}} +}; +/* + * DMA buffer calls + */ diff --git a/sound/oss/os.h b/sound/oss/os.h new file mode 100644 index 00000000..a1a962d7 --- /dev/null +++ b/sound/oss/os.h @@ -0,0 +1,46 @@ +#define ALLOW_SELECT +#undef NO_INLINE_ASM +#define SHORT_BANNERS +#define MANUAL_PNP +#undef DO_TIMINGS + +#include <linux/module.h> + +#ifdef __KERNEL__ +#include <linux/string.h> +#include <linux/fs.h> +#include <asm/dma.h> +#include <asm/io.h> +#include <asm/param.h> +#include <linux/sched.h> +#include <linux/slab.h> +#include <linux/ioport.h> +#include <asm/page.h> +#include <asm/system.h> +#include <linux/vmalloc.h> +#include <asm/uaccess.h> +#include <linux/poll.h> +#include <linux/pci.h> +#endif + +#include <linux/soundcard.h> + +#define FALSE 0 +#define TRUE 1 + +extern int sound_alloc_dma(int chn, char *deviceID); +extern int sound_open_dma(int chn, char *deviceID); +extern void sound_free_dma(int chn); +extern void sound_close_dma(int chn); + +extern void reprogram_timer(void); + +#define USE_AUTOINIT_DMA + +extern void *sound_mem_blocks[1024]; +extern int sound_nblocks; + +#undef PSEUDO_DMA_AUTOINIT +#define ALLOW_BUFFER_MAPPING + +extern const struct file_operations oss_sound_fops; diff --git a/sound/oss/pas2.h b/sound/oss/pas2.h new file mode 100644 index 00000000..fa12c55f --- /dev/null +++ b/sound/oss/pas2.h @@ -0,0 +1,17 @@ + +/* From pas_card.c */ +int pas_set_intr(int mask); +int pas_remove_intr(int mask); +unsigned char pas_read(int ioaddr); +void pas_write(unsigned char data, int ioaddr); + +/* From pas_audio.c */ +void pas_pcm_interrupt(unsigned char status, int cause); +void pas_pcm_init(struct address_info *hw_config); + +/* From pas_mixer.c */ +int pas_init_mixer(void); + +/* From pas_midi.c */ +void pas_midi_init(void); +void pas_midi_interrupt(void); diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c new file mode 100644 index 00000000..7f377ec3 --- /dev/null +++ b/sound/oss/pas2_card.c @@ -0,0 +1,455 @@ +/* + * sound/oss/pas2_card.c + * + * Detection routine for the Pro Audio Spectrum cards. + */ + +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/spinlock.h> +#include "sound_config.h" + +#include "pas2.h" +#include "sb.h" + +static unsigned char dma_bits[] = { + 4, 1, 2, 3, 0, 5, 6, 7 +}; + +static unsigned char irq_bits[] = { + 0, 0, 1, 2, 3, 4, 5, 6, 0, 1, 7, 8, 9, 0, 10, 11 +}; + +static unsigned char sb_irq_bits[] = { + 0x00, 0x00, 0x08, 0x10, 0x00, 0x18, 0x00, 0x20, + 0x00, 0x08, 0x28, 0x30, 0x38, 0, 0 +}; + +static unsigned char sb_dma_bits[] = { + 0x00, 0x40, 0x80, 0xC0, 0, 0, 0, 0 +}; + +/* + * The Address Translation code is used to convert I/O register addresses to + * be relative to the given base -register + */ + +int pas_translate_code = 0; +static int pas_intr_mask; +static int pas_irq; +static int pas_sb_base; +DEFINE_SPINLOCK(pas_lock); +#ifndef CONFIG_PAS_JOYSTICK +static int joystick; +#else +static int joystick = 1; +#endif +#ifdef SYMPHONY_PAS +static int symphony = 1; +#else +static int symphony; +#endif +#ifdef BROKEN_BUS_CLOCK +static int broken_bus_clock = 1; +#else +static int broken_bus_clock; +#endif + +static struct address_info cfg; +static struct address_info cfg2; + +char pas_model = 0; +static char *pas_model_names[] = { + "", + "Pro AudioSpectrum+", + "CDPC", + "Pro AudioSpectrum 16", + "Pro AudioSpectrum 16D" +}; + +/* + * pas_read() and pas_write() are equivalents of inb and outb + * These routines perform the I/O address translation required + * to support other than the default base address + */ + +extern void mix_write(unsigned char data, int ioaddr); + +unsigned char pas_read(int ioaddr) +{ + return inb(ioaddr + pas_translate_code); +} + +void pas_write(unsigned char data, int ioaddr) +{ + outb((data), ioaddr + pas_translate_code); +} + +/******************* Begin of the Interrupt Handler ********************/ + +static irqreturn_t pasintr(int irq, void *dev_id) +{ + int status; + + status = pas_read(0x0B89); + pas_write(status, 0x0B89); /* Clear interrupt */ + + if (status & 0x08) + { + pas_pcm_interrupt(status, 1); + status &= ~0x08; + } + if (status & 0x10) + { + pas_midi_interrupt(); + status &= ~0x10; + } + return IRQ_HANDLED; +} + +int pas_set_intr(int mask) +{ + if (!mask) + return 0; + + pas_intr_mask |= mask; + + pas_write(pas_intr_mask, 0x0B8B); + return 0; +} + +int pas_remove_intr(int mask) +{ + if (!mask) + return 0; + + pas_intr_mask &= ~mask; + pas_write(pas_intr_mask, 0x0B8B); + + return 0; +} + +/******************* End of the Interrupt handler **********************/ + +/******************* Begin of the Initialization Code ******************/ + +static int __init config_pas_hw(struct address_info *hw_config) +{ + char ok = 1; + unsigned int_ptrs; /* scsi/sound interrupt pointers */ + + pas_irq = hw_config->irq; + + pas_write(0x00, 0x0B8B); + pas_write(0x36, 0x138B); + pas_write(0x36, 0x1388); + pas_write(0, 0x1388); + pas_write(0x74, 0x138B); + pas_write(0x74, 0x1389); + pas_write(0, 0x1389); + + pas_write(0x80 | 0x40 | 0x20 | 1, 0x0B8A); + pas_write(0x80 | 0x20 | 0x10 | 0x08 | 0x01, 0xF8A); + pas_write(0x01 | 0x02 | 0x04 | 0x10 /* + * | + * 0x80 + */ , 0xB88); + + pas_write(0x80 | (joystick ? 0x40 : 0), 0xF388); + + if (pas_irq < 0 || pas_irq > 15) + { + printk(KERN_ERR "PAS16: Invalid IRQ %d", pas_irq); + hw_config->irq=-1; + ok = 0; + } + else + { + int_ptrs = pas_read(0xF38A); + int_ptrs = (int_ptrs & 0xf0) | irq_bits[pas_irq]; + pas_write(int_ptrs, 0xF38A); + if (!irq_bits[pas_irq]) + { + printk(KERN_ERR "PAS16: Invalid IRQ %d", pas_irq); + hw_config->irq=-1; + ok = 0; + } + else + { + if (request_irq(pas_irq, pasintr, 0, "PAS16",hw_config) < 0) { + printk(KERN_ERR "PAS16: Cannot allocate IRQ %d\n",pas_irq); + hw_config->irq=-1; + ok = 0; + } + } + } + + if (hw_config->dma < 0 || hw_config->dma > 7) + { + printk(KERN_ERR "PAS16: Invalid DMA selection %d", hw_config->dma); + hw_config->dma=-1; + ok = 0; + } + else + { + pas_write(dma_bits[hw_config->dma], 0xF389); + if (!dma_bits[hw_config->dma]) + { + printk(KERN_ERR "PAS16: Invalid DMA selection %d", hw_config->dma); + hw_config->dma=-1; + ok = 0; + } + else + { + if (sound_alloc_dma(hw_config->dma, "PAS16")) + { + printk(KERN_ERR "pas2_card.c: Can't allocate DMA channel\n"); + hw_config->dma=-1; + ok = 0; + } + } + } + + /* + * This fixes the timing problems of the PAS due to the Symphony chipset + * as per Media Vision. Only define this if your PAS doesn't work correctly. + */ + + if(symphony) + { + outb((0x05), 0xa8); + outb((0x60), 0xa9); + } + + if(broken_bus_clock) + pas_write(0x01 | 0x10 | 0x20 | 0x04, 0x8388); + else + /* + * pas_write(0x01, 0x8388); + */ + pas_write(0x01 | 0x10 | 0x20, 0x8388); + + pas_write(0x18, 0x838A); /* ??? */ + pas_write(0x20 | 0x01, 0x0B8A); /* Mute off, filter = 17.897 kHz */ + pas_write(8, 0xBF8A); + + mix_write(0x80 | 5, 0x078B); + mix_write(5, 0x078B); + + { + struct address_info *sb_config; + + sb_config = &cfg2; + if (sb_config->io_base) + { + unsigned char irq_dma; + + /* + * Turn on Sound Blaster compatibility + * bit 1 = SB emulation + * bit 0 = MPU401 emulation (CDPC only :-( ) + */ + + pas_write(0x02, 0xF788); + + /* + * "Emulation address" + */ + + pas_write((sb_config->io_base >> 4) & 0x0f, 0xF789); + pas_sb_base = sb_config->io_base; + + if (!sb_dma_bits[sb_config->dma]) + printk(KERN_ERR "PAS16 Warning: Invalid SB DMA %d\n\n", sb_config->dma); + + if (!sb_irq_bits[sb_config->irq]) + printk(KERN_ERR "PAS16 Warning: Invalid SB IRQ %d\n\n", sb_config->irq); + + irq_dma = sb_dma_bits[sb_config->dma] | + sb_irq_bits[sb_config->irq]; + + pas_write(irq_dma, 0xFB8A); + } + else + pas_write(0x00, 0xF788); + } + + if (!ok) + printk(KERN_WARNING "PAS16: Driver not enabled\n"); + + return ok; +} + +static int __init detect_pas_hw(struct address_info *hw_config) +{ + unsigned char board_id, foo; + + /* + * WARNING: Setting an option like W:1 or so that disables warm boot reset + * of the card will screw up this detect code something fierce. Adding code + * to handle this means possibly interfering with other cards on the bus if + * you have something on base port 0x388. SO be forewarned. + */ + + outb((0xBC), 0x9A01); /* Activate first board */ + outb((hw_config->io_base >> 2), 0x9A01); /* Set base address */ + pas_translate_code = hw_config->io_base - 0x388; + pas_write(1, 0xBF88); /* Select one wait states */ + + board_id = pas_read(0x0B8B); + + if (board_id == 0xff) + return 0; + + /* + * We probably have a PAS-series board, now check for a PAS16-series board + * by trying to change the board revision bits. PAS16-series hardware won't + * let you do this - the bits are read-only. + */ + + foo = board_id ^ 0xe0; + + pas_write(foo, 0x0B8B); + foo = pas_read(0x0B8B); + pas_write(board_id, 0x0B8B); + + if (board_id != foo) + return 0; + + pas_model = pas_read(0xFF88); + + return pas_model; +} + +static void __init attach_pas_card(struct address_info *hw_config) +{ + pas_irq = hw_config->irq; + + if (detect_pas_hw(hw_config)) + { + + if ((pas_model = pas_read(0xFF88))) + { + char temp[100]; + + sprintf(temp, + "%s rev %d", pas_model_names[(int) pas_model], + pas_read(0x2789)); + conf_printf(temp, hw_config); + } + if (config_pas_hw(hw_config)) + { + pas_pcm_init(hw_config); + pas_midi_init(); + pas_init_mixer(); + } + } +} + +static inline int __init probe_pas(struct address_info *hw_config) +{ + return detect_pas_hw(hw_config); +} + +static void __exit unload_pas(struct address_info *hw_config) +{ + extern int pas_audiodev; + extern int pas2_mididev; + + if (hw_config->dma>0) + sound_free_dma(hw_config->dma); + if (hw_config->irq>0) + free_irq(hw_config->irq, hw_config); + + if(pas_audiodev!=-1) + sound_unload_mixerdev(audio_devs[pas_audiodev]->mixer_dev); + if(pas2_mididev!=-1) + sound_unload_mididev(pas2_mididev); + if(pas_audiodev!=-1) + sound_unload_audiodev(pas_audiodev); +} + +static int __initdata io = -1; +static int __initdata irq = -1; +static int __initdata dma = -1; +static int __initdata dma16 = -1; /* Set this for modules that need it */ + +static int __initdata sb_io = 0; +static int __initdata sb_irq = -1; +static int __initdata sb_dma = -1; +static int __initdata sb_dma16 = -1; + +module_param(io, int, 0); +module_param(irq, int, 0); +module_param(dma, int, 0); +module_param(dma16, int, 0); + +module_param(sb_io, int, 0); +module_param(sb_irq, int, 0); +module_param(sb_dma, int, 0); +module_param(sb_dma16, int, 0); + +module_param(joystick, bool, 0); +module_param(symphony, bool, 0); +module_param(broken_bus_clock, bool, 0); + +MODULE_LICENSE("GPL"); + +static int __init init_pas2(void) +{ + printk(KERN_INFO "Pro Audio Spectrum driver Copyright (C) by Hannu Savolainen 1993-1996\n"); + + cfg.io_base = io; + cfg.irq = irq; + cfg.dma = dma; + cfg.dma2 = dma16; + + cfg2.io_base = sb_io; + cfg2.irq = sb_irq; + cfg2.dma = sb_dma; + cfg2.dma2 = sb_dma16; + + if (cfg.io_base == -1 || cfg.dma == -1 || cfg.irq == -1) { + printk(KERN_INFO "I/O, IRQ, DMA and type are mandatory\n"); + return -EINVAL; + } + + if (!probe_pas(&cfg)) + return -ENODEV; + attach_pas_card(&cfg); + + return 0; +} + +static void __exit cleanup_pas2(void) +{ + unload_pas(&cfg); +} + +module_init(init_pas2); +module_exit(cleanup_pas2); + +#ifndef MODULE +static int __init setup_pas2(char *str) +{ + /* io, irq, dma, dma2, sb_io, sb_irq, sb_dma, sb_dma2 */ + int ints[9]; + + str = get_options(str, ARRAY_SIZE(ints), ints); + + io = ints[1]; + irq = ints[2]; + dma = ints[3]; + dma16 = ints[4]; + + sb_io = ints[5]; + sb_irq = ints[6]; + sb_dma = ints[7]; + sb_dma16 = ints[8]; + + return 1; +} + +__setup("pas2=", setup_pas2); +#endif diff --git a/sound/oss/pas2_midi.c b/sound/oss/pas2_midi.c new file mode 100644 index 00000000..1122d10a --- /dev/null +++ b/sound/oss/pas2_midi.c @@ -0,0 +1,262 @@ +/* + * sound/oss/pas2_midi.c + * + * The low level driver for the PAS Midi Interface. + */ +/* + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * + * Bartlomiej Zolnierkiewicz : Added __init to pas_init_mixer() + */ + +#include <linux/init.h> +#include <linux/spinlock.h> +#include "sound_config.h" + +#include "pas2.h" + +extern spinlock_t pas_lock; + +static int midi_busy, input_opened; +static int my_dev; + +int pas2_mididev=-1; + +static unsigned char tmp_queue[256]; +static volatile int qlen; +static volatile unsigned char qhead, qtail; + +static void (*midi_input_intr) (int dev, unsigned char data); + +static int pas_midi_open(int dev, int mode, + void (*input) (int dev, unsigned char data), + void (*output) (int dev) +) +{ + int err; + unsigned long flags; + unsigned char ctrl; + + + if (midi_busy) + return -EBUSY; + + /* + * Reset input and output FIFO pointers + */ + pas_write(0x20 | 0x40, + 0x178b); + + spin_lock_irqsave(&pas_lock, flags); + + if ((err = pas_set_intr(0x10)) < 0) + { + spin_unlock_irqrestore(&pas_lock, flags); + return err; + } + /* + * Enable input available and output FIFO empty interrupts + */ + + ctrl = 0; + input_opened = 0; + midi_input_intr = input; + + if (mode == OPEN_READ || mode == OPEN_READWRITE) + { + ctrl |= 0x04; /* Enable input */ + input_opened = 1; + } + if (mode == OPEN_WRITE || mode == OPEN_READWRITE) + { + ctrl |= 0x08 | 0x10; /* Enable output */ + } + pas_write(ctrl, 0x178b); + + /* + * Acknowledge any pending interrupts + */ + + pas_write(0xff, 0x1B88); + + spin_unlock_irqrestore(&pas_lock, flags); + + midi_busy = 1; + qlen = qhead = qtail = 0; + return 0; +} + +static void pas_midi_close(int dev) +{ + + /* + * Reset FIFO pointers, disable intrs + */ + pas_write(0x20 | 0x40, 0x178b); + + pas_remove_intr(0x10); + midi_busy = 0; +} + +static int dump_to_midi(unsigned char midi_byte) +{ + int fifo_space, x; + + fifo_space = ((x = pas_read(0x1B89)) >> 4) & 0x0f; + + /* + * The MIDI FIFO space register and it's documentation is nonunderstandable. + * There seem to be no way to differentiate between buffer full and buffer + * empty situations. For this reason we don't never write the buffer + * completely full. In this way we can assume that 0 (or is it 15) + * means that the buffer is empty. + */ + + if (fifo_space < 2 && fifo_space != 0) /* Full (almost) */ + return 0; /* Ask upper layers to retry after some time */ + + pas_write(midi_byte, 0x178A); + + return 1; +} + +static int pas_midi_out(int dev, unsigned char midi_byte) +{ + + unsigned long flags; + + /* + * Drain the local queue first + */ + + spin_lock_irqsave(&pas_lock, flags); + + while (qlen && dump_to_midi(tmp_queue[qhead])) + { + qlen--; + qhead++; + } + + spin_unlock_irqrestore(&pas_lock, flags); + + /* + * Output the byte if the local queue is empty. + */ + + if (!qlen) + if (dump_to_midi(midi_byte)) + return 1; + + /* + * Put to the local queue + */ + + if (qlen >= 256) + return 0; /* Local queue full */ + + spin_lock_irqsave(&pas_lock, flags); + + tmp_queue[qtail] = midi_byte; + qlen++; + qtail++; + + spin_unlock_irqrestore(&pas_lock, flags); + + return 1; +} + +static int pas_midi_start_read(int dev) +{ + return 0; +} + +static int pas_midi_end_read(int dev) +{ + return 0; +} + +static void pas_midi_kick(int dev) +{ +} + +static int pas_buffer_status(int dev) +{ + return qlen; +} + +#define MIDI_SYNTH_NAME "Pro Audio Spectrum Midi" +#define MIDI_SYNTH_CAPS SYNTH_CAP_INPUT +#include "midi_synth.h" + +static struct midi_operations pas_midi_operations = +{ + .owner = THIS_MODULE, + .info = {"Pro Audio Spectrum", 0, 0, SNDCARD_PAS}, + .converter = &std_midi_synth, + .in_info = {0}, + .open = pas_midi_open, + .close = pas_midi_close, + .outputc = pas_midi_out, + .start_read = pas_midi_start_read, + .end_read = pas_midi_end_read, + .kick = pas_midi_kick, + .buffer_status = pas_buffer_status, +}; + +void __init pas_midi_init(void) +{ + int dev = sound_alloc_mididev(); + + if (dev == -1) + { + printk(KERN_WARNING "pas_midi_init: Too many midi devices detected\n"); + return; + } + std_midi_synth.midi_dev = my_dev = dev; + midi_devs[dev] = &pas_midi_operations; + pas2_mididev = dev; + sequencer_init(); +} + +void pas_midi_interrupt(void) +{ + unsigned char stat; + int i, incount; + + stat = pas_read(0x1B88); + + if (stat & 0x04) /* Input data available */ + { + incount = pas_read(0x1B89) & 0x0f; /* Input FIFO size */ + if (!incount) + incount = 16; + + for (i = 0; i < incount; i++) + if (input_opened) + { + midi_input_intr(my_dev, pas_read(0x178A)); + } else + pas_read(0x178A); /* Flush */ + } + if (stat & (0x08 | 0x10)) + { + spin_lock(&pas_lock);/* called in irq context */ + + while (qlen && dump_to_midi(tmp_queue[qhead])) + { + qlen--; + qhead++; + } + + spin_unlock(&pas_lock); + } + if (stat & 0x40) + { + printk(KERN_WARNING "MIDI output overrun %x,%x\n", pas_read(0x1B89), stat); + } + pas_write(stat, 0x1B88); /* Acknowledge interrupts */ +} diff --git a/sound/oss/pas2_mixer.c b/sound/oss/pas2_mixer.c new file mode 100644 index 00000000..a0bcb85c --- /dev/null +++ b/sound/oss/pas2_mixer.c @@ -0,0 +1,336 @@ + +/* + * sound/oss/pas2_mixer.c + * + * Mixer routines for the Pro Audio Spectrum cards. + */ + +/* + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + */ +/* + * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed) + * Bartlomiej Zolnierkiewicz : added __init to pas_init_mixer() + */ +#include <linux/init.h> +#include "sound_config.h" + +#include "pas2.h" + +#ifndef DEB +#define DEB(what) /* (what) */ +#endif + +extern int pas_translate_code; +extern char pas_model; +extern int *pas_osp; +extern int pas_audiodev; + +static int rec_devices = (SOUND_MASK_MIC); /* Default recording source */ +static int mode_control; + +#define POSSIBLE_RECORDING_DEVICES (SOUND_MASK_SYNTH | SOUND_MASK_SPEAKER | SOUND_MASK_LINE | SOUND_MASK_MIC | \ + SOUND_MASK_CD | SOUND_MASK_ALTPCM) + +#define SUPPORTED_MIXER_DEVICES (SOUND_MASK_SYNTH | SOUND_MASK_PCM | SOUND_MASK_SPEAKER | SOUND_MASK_LINE | SOUND_MASK_MIC | \ + SOUND_MASK_CD | SOUND_MASK_ALTPCM | SOUND_MASK_IMIX | \ + SOUND_MASK_VOLUME | SOUND_MASK_BASS | SOUND_MASK_TREBLE | SOUND_MASK_RECLEV) + +static int *levels; + +static int default_levels[32] = +{ + 0x3232, /* Master Volume */ + 0x3232, /* Bass */ + 0x3232, /* Treble */ + 0x5050, /* FM */ + 0x4b4b, /* PCM */ + 0x3232, /* PC Speaker */ + 0x4b4b, /* Ext Line */ + 0x4b4b, /* Mic */ + 0x4b4b, /* CD */ + 0x6464, /* Recording monitor */ + 0x4b4b, /* SB PCM */ + 0x6464 /* Recording level */ +}; + +void +mix_write(unsigned char data, int ioaddr) +{ + /* + * The Revision D cards have a problem with their MVA508 interface. The + * kludge-o-rama fix is to make a 16-bit quantity with identical LSB and + * MSBs out of the output byte and to do a 16-bit out to the mixer port - + * 1. We need to do this because it isn't timing problem but chip access + * sequence problem. + */ + + if (pas_model == 4) + { + outw(data | (data << 8), (ioaddr + pas_translate_code) - 1); + outb((0x80), 0); + } else + pas_write(data, ioaddr); +} + +static int +mixer_output(int right_vol, int left_vol, int div, int bits, + int mixer) /* Input or output mixer */ +{ + int left = left_vol * div / 100; + int right = right_vol * div / 100; + + + if (bits & 0x10) + { + left |= mixer; + right |= mixer; + } + if (bits == 0x03 || bits == 0x04) + { + mix_write(0x80 | bits, 0x078B); + mix_write(left, 0x078B); + right_vol = left_vol; + } else + { + mix_write(0x80 | 0x20 | bits, 0x078B); + mix_write(left, 0x078B); + mix_write(0x80 | 0x40 | bits, 0x078B); + mix_write(right, 0x078B); + } + + return (left_vol | (right_vol << 8)); +} + +static void +set_mode(int new_mode) +{ + mix_write(0x80 | 0x05, 0x078B); + mix_write(new_mode, 0x078B); + + mode_control = new_mode; +} + +static int +pas_mixer_set(int whichDev, unsigned int level) +{ + int left, right, devmask, changed, i, mixer = 0; + + DEB(printk("static int pas_mixer_set(int whichDev = %d, unsigned int level = %X)\n", whichDev, level)); + + left = level & 0x7f; + right = (level & 0x7f00) >> 8; + + if (whichDev < SOUND_MIXER_NRDEVICES) { + if ((1 << whichDev) & rec_devices) + mixer = 0x20; + else + mixer = 0x00; + } + + switch (whichDev) + { + case SOUND_MIXER_VOLUME: /* Master volume (0-63) */ + levels[whichDev] = mixer_output(right, left, 63, 0x01, 0); + break; + + /* + * Note! Bass and Treble are mono devices. Will use just the left + * channel. + */ + case SOUND_MIXER_BASS: /* Bass (0-12) */ + levels[whichDev] = mixer_output(right, left, 12, 0x03, 0); + break; + case SOUND_MIXER_TREBLE: /* Treble (0-12) */ + levels[whichDev] = mixer_output(right, left, 12, 0x04, 0); + break; + + case SOUND_MIXER_SYNTH: /* Internal synthesizer (0-31) */ + levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x00, mixer); + break; + case SOUND_MIXER_PCM: /* PAS PCM (0-31) */ + levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x05, mixer); + break; + case SOUND_MIXER_ALTPCM: /* SB PCM (0-31) */ + levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x07, mixer); + break; + case SOUND_MIXER_SPEAKER: /* PC speaker (0-31) */ + levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x06, mixer); + break; + case SOUND_MIXER_LINE: /* External line (0-31) */ + levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x02, mixer); + break; + case SOUND_MIXER_CD: /* CD (0-31) */ + levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x03, mixer); + break; + case SOUND_MIXER_MIC: /* External microphone (0-31) */ + levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x04, mixer); + break; + case SOUND_MIXER_IMIX: /* Recording monitor (0-31) (Output mixer only) */ + levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x01, + 0x00); + break; + case SOUND_MIXER_RECLEV: /* Recording level (0-15) */ + levels[whichDev] = mixer_output(right, left, 15, 0x02, 0); + break; + + + case SOUND_MIXER_RECSRC: + devmask = level & POSSIBLE_RECORDING_DEVICES; + + changed = devmask ^ rec_devices; + rec_devices = devmask; + + for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) + if (changed & (1 << i)) + { + pas_mixer_set(i, levels[i]); + } + return rec_devices; + break; + + default: + return -EINVAL; + } + + return (levels[whichDev]); +} + +/*****/ + +static void +pas_mixer_reset(void) +{ + int foo; + + DEB(printk("pas2_mixer.c: void pas_mixer_reset(void)\n")); + + for (foo = 0; foo < SOUND_MIXER_NRDEVICES; foo++) + pas_mixer_set(foo, levels[foo]); + + set_mode(0x04 | 0x01); +} + +static int pas_mixer_ioctl(int dev, unsigned int cmd, void __user *arg) +{ + int level,v ; + int __user *p = (int __user *)arg; + + DEB(printk("pas2_mixer.c: int pas_mixer_ioctl(unsigned int cmd = %X, unsigned int arg = %X)\n", cmd, arg)); + if (cmd == SOUND_MIXER_PRIVATE1) { /* Set loudness bit */ + if (get_user(level, p)) + return -EFAULT; + if (level == -1) /* Return current settings */ + level = (mode_control & 0x04); + else { + mode_control &= ~0x04; + if (level) + mode_control |= 0x04; + set_mode(mode_control); + } + level = !!level; + return put_user(level, p); + } + if (cmd == SOUND_MIXER_PRIVATE2) { /* Set enhance bit */ + if (get_user(level, p)) + return -EFAULT; + if (level == -1) { /* Return current settings */ + if (!(mode_control & 0x03)) + level = 0; + else + level = ((mode_control & 0x03) + 1) * 20; + } else { + int i = 0; + + level &= 0x7f; + if (level) + i = (level / 20) - 1; + mode_control &= ~0x03; + mode_control |= i & 0x03; + set_mode(mode_control); + if (i) + i = (i + 1) * 20; + level = i; + } + return put_user(level, p); + } + if (cmd == SOUND_MIXER_PRIVATE3) { /* Set mute bit */ + if (get_user(level, p)) + return -EFAULT; + if (level == -1) /* Return current settings */ + level = !(pas_read(0x0B8A) & 0x20); + else { + if (level) + pas_write(pas_read(0x0B8A) & (~0x20), 0x0B8A); + else + pas_write(pas_read(0x0B8A) | 0x20, 0x0B8A); + + level = !(pas_read(0x0B8A) & 0x20); + } + return put_user(level, p); + } + if (((cmd >> 8) & 0xff) == 'M') { + if (get_user(v, p)) + return -EFAULT; + if (_SIOC_DIR(cmd) & _SIOC_WRITE) { + v = pas_mixer_set(cmd & 0xff, v); + } else { + switch (cmd & 0xff) { + case SOUND_MIXER_RECSRC: + v = rec_devices; + break; + + case SOUND_MIXER_STEREODEVS: + v = SUPPORTED_MIXER_DEVICES & ~(SOUND_MASK_BASS | SOUND_MASK_TREBLE); + break; + + case SOUND_MIXER_DEVMASK: + v = SUPPORTED_MIXER_DEVICES; + break; + + case SOUND_MIXER_RECMASK: + v = POSSIBLE_RECORDING_DEVICES & SUPPORTED_MIXER_DEVICES; + break; + + case SOUND_MIXER_CAPS: + v = 0; /* No special capabilities */ + break; + + default: + v = levels[cmd & 0xff]; + break; + } + } + return put_user(v, p); + } + return -EINVAL; +} + +static struct mixer_operations pas_mixer_operations = +{ + .owner = THIS_MODULE, + .id = "PAS16", + .name = "Pro Audio Spectrum 16", + .ioctl = pas_mixer_ioctl +}; + +int __init +pas_init_mixer(void) +{ + int d; + + levels = load_mixer_volumes("PAS16_1", default_levels, 1); + + pas_mixer_reset(); + + if ((d = sound_alloc_mixerdev()) != -1) + { + audio_devs[pas_audiodev]->mixer_dev = d; + mixer_devs[d] = &pas_mixer_operations; + } + return 1; +} diff --git a/sound/oss/pas2_pcm.c b/sound/oss/pas2_pcm.c new file mode 100644 index 00000000..8f7d1757 --- /dev/null +++ b/sound/oss/pas2_pcm.c @@ -0,0 +1,437 @@ +/* + * pas2_pcm.c Audio routines for PAS16 + * + * + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * + * + * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed) + * Alan Cox : Swatted a double allocation of device bug. Made a few + * more things module options. + * Bartlomiej Zolnierkiewicz : Added __init to pas_pcm_init() + */ + +#include <linux/init.h> +#include <linux/spinlock.h> +#include <linux/timex.h> +#include "sound_config.h" + +#include "pas2.h" + +#ifndef DEB +#define DEB(WHAT) +#endif + +#define PAS_PCM_INTRBITS (0x08) +/* + * Sample buffer timer interrupt enable + */ + +#define PCM_NON 0 +#define PCM_DAC 1 +#define PCM_ADC 2 + +static unsigned long pcm_speed; /* sampling rate */ +static unsigned char pcm_channels = 1; /* channels (1 or 2) */ +static unsigned char pcm_bits = 8; /* bits/sample (8 or 16) */ +static unsigned char pcm_filter; /* filter FLAG */ +static unsigned char pcm_mode = PCM_NON; +static unsigned long pcm_count; +static unsigned short pcm_bitsok = 8; /* mask of OK bits */ +static int pcm_busy; +int pas_audiodev = -1; +static int open_mode; + +extern spinlock_t pas_lock; + +static int pcm_set_speed(int arg) +{ + int foo, tmp; + unsigned long flags; + + if (arg == 0) + return pcm_speed; + + if (arg > 44100) + arg = 44100; + if (arg < 5000) + arg = 5000; + + if (pcm_channels & 2) + { + foo = ((CLOCK_TICK_RATE / 2) + (arg / 2)) / arg; + arg = ((CLOCK_TICK_RATE / 2) + (foo / 2)) / foo; + } + else + { + foo = (CLOCK_TICK_RATE + (arg / 2)) / arg; + arg = (CLOCK_TICK_RATE + (foo / 2)) / foo; + } + + pcm_speed = arg; + + tmp = pas_read(0x0B8A); + + /* + * Set anti-aliasing filters according to sample rate. You really *NEED* + * to enable this feature for all normal recording unless you want to + * experiment with aliasing effects. + * These filters apply to the selected "recording" source. + * I (pfw) don't know the encoding of these 5 bits. The values shown + * come from the SDK found on ftp.uwp.edu:/pub/msdos/proaudio/. + * + * I cleared bit 5 of these values, since that bit controls the master + * mute flag. (Olav Wölfelschneider) + * + */ +#if !defined NO_AUTO_FILTER_SET + tmp &= 0xe0; + if (pcm_speed >= 2 * 17897) + tmp |= 0x01; + else if (pcm_speed >= 2 * 15909) + tmp |= 0x02; + else if (pcm_speed >= 2 * 11931) + tmp |= 0x09; + else if (pcm_speed >= 2 * 8948) + tmp |= 0x11; + else if (pcm_speed >= 2 * 5965) + tmp |= 0x19; + else if (pcm_speed >= 2 * 2982) + tmp |= 0x04; + pcm_filter = tmp; +#endif + + spin_lock_irqsave(&pas_lock, flags); + + pas_write(tmp & ~(0x40 | 0x80), 0x0B8A); + pas_write(0x00 | 0x30 | 0x04, 0x138B); + pas_write(foo & 0xff, 0x1388); + pas_write((foo >> 8) & 0xff, 0x1388); + pas_write(tmp, 0x0B8A); + + spin_unlock_irqrestore(&pas_lock, flags); + + return pcm_speed; +} + +static int pcm_set_channels(int arg) +{ + + if ((arg != 1) && (arg != 2)) + return pcm_channels; + + if (arg != pcm_channels) + { + pas_write(pas_read(0xF8A) ^ 0x20, 0xF8A); + + pcm_channels = arg; + pcm_set_speed(pcm_speed); /* The speed must be reinitialized */ + } + return pcm_channels; +} + +static int pcm_set_bits(int arg) +{ + if (arg == 0) + return pcm_bits; + + if ((arg & pcm_bitsok) != arg) + return pcm_bits; + + if (arg != pcm_bits) + { + pas_write(pas_read(0x8389) ^ 0x04, 0x8389); + + pcm_bits = arg; + } + return pcm_bits; +} + +static int pas_audio_ioctl(int dev, unsigned int cmd, void __user *arg) +{ + int val, ret; + int __user *p = arg; + + DEB(printk("pas2_pcm.c: static int pas_audio_ioctl(unsigned int cmd = %X, unsigned int arg = %X)\n", cmd, arg)); + + switch (cmd) + { + case SOUND_PCM_WRITE_RATE: + if (get_user(val, p)) + return -EFAULT; + ret = pcm_set_speed(val); + break; + + case SOUND_PCM_READ_RATE: + ret = pcm_speed; + break; + + case SNDCTL_DSP_STEREO: + if (get_user(val, p)) + return -EFAULT; + ret = pcm_set_channels(val + 1) - 1; + break; + + case SOUND_PCM_WRITE_CHANNELS: + if (get_user(val, p)) + return -EFAULT; + ret = pcm_set_channels(val); + break; + + case SOUND_PCM_READ_CHANNELS: + ret = pcm_channels; + break; + + case SNDCTL_DSP_SETFMT: + if (get_user(val, p)) + return -EFAULT; + ret = pcm_set_bits(val); + break; + + case SOUND_PCM_READ_BITS: + ret = pcm_bits; + break; + + default: + return -EINVAL; + } + return put_user(ret, p); +} + +static void pas_audio_reset(int dev) +{ + DEB(printk("pas2_pcm.c: static void pas_audio_reset(void)\n")); + + pas_write(pas_read(0xF8A) & ~0x40, 0xF8A); /* Disable PCM */ +} + +static int pas_audio_open(int dev, int mode) +{ + int err; + unsigned long flags; + + DEB(printk("pas2_pcm.c: static int pas_audio_open(int mode = %X)\n", mode)); + + spin_lock_irqsave(&pas_lock, flags); + if (pcm_busy) + { + spin_unlock_irqrestore(&pas_lock, flags); + return -EBUSY; + } + pcm_busy = 1; + spin_unlock_irqrestore(&pas_lock, flags); + + if ((err = pas_set_intr(PAS_PCM_INTRBITS)) < 0) + return err; + + + pcm_count = 0; + open_mode = mode; + + return 0; +} + +static void pas_audio_close(int dev) +{ + unsigned long flags; + + DEB(printk("pas2_pcm.c: static void pas_audio_close(void)\n")); + + spin_lock_irqsave(&pas_lock, flags); + + pas_audio_reset(dev); + pas_remove_intr(PAS_PCM_INTRBITS); + pcm_mode = PCM_NON; + + pcm_busy = 0; + spin_unlock_irqrestore(&pas_lock, flags); +} + +static void pas_audio_output_block(int dev, unsigned long buf, int count, + int intrflag) +{ + unsigned long flags, cnt; + + DEB(printk("pas2_pcm.c: static void pas_audio_output_block(char *buf = %P, int count = %X)\n", buf, count)); + + cnt = count; + if (audio_devs[dev]->dmap_out->dma > 3) + cnt >>= 1; + + if (audio_devs[dev]->flags & DMA_AUTOMODE && + intrflag && + cnt == pcm_count) + return; + + spin_lock_irqsave(&pas_lock, flags); + + pas_write(pas_read(0xF8A) & ~0x40, + 0xF8A); + + /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_WRITE); */ + + if (audio_devs[dev]->dmap_out->dma > 3) + count >>= 1; + + if (count != pcm_count) + { + pas_write(pas_read(0x0B8A) & ~0x80, 0x0B8A); + pas_write(0x40 | 0x30 | 0x04, 0x138B); + pas_write(count & 0xff, 0x1389); + pas_write((count >> 8) & 0xff, 0x1389); + pas_write(pas_read(0x0B8A) | 0x80, 0x0B8A); + + pcm_count = count; + } + pas_write(pas_read(0x0B8A) | 0x80 | 0x40, 0x0B8A); +#ifdef NO_TRIGGER + pas_write(pas_read(0xF8A) | 0x40 | 0x10, 0xF8A); +#endif + + pcm_mode = PCM_DAC; + + spin_unlock_irqrestore(&pas_lock, flags); +} + +static void pas_audio_start_input(int dev, unsigned long buf, int count, + int intrflag) +{ + unsigned long flags; + int cnt; + + DEB(printk("pas2_pcm.c: static void pas_audio_start_input(char *buf = %P, int count = %X)\n", buf, count)); + + cnt = count; + if (audio_devs[dev]->dmap_out->dma > 3) + cnt >>= 1; + + if (audio_devs[pas_audiodev]->flags & DMA_AUTOMODE && + intrflag && + cnt == pcm_count) + return; + + spin_lock_irqsave(&pas_lock, flags); + + /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_READ); */ + + if (audio_devs[dev]->dmap_out->dma > 3) + count >>= 1; + + if (count != pcm_count) + { + pas_write(pas_read(0x0B8A) & ~0x80, 0x0B8A); + pas_write(0x40 | 0x30 | 0x04, 0x138B); + pas_write(count & 0xff, 0x1389); + pas_write((count >> 8) & 0xff, 0x1389); + pas_write(pas_read(0x0B8A) | 0x80, 0x0B8A); + + pcm_count = count; + } + pas_write(pas_read(0x0B8A) | 0x80 | 0x40, 0x0B8A); +#ifdef NO_TRIGGER + pas_write((pas_read(0xF8A) | 0x40) & ~0x10, 0xF8A); +#endif + + pcm_mode = PCM_ADC; + + spin_unlock_irqrestore(&pas_lock, flags); +} + +#ifndef NO_TRIGGER +static void pas_audio_trigger(int dev, int state) +{ + unsigned long flags; + + spin_lock_irqsave(&pas_lock, flags); + state &= open_mode; + + if (state & PCM_ENABLE_OUTPUT) + pas_write(pas_read(0xF8A) | 0x40 | 0x10, 0xF8A); + else if (state & PCM_ENABLE_INPUT) + pas_write((pas_read(0xF8A) | 0x40) & ~0x10, 0xF8A); + else + pas_write(pas_read(0xF8A) & ~0x40, 0xF8A); + + spin_unlock_irqrestore(&pas_lock, flags); +} +#endif + +static int pas_audio_prepare_for_input(int dev, int bsize, int bcount) +{ + pas_audio_reset(dev); + return 0; +} + +static int pas_audio_prepare_for_output(int dev, int bsize, int bcount) +{ + pas_audio_reset(dev); + return 0; +} + +static struct audio_driver pas_audio_driver = +{ + .owner = THIS_MODULE, + .open = pas_audio_open, + .close = pas_audio_close, + .output_block = pas_audio_output_block, + .start_input = pas_audio_start_input, + .ioctl = pas_audio_ioctl, + .prepare_for_input = pas_audio_prepare_for_input, + .prepare_for_output = pas_audio_prepare_for_output, + .halt_io = pas_audio_reset, + .trigger = pas_audio_trigger +}; + +void __init pas_pcm_init(struct address_info *hw_config) +{ + DEB(printk("pas2_pcm.c: long pas_pcm_init()\n")); + + pcm_bitsok = 8; + if (pas_read(0xEF8B) & 0x08) + pcm_bitsok |= 16; + + pcm_set_speed(DSP_DEFAULT_SPEED); + + if ((pas_audiodev = sound_install_audiodrv(AUDIO_DRIVER_VERSION, + "Pro Audio Spectrum", + &pas_audio_driver, + sizeof(struct audio_driver), + DMA_AUTOMODE, + AFMT_U8 | AFMT_S16_LE, + NULL, + hw_config->dma, + hw_config->dma)) < 0) + printk(KERN_WARNING "PAS16: Too many PCM devices available\n"); +} + +void pas_pcm_interrupt(unsigned char status, int cause) +{ + if (cause == 1) + { + /* + * Halt the PCM first. Otherwise we don't have time to start a new + * block before the PCM chip proceeds to the next sample + */ + + if (!(audio_devs[pas_audiodev]->flags & DMA_AUTOMODE)) + pas_write(pas_read(0xF8A) & ~0x40, 0xF8A); + + switch (pcm_mode) + { + case PCM_DAC: + DMAbuf_outputintr(pas_audiodev, 1); + break; + + case PCM_ADC: + DMAbuf_inputintr(pas_audiodev); + break; + + default: + printk(KERN_WARNING "PAS: Unexpected PCM interrupt\n"); + } + } +} diff --git a/sound/oss/pss.c b/sound/oss/pss.c new file mode 100644 index 00000000..9b800ce5 --- /dev/null +++ b/sound/oss/pss.c @@ -0,0 +1,1266 @@ +/* + * sound/oss/pss.c + * + * The low level driver for the Personal Sound System (ECHO ESC614). + * + * + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * + * + * Thomas Sailer ioctl code reworked (vmalloc/vfree removed) + * Alan Cox modularisation, clean up. + * + * 98-02-21: Vladimir Michl <vladimir.michl@upol.cz> + * Added mixer device for Beethoven ADSP-16 (master volume, + * bass, treble, synth), only for speakers. + * Fixed bug in pss_write (exchange parameters) + * Fixed config port of SB + * Requested two regions for PSS (PSS mixer, PSS config) + * Modified pss_download_boot + * To probe_pss_mss added test for initialize AD1848 + * 98-05-28: Vladimir Michl <vladimir.michl@upol.cz> + * Fixed computation of mixer volumes + * 04-05-1999: Anthony Barbachan <barbcode@xmen.cis.fordham.edu> + * Added code that allows the user to enable his cdrom and/or + * joystick through the module parameters pss_cdrom_port and + * pss_enable_joystick. pss_cdrom_port takes a port address as its + * argument. pss_enable_joystick takes either a 0 or a non-0 as its + * argument. + * 04-06-1999: Anthony Barbachan <barbcode@xmen.cis.fordham.edu> + * Separated some code into new functions for easier reuse. + * Cleaned up and streamlined new code. Added code to allow a user + * to only use this driver for enabling non-sound components + * through the new module parameter pss_no_sound (flag). Added + * code that would allow a user to decide whether the driver should + * reset the configured hardware settings for the PSS board through + * the module parameter pss_keep_settings (flag). This flag will + * allow a user to free up resources in use by this card if needbe, + * furthermore it allows him to use this driver to just enable the + * emulations and then be unloaded as it is no longer needed. Both + * new settings are only available to this driver if compiled as a + * module. The default settings of all new parameters are set to + * load the driver as it did in previous versions. + * 04-07-1999: Anthony Barbachan <barbcode@xmen.cis.fordham.edu> + * Added module parameter pss_firmware to allow the user to tell + * the driver where the firmware file is located. The default + * setting is the previous hardcoded setting "/etc/sound/pss_synth". + * 00-03-03: Christoph Hellwig <chhellwig@infradead.org> + * Adapted to module_init/module_exit + * 11-10-2000: Bartlomiej Zolnierkiewicz <bkz@linux-ide.org> + * Added __init to probe_pss(), attach_pss() and probe_pss_mpu() + * 02-Jan-2001: Chris Rankin + * Specify that this module owns the coprocessor + */ + + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/spinlock.h> + +#include "sound_config.h" +#include "sound_firmware.h" + +#include "ad1848.h" +#include "mpu401.h" + +/* + * PSS registers. + */ +#define REG(x) (devc->base+x) +#define PSS_DATA 0 +#define PSS_STATUS 2 +#define PSS_CONTROL 2 +#define PSS_ID 4 +#define PSS_IRQACK 4 +#define PSS_PIO 0x1a + +/* + * Config registers + */ +#define CONF_PSS 0x10 +#define CONF_WSS 0x12 +#define CONF_SB 0x14 +#define CONF_CDROM 0x16 +#define CONF_MIDI 0x18 + +/* + * Status bits. + */ +#define PSS_FLAG3 0x0800 +#define PSS_FLAG2 0x0400 +#define PSS_FLAG1 0x1000 +#define PSS_FLAG0 0x0800 +#define PSS_WRITE_EMPTY 0x8000 +#define PSS_READ_FULL 0x4000 + +/* + * WSS registers + */ +#define WSS_INDEX 4 +#define WSS_DATA 5 + +/* + * WSS status bits + */ +#define WSS_INITIALIZING 0x80 +#define WSS_AUTOCALIBRATION 0x20 + +#define NO_WSS_MIXER -1 + +#include "coproc.h" + +#include "pss_boot.h" + +/* If compiled into kernel, it enable or disable pss mixer */ +#ifdef CONFIG_PSS_MIXER +static int pss_mixer = 1; +#else +static int pss_mixer; +#endif + + +typedef struct pss_mixerdata { + unsigned int volume_l; + unsigned int volume_r; + unsigned int bass; + unsigned int treble; + unsigned int synth; +} pss_mixerdata; + +typedef struct pss_confdata { + int base; + int irq; + int dma; + int *osp; + pss_mixerdata mixer; + int ad_mixer_dev; +} pss_confdata; + +static pss_confdata pss_data; +static pss_confdata *devc = &pss_data; +static DEFINE_SPINLOCK(lock); + +static int pss_initialized; +static int nonstandard_microcode; +static int pss_cdrom_port = -1; /* Parameter for the PSS cdrom port */ +static int pss_enable_joystick; /* Parameter for enabling the joystick */ +static coproc_operations pss_coproc_operations; + +static void pss_write(pss_confdata *devc, int data) +{ + unsigned long i, limit; + + limit = jiffies + HZ/10; /* The timeout is 0.1 seconds */ + /* + * Note! the i<5000000 is an emergency exit. The dsp_command() is sometimes + * called while interrupts are disabled. This means that the timer is + * disabled also. However the timeout situation is a abnormal condition. + * Normally the DSP should be ready to accept commands after just couple of + * loops. + */ + + for (i = 0; i < 5000000 && time_before(jiffies, limit); i++) + { + if (inw(REG(PSS_STATUS)) & PSS_WRITE_EMPTY) + { + outw(data, REG(PSS_DATA)); + return; + } + } + printk(KERN_WARNING "PSS: DSP Command (%04x) Timeout.\n", data); +} + +static int __init probe_pss(struct address_info *hw_config) +{ + unsigned short id; + int irq, dma; + + devc->base = hw_config->io_base; + irq = devc->irq = hw_config->irq; + dma = devc->dma = hw_config->dma; + devc->osp = hw_config->osp; + + if (devc->base != 0x220 && devc->base != 0x240) + if (devc->base != 0x230 && devc->base != 0x250) /* Some cards use these */ + return 0; + + if (!request_region(devc->base, 0x10, "PSS mixer, SB emulation")) { + printk(KERN_ERR "PSS: I/O port conflict\n"); + return 0; + } + id = inw(REG(PSS_ID)); + if ((id >> 8) != 'E') { + printk(KERN_ERR "No PSS signature detected at 0x%x (0x%x)\n", devc->base, id); + release_region(devc->base, 0x10); + return 0; + } + if (!request_region(devc->base + 0x10, 0x9, "PSS config")) { + printk(KERN_ERR "PSS: I/O port conflict\n"); + release_region(devc->base, 0x10); + return 0; + } + return 1; +} + +static int set_irq(pss_confdata * devc, int dev, int irq) +{ + static unsigned short irq_bits[16] = + { + 0x0000, 0x0000, 0x0000, 0x0008, + 0x0000, 0x0010, 0x0000, 0x0018, + 0x0000, 0x0020, 0x0028, 0x0030, + 0x0038, 0x0000, 0x0000, 0x0000 + }; + + unsigned short tmp, bits; + + if (irq < 0 || irq > 15) + return 0; + + tmp = inw(REG(dev)) & ~0x38; /* Load confreg, mask IRQ bits out */ + + if ((bits = irq_bits[irq]) == 0 && irq != 0) + { + printk(KERN_ERR "PSS: Invalid IRQ %d\n", irq); + return 0; + } + outw(tmp | bits, REG(dev)); + return 1; +} + +static void set_io_base(pss_confdata * devc, int dev, int base) +{ + unsigned short tmp = inw(REG(dev)) & 0x003f; + unsigned short bits = (base & 0x0ffc) << 4; + + outw(bits | tmp, REG(dev)); +} + +static int set_dma(pss_confdata * devc, int dev, int dma) +{ + static unsigned short dma_bits[8] = + { + 0x0001, 0x0002, 0x0000, 0x0003, + 0x0000, 0x0005, 0x0006, 0x0007 + }; + + unsigned short tmp, bits; + + if (dma < 0 || dma > 7) + return 0; + + tmp = inw(REG(dev)) & ~0x07; /* Load confreg, mask DMA bits out */ + + if ((bits = dma_bits[dma]) == 0 && dma != 4) + { + printk(KERN_ERR "PSS: Invalid DMA %d\n", dma); + return 0; + } + outw(tmp | bits, REG(dev)); + return 1; +} + +static int pss_reset_dsp(pss_confdata * devc) +{ + unsigned long i, limit = jiffies + HZ/10; + + outw(0x2000, REG(PSS_CONTROL)); + for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++) + inw(REG(PSS_CONTROL)); + outw(0x0000, REG(PSS_CONTROL)); + return 1; +} + +static int pss_put_dspword(pss_confdata * devc, unsigned short word) +{ + int i, val; + + for (i = 0; i < 327680; i++) + { + val = inw(REG(PSS_STATUS)); + if (val & PSS_WRITE_EMPTY) + { + outw(word, REG(PSS_DATA)); + return 1; + } + } + return 0; +} + +static int pss_get_dspword(pss_confdata * devc, unsigned short *word) +{ + int i, val; + + for (i = 0; i < 327680; i++) + { + val = inw(REG(PSS_STATUS)); + if (val & PSS_READ_FULL) + { + *word = inw(REG(PSS_DATA)); + return 1; + } + } + return 0; +} + +static int pss_download_boot(pss_confdata * devc, unsigned char *block, int size, int flags) +{ + int i, val, count; + unsigned long limit; + + if (flags & CPF_FIRST) + { +/*_____ Warn DSP software that a boot is coming */ + outw(0x00fe, REG(PSS_DATA)); + + limit = jiffies + HZ/10; + for (i = 0; i < 32768 && time_before(jiffies, limit); i++) + if (inw(REG(PSS_DATA)) == 0x5500) + break; + + outw(*block++, REG(PSS_DATA)); + pss_reset_dsp(devc); + } + count = 1; + while ((flags&CPF_LAST) || count<size ) + { + int j; + + for (j = 0; j < 327670; j++) + { +/*_____ Wait for BG to appear */ + if (inw(REG(PSS_STATUS)) & PSS_FLAG3) + break; + } + + if (j == 327670) + { + /* It's ok we timed out when the file was empty */ + if (count >= size && flags & CPF_LAST) + break; + else + { + printk("\n"); + printk(KERN_ERR "PSS: Download timeout problems, byte %d=%d\n", count, size); + return 0; + } + } +/*_____ Send the next byte */ + if (count >= size) + { + /* If not data in block send 0xffff */ + outw (0xffff, REG (PSS_DATA)); + } + else + { + /*_____ Send the next byte */ + outw (*block++, REG (PSS_DATA)); + }; + count++; + } + + if (flags & CPF_LAST) + { +/*_____ Why */ + outw(0, REG(PSS_DATA)); + + limit = jiffies + HZ/10; + for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++) + val = inw(REG(PSS_STATUS)); + + limit = jiffies + HZ/10; + for (i = 0; i < 32768 && time_after_eq(limit, jiffies); i++) + { + val = inw(REG(PSS_STATUS)); + if (val & 0x4000) + break; + } + + /* now read the version */ + for (i = 0; i < 32000; i++) + { + val = inw(REG(PSS_STATUS)); + if (val & PSS_READ_FULL) + break; + } + if (i == 32000) + return 0; + + val = inw(REG(PSS_DATA)); + /* printk( "<PSS: microcode version %d.%d loaded>", val/16, val % 16); */ + } + return 1; +} + +/* Mixer */ +static void set_master_volume(pss_confdata *devc, int left, int right) +{ + static unsigned char log_scale[101] = { + 0xdb, 0xe0, 0xe3, 0xe5, 0xe7, 0xe9, 0xea, 0xeb, 0xec, 0xed, 0xed, 0xee, + 0xef, 0xef, 0xf0, 0xf0, 0xf1, 0xf1, 0xf2, 0xf2, 0xf2, 0xf3, 0xf3, 0xf3, + 0xf4, 0xf4, 0xf4, 0xf5, 0xf5, 0xf5, 0xf5, 0xf6, 0xf6, 0xf6, 0xf6, 0xf7, + 0xf7, 0xf7, 0xf7, 0xf7, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf9, 0xf9, 0xf9, + 0xf9, 0xf9, 0xf9, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfb, 0xfb, + 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, + 0xfc, 0xfc, 0xfc, 0xfc, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, + 0xfd, 0xfd, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, + 0xfe, 0xfe, 0xff, 0xff, 0xff + }; + pss_write(devc, 0x0010); + pss_write(devc, log_scale[left] | 0x0000); + pss_write(devc, 0x0010); + pss_write(devc, log_scale[right] | 0x0100); +} + +static void set_synth_volume(pss_confdata *devc, int volume) +{ + int vol = ((0x8000*volume)/100L); + pss_write(devc, 0x0080); + pss_write(devc, vol); + pss_write(devc, 0x0081); + pss_write(devc, vol); +} + +static void set_bass(pss_confdata *devc, int level) +{ + int vol = (int)(((0xfd - 0xf0) * level)/100L) + 0xf0; + pss_write(devc, 0x0010); + pss_write(devc, vol | 0x0200); +}; + +static void set_treble(pss_confdata *devc, int level) +{ + int vol = (((0xfd - 0xf0) * level)/100L) + 0xf0; + pss_write(devc, 0x0010); + pss_write(devc, vol | 0x0300); +}; + +static void pss_mixer_reset(pss_confdata *devc) +{ + set_master_volume(devc, 33, 33); + set_bass(devc, 50); + set_treble(devc, 50); + set_synth_volume(devc, 30); + pss_write (devc, 0x0010); + pss_write (devc, 0x0800 | 0xce); /* Stereo */ + + if(pss_mixer) + { + devc->mixer.volume_l = devc->mixer.volume_r = 33; + devc->mixer.bass = 50; + devc->mixer.treble = 50; + devc->mixer.synth = 30; + } +} + +static int set_volume_mono(unsigned __user *p, unsigned int *aleft) +{ + unsigned int left, volume; + if (get_user(volume, p)) + return -EFAULT; + + left = volume & 0xff; + if (left > 100) + left = 100; + *aleft = left; + return 0; +} + +static int set_volume_stereo(unsigned __user *p, + unsigned int *aleft, + unsigned int *aright) +{ + unsigned int left, right, volume; + if (get_user(volume, p)) + return -EFAULT; + + left = volume & 0xff; + if (left > 100) + left = 100; + right = (volume >> 8) & 0xff; + if (right > 100) + right = 100; + *aleft = left; + *aright = right; + return 0; +} + +static int ret_vol_mono(int left) +{ + return ((left << 8) | left); +} + +static int ret_vol_stereo(int left, int right) +{ + return ((right << 8) | left); +} + +static int call_ad_mixer(pss_confdata *devc,unsigned int cmd, void __user *arg) +{ + if (devc->ad_mixer_dev != NO_WSS_MIXER) + return mixer_devs[devc->ad_mixer_dev]->ioctl(devc->ad_mixer_dev, cmd, arg); + else + return -EINVAL; +} + +static int pss_mixer_ioctl (int dev, unsigned int cmd, void __user *arg) +{ + pss_confdata *devc = mixer_devs[dev]->devc; + int cmdf = cmd & 0xff; + + if ((cmdf != SOUND_MIXER_VOLUME) && (cmdf != SOUND_MIXER_BASS) && + (cmdf != SOUND_MIXER_TREBLE) && (cmdf != SOUND_MIXER_SYNTH) && + (cmdf != SOUND_MIXER_DEVMASK) && (cmdf != SOUND_MIXER_STEREODEVS) && + (cmdf != SOUND_MIXER_RECMASK) && (cmdf != SOUND_MIXER_CAPS) && + (cmdf != SOUND_MIXER_RECSRC)) + { + return call_ad_mixer(devc, cmd, arg); + } + + if (((cmd >> 8) & 0xff) != 'M') + return -EINVAL; + + if (_SIOC_DIR (cmd) & _SIOC_WRITE) + { + switch (cmdf) + { + case SOUND_MIXER_RECSRC: + if (devc->ad_mixer_dev != NO_WSS_MIXER) + return call_ad_mixer(devc, cmd, arg); + else + { + int v; + if (get_user(v, (int __user *)arg)) + return -EFAULT; + if (v != 0) + return -EINVAL; + return 0; + } + case SOUND_MIXER_VOLUME: + if (set_volume_stereo(arg, + &devc->mixer.volume_l, + &devc->mixer.volume_r)) + return -EFAULT; + set_master_volume(devc, devc->mixer.volume_l, + devc->mixer.volume_r); + return ret_vol_stereo(devc->mixer.volume_l, + devc->mixer.volume_r); + + case SOUND_MIXER_BASS: + if (set_volume_mono(arg, &devc->mixer.bass)) + return -EFAULT; + set_bass(devc, devc->mixer.bass); + return ret_vol_mono(devc->mixer.bass); + + case SOUND_MIXER_TREBLE: + if (set_volume_mono(arg, &devc->mixer.treble)) + return -EFAULT; + set_treble(devc, devc->mixer.treble); + return ret_vol_mono(devc->mixer.treble); + + case SOUND_MIXER_SYNTH: + if (set_volume_mono(arg, &devc->mixer.synth)) + return -EFAULT; + set_synth_volume(devc, devc->mixer.synth); + return ret_vol_mono(devc->mixer.synth); + + default: + return -EINVAL; + } + } + else + { + int val, and_mask = 0, or_mask = 0; + /* + * Return parameters + */ + switch (cmdf) + { + case SOUND_MIXER_DEVMASK: + if (call_ad_mixer(devc, cmd, arg) == -EINVAL) + break; + and_mask = ~0; + or_mask = SOUND_MASK_VOLUME | SOUND_MASK_BASS | SOUND_MASK_TREBLE | SOUND_MASK_SYNTH; + break; + + case SOUND_MIXER_STEREODEVS: + if (call_ad_mixer(devc, cmd, arg) == -EINVAL) + break; + and_mask = ~0; + or_mask = SOUND_MASK_VOLUME; + break; + + case SOUND_MIXER_RECMASK: + if (devc->ad_mixer_dev != NO_WSS_MIXER) + return call_ad_mixer(devc, cmd, arg); + break; + + case SOUND_MIXER_CAPS: + if (devc->ad_mixer_dev != NO_WSS_MIXER) + return call_ad_mixer(devc, cmd, arg); + or_mask = SOUND_CAP_EXCL_INPUT; + break; + + case SOUND_MIXER_RECSRC: + if (devc->ad_mixer_dev != NO_WSS_MIXER) + return call_ad_mixer(devc, cmd, arg); + break; + + case SOUND_MIXER_VOLUME: + or_mask = ret_vol_stereo(devc->mixer.volume_l, devc->mixer.volume_r); + break; + + case SOUND_MIXER_BASS: + or_mask = ret_vol_mono(devc->mixer.bass); + break; + + case SOUND_MIXER_TREBLE: + or_mask = ret_vol_mono(devc->mixer.treble); + break; + + case SOUND_MIXER_SYNTH: + or_mask = ret_vol_mono(devc->mixer.synth); + break; + default: + return -EINVAL; + } + if (get_user(val, (int __user *)arg)) + return -EFAULT; + val &= and_mask; + val |= or_mask; + if (put_user(val, (int __user *)arg)) + return -EFAULT; + return val; + } +} + +static struct mixer_operations pss_mixer_operations = +{ + .owner = THIS_MODULE, + .id = "SOUNDPORT", + .name = "PSS-AD1848", + .ioctl = pss_mixer_ioctl +}; + +static void disable_all_emulations(void) +{ + outw(0x0000, REG(CONF_PSS)); /* 0x0400 enables joystick */ + outw(0x0000, REG(CONF_WSS)); + outw(0x0000, REG(CONF_SB)); + outw(0x0000, REG(CONF_MIDI)); + outw(0x0000, REG(CONF_CDROM)); +} + +static void configure_nonsound_components(void) +{ + /* Configure Joystick port */ + + if(pss_enable_joystick) + { + outw(0x0400, REG(CONF_PSS)); /* 0x0400 enables joystick */ + printk(KERN_INFO "PSS: joystick enabled.\n"); + } + else + { + printk(KERN_INFO "PSS: joystick port not enabled.\n"); + } + + /* Configure CDROM port */ + + if (pss_cdrom_port == -1) { /* If cdrom port enablation wasn't requested */ + printk(KERN_INFO "PSS: CDROM port not enabled.\n"); + } else if (check_region(pss_cdrom_port, 2)) { + printk(KERN_ERR "PSS: CDROM I/O port conflict.\n"); + } else { + set_io_base(devc, CONF_CDROM, pss_cdrom_port); + printk(KERN_INFO "PSS: CDROM I/O port set to 0x%x.\n", pss_cdrom_port); + } +} + +static int __init attach_pss(struct address_info *hw_config) +{ + unsigned short id; + char tmp[100]; + + devc->base = hw_config->io_base; + devc->irq = hw_config->irq; + devc->dma = hw_config->dma; + devc->osp = hw_config->osp; + devc->ad_mixer_dev = NO_WSS_MIXER; + + if (!probe_pss(hw_config)) + return 0; + + id = inw(REG(PSS_ID)) & 0x00ff; + + /* + * Disable all emulations. Will be enabled later (if required). + */ + + disable_all_emulations(); + +#ifdef YOU_REALLY_WANT_TO_ALLOCATE_THESE_RESOURCES + if (sound_alloc_dma(hw_config->dma, "PSS")) + { + printk("pss.c: Can't allocate DMA channel.\n"); + release_region(hw_config->io_base, 0x10); + release_region(hw_config->io_base+0x10, 0x9); + return 0; + } + if (!set_irq(devc, CONF_PSS, devc->irq)) + { + printk("PSS: IRQ allocation error.\n"); + release_region(hw_config->io_base, 0x10); + release_region(hw_config->io_base+0x10, 0x9); + return 0; + } + if (!set_dma(devc, CONF_PSS, devc->dma)) + { + printk(KERN_ERR "PSS: DMA allocation error\n"); + release_region(hw_config->io_base, 0x10); + release_region(hw_config->io_base+0x10, 0x9); + return 0; + } +#endif + + configure_nonsound_components(); + pss_initialized = 1; + sprintf(tmp, "ECHO-PSS Rev. %d", id); + conf_printf(tmp, hw_config); + return 1; +} + +static int __init probe_pss_mpu(struct address_info *hw_config) +{ + struct resource *ports; + int timeout; + + if (!pss_initialized) + return 0; + + ports = request_region(hw_config->io_base, 2, "mpu401"); + + if (!ports) { + printk(KERN_ERR "PSS: MPU I/O port conflict\n"); + return 0; + } + set_io_base(devc, CONF_MIDI, hw_config->io_base); + if (!set_irq(devc, CONF_MIDI, hw_config->irq)) { + printk(KERN_ERR "PSS: MIDI IRQ allocation error.\n"); + goto fail; + } + if (!pss_synthLen) { + printk(KERN_ERR "PSS: Can't enable MPU. MIDI synth microcode not available.\n"); + goto fail; + } + if (!pss_download_boot(devc, pss_synth, pss_synthLen, CPF_FIRST | CPF_LAST)) { + printk(KERN_ERR "PSS: Unable to load MIDI synth microcode to DSP.\n"); + goto fail; + } + + /* + * Finally wait until the DSP algorithm has initialized itself and + * deactivates receive interrupt. + */ + + for (timeout = 900000; timeout > 0; timeout--) + { + if ((inb(hw_config->io_base + 1) & 0x80) == 0) /* Input data avail */ + inb(hw_config->io_base); /* Discard it */ + else + break; /* No more input */ + } + + if (!probe_mpu401(hw_config, ports)) + goto fail; + + attach_mpu401(hw_config, THIS_MODULE); /* Slot 1 */ + if (hw_config->slots[1] != -1) /* The MPU driver installed itself */ + midi_devs[hw_config->slots[1]]->coproc = &pss_coproc_operations; + return 1; +fail: + release_region(hw_config->io_base, 2); + return 0; +} + +static int pss_coproc_open(void *dev_info, int sub_device) +{ + switch (sub_device) + { + case COPR_MIDI: + if (pss_synthLen == 0) + { + printk(KERN_ERR "PSS: MIDI synth microcode not available.\n"); + return -EIO; + } + if (nonstandard_microcode) + if (!pss_download_boot(devc, pss_synth, pss_synthLen, CPF_FIRST | CPF_LAST)) + { + printk(KERN_ERR "PSS: Unable to load MIDI synth microcode to DSP.\n"); + return -EIO; + } + nonstandard_microcode = 0; + break; + + default: + break; + } + return 0; +} + +static void pss_coproc_close(void *dev_info, int sub_device) +{ + return; +} + +static void pss_coproc_reset(void *dev_info) +{ + if (pss_synthLen) + if (!pss_download_boot(devc, pss_synth, pss_synthLen, CPF_FIRST | CPF_LAST)) + { + printk(KERN_ERR "PSS: Unable to load MIDI synth microcode to DSP.\n"); + } + nonstandard_microcode = 0; +} + +static int download_boot_block(void *dev_info, copr_buffer * buf) +{ + if (buf->len <= 0 || buf->len > sizeof(buf->data)) + return -EINVAL; + + if (!pss_download_boot(devc, buf->data, buf->len, buf->flags)) + { + printk(KERN_ERR "PSS: Unable to load microcode block to DSP.\n"); + return -EIO; + } + nonstandard_microcode = 1; /* The MIDI microcode has been overwritten */ + return 0; +} + +static int pss_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg, int local) +{ + copr_buffer *buf; + copr_msg *mbuf; + copr_debug_buf dbuf; + unsigned short tmp; + unsigned long flags; + unsigned short *data; + int i, err; + /* printk( "PSS coproc ioctl %x %x %d\n", cmd, arg, local); */ + + switch (cmd) + { + case SNDCTL_COPR_RESET: + pss_coproc_reset(dev_info); + return 0; + + case SNDCTL_COPR_LOAD: + buf = vmalloc(sizeof(copr_buffer)); + if (buf == NULL) + return -ENOSPC; + if (copy_from_user(buf, arg, sizeof(copr_buffer))) { + vfree(buf); + return -EFAULT; + } + err = download_boot_block(dev_info, buf); + vfree(buf); + return err; + + case SNDCTL_COPR_SENDMSG: + mbuf = vmalloc(sizeof(copr_msg)); + if (mbuf == NULL) + return -ENOSPC; + if (copy_from_user(mbuf, arg, sizeof(copr_msg))) { + vfree(mbuf); + return -EFAULT; + } + data = (unsigned short *)(mbuf->data); + spin_lock_irqsave(&lock, flags); + for (i = 0; i < mbuf->len; i++) { + if (!pss_put_dspword(devc, *data++)) { + spin_unlock_irqrestore(&lock,flags); + mbuf->len = i; /* feed back number of WORDs sent */ + err = copy_to_user(arg, mbuf, sizeof(copr_msg)); + vfree(mbuf); + return err ? -EFAULT : -EIO; + } + } + spin_unlock_irqrestore(&lock,flags); + vfree(mbuf); + return 0; + + case SNDCTL_COPR_RCVMSG: + err = 0; + mbuf = vmalloc(sizeof(copr_msg)); + if (mbuf == NULL) + return -ENOSPC; + data = (unsigned short *)mbuf->data; + spin_lock_irqsave(&lock, flags); + for (i = 0; i < sizeof(mbuf->data)/sizeof(unsigned short); i++) { + mbuf->len = i; /* feed back number of WORDs read */ + if (!pss_get_dspword(devc, data++)) { + if (i == 0) + err = -EIO; + break; + } + } + spin_unlock_irqrestore(&lock,flags); + if (copy_to_user(arg, mbuf, sizeof(copr_msg))) + err = -EFAULT; + vfree(mbuf); + return err; + + case SNDCTL_COPR_RDATA: + if (copy_from_user(&dbuf, arg, sizeof(dbuf))) + return -EFAULT; + spin_lock_irqsave(&lock, flags); + if (!pss_put_dspword(devc, 0x00d0)) { + spin_unlock_irqrestore(&lock,flags); + return -EIO; + } + if (!pss_put_dspword(devc, (unsigned short)(dbuf.parm1 & 0xffff))) { + spin_unlock_irqrestore(&lock,flags); + return -EIO; + } + if (!pss_get_dspword(devc, &tmp)) { + spin_unlock_irqrestore(&lock,flags); + return -EIO; + } + dbuf.parm1 = tmp; + spin_unlock_irqrestore(&lock,flags); + if (copy_to_user(arg, &dbuf, sizeof(dbuf))) + return -EFAULT; + return 0; + + case SNDCTL_COPR_WDATA: + if (copy_from_user(&dbuf, arg, sizeof(dbuf))) + return -EFAULT; + spin_lock_irqsave(&lock, flags); + if (!pss_put_dspword(devc, 0x00d1)) { + spin_unlock_irqrestore(&lock,flags); + return -EIO; + } + if (!pss_put_dspword(devc, (unsigned short) (dbuf.parm1 & 0xffff))) { + spin_unlock_irqrestore(&lock,flags); + return -EIO; + } + tmp = (unsigned int)dbuf.parm2 & 0xffff; + if (!pss_put_dspword(devc, tmp)) { + spin_unlock_irqrestore(&lock,flags); + return -EIO; + } + spin_unlock_irqrestore(&lock,flags); + return 0; + + case SNDCTL_COPR_WCODE: + if (copy_from_user(&dbuf, arg, sizeof(dbuf))) + return -EFAULT; + spin_lock_irqsave(&lock, flags); + if (!pss_put_dspword(devc, 0x00d3)) { + spin_unlock_irqrestore(&lock,flags); + return -EIO; + } + if (!pss_put_dspword(devc, (unsigned short)(dbuf.parm1 & 0xffff))) { + spin_unlock_irqrestore(&lock,flags); + return -EIO; + } + tmp = (unsigned int)dbuf.parm2 & 0x00ff; + if (!pss_put_dspword(devc, tmp)) { + spin_unlock_irqrestore(&lock,flags); + return -EIO; + } + tmp = ((unsigned int)dbuf.parm2 >> 8) & 0xffff; + if (!pss_put_dspword(devc, tmp)) { + spin_unlock_irqrestore(&lock,flags); + return -EIO; + } + spin_unlock_irqrestore(&lock,flags); + return 0; + + case SNDCTL_COPR_RCODE: + if (copy_from_user(&dbuf, arg, sizeof(dbuf))) + return -EFAULT; + spin_lock_irqsave(&lock, flags); + if (!pss_put_dspword(devc, 0x00d2)) { + spin_unlock_irqrestore(&lock,flags); + return -EIO; + } + if (!pss_put_dspword(devc, (unsigned short)(dbuf.parm1 & 0xffff))) { + spin_unlock_irqrestore(&lock,flags); + return -EIO; + } + if (!pss_get_dspword(devc, &tmp)) { /* Read MSB */ + spin_unlock_irqrestore(&lock,flags); + return -EIO; + } + dbuf.parm1 = tmp << 8; + if (!pss_get_dspword(devc, &tmp)) { /* Read LSB */ + spin_unlock_irqrestore(&lock,flags); + return -EIO; + } + dbuf.parm1 |= tmp & 0x00ff; + spin_unlock_irqrestore(&lock,flags); + if (copy_to_user(arg, &dbuf, sizeof(dbuf))) + return -EFAULT; + return 0; + + default: + return -EINVAL; + } + return -EINVAL; +} + +static coproc_operations pss_coproc_operations = +{ + "ADSP-2115", + THIS_MODULE, + pss_coproc_open, + pss_coproc_close, + pss_coproc_ioctl, + pss_coproc_reset, + &pss_data +}; + +static int __init probe_pss_mss(struct address_info *hw_config) +{ + volatile int timeout; + struct resource *ports; + int my_mix = -999; /* gcc shut up */ + + if (!pss_initialized) + return 0; + + if (!request_region(hw_config->io_base, 4, "WSS config")) { + printk(KERN_ERR "PSS: WSS I/O port conflicts.\n"); + return 0; + } + ports = request_region(hw_config->io_base + 4, 4, "ad1848"); + if (!ports) { + printk(KERN_ERR "PSS: WSS I/O port conflicts.\n"); + release_region(hw_config->io_base, 4); + return 0; + } + set_io_base(devc, CONF_WSS, hw_config->io_base); + if (!set_irq(devc, CONF_WSS, hw_config->irq)) { + printk("PSS: WSS IRQ allocation error.\n"); + goto fail; + } + if (!set_dma(devc, CONF_WSS, hw_config->dma)) { + printk(KERN_ERR "PSS: WSS DMA allocation error\n"); + goto fail; + } + /* + * For some reason the card returns 0xff in the WSS status register + * immediately after boot. Probably MIDI+SB emulation algorithm + * downloaded to the ADSP2115 spends some time initializing the card. + * Let's try to wait until it finishes this task. + */ + for (timeout = 0; timeout < 100000 && (inb(hw_config->io_base + WSS_INDEX) & + WSS_INITIALIZING); timeout++) + ; + + outb((0x0b), hw_config->io_base + WSS_INDEX); /* Required by some cards */ + + for (timeout = 0; (inb(hw_config->io_base + WSS_DATA) & WSS_AUTOCALIBRATION) && + (timeout < 100000); timeout++) + ; + + if (!probe_ms_sound(hw_config, ports)) + goto fail; + + devc->ad_mixer_dev = NO_WSS_MIXER; + if (pss_mixer) + { + if ((my_mix = sound_install_mixer (MIXER_DRIVER_VERSION, + "PSS-SPEAKERS and AD1848 (through MSS audio codec)", + &pss_mixer_operations, + sizeof (struct mixer_operations), + devc)) < 0) + { + printk(KERN_ERR "Could not install PSS mixer\n"); + goto fail; + } + } + pss_mixer_reset(devc); + attach_ms_sound(hw_config, ports, THIS_MODULE); /* Slot 0 */ + + if (hw_config->slots[0] != -1) + { + /* The MSS driver installed itself */ + audio_devs[hw_config->slots[0]]->coproc = &pss_coproc_operations; + if (pss_mixer && (num_mixers == (my_mix + 2))) + { + /* The MSS mixer installed */ + devc->ad_mixer_dev = audio_devs[hw_config->slots[0]]->mixer_dev; + } + } + return 1; +fail: + release_region(hw_config->io_base + 4, 4); + release_region(hw_config->io_base, 4); + return 0; +} + +static inline void __exit unload_pss(struct address_info *hw_config) +{ + release_region(hw_config->io_base, 0x10); + release_region(hw_config->io_base+0x10, 0x9); +} + +static inline void __exit unload_pss_mpu(struct address_info *hw_config) +{ + unload_mpu401(hw_config); +} + +static inline void __exit unload_pss_mss(struct address_info *hw_config) +{ + unload_ms_sound(hw_config); +} + + +static struct address_info cfg; +static struct address_info cfg2; +static struct address_info cfg_mpu; + +static int pss_io __initdata = -1; +static int mss_io __initdata = -1; +static int mss_irq __initdata = -1; +static int mss_dma __initdata = -1; +static int mpu_io __initdata = -1; +static int mpu_irq __initdata = -1; +static int pss_no_sound = 0; /* Just configure non-sound components */ +static int pss_keep_settings = 1; /* Keep hardware settings at module exit */ +static char *pss_firmware = "/etc/sound/pss_synth"; + +module_param(pss_io, int, 0); +MODULE_PARM_DESC(pss_io, "Set i/o base of PSS card (probably 0x220 or 0x240)"); +module_param(mss_io, int, 0); +MODULE_PARM_DESC(mss_io, "Set WSS (audio) i/o base (0x530, 0x604, 0xE80, 0xF40, or other. Address must end in 0 or 4 and must be from 0x100 to 0xFF4)"); +module_param(mss_irq, int, 0); +MODULE_PARM_DESC(mss_irq, "Set WSS (audio) IRQ (3, 5, 7, 9, 10, 11, 12)"); +module_param(mss_dma, int, 0); +MODULE_PARM_DESC(mss_dma, "Set WSS (audio) DMA (0, 1, 3)"); +module_param(mpu_io, int, 0); +MODULE_PARM_DESC(mpu_io, "Set MIDI i/o base (0x330 or other. Address must be on 4 location boundaries and must be from 0x100 to 0xFFC)"); +module_param(mpu_irq, int, 0); +MODULE_PARM_DESC(mpu_irq, "Set MIDI IRQ (3, 5, 7, 9, 10, 11, 12)"); +module_param(pss_cdrom_port, int, 0); +MODULE_PARM_DESC(pss_cdrom_port, "Set the PSS CDROM port i/o base (0x340 or other)"); +module_param(pss_enable_joystick, bool, 0); +MODULE_PARM_DESC(pss_enable_joystick, "Enables the PSS joystick port (1 to enable, 0 to disable)"); +module_param(pss_no_sound, bool, 0); +MODULE_PARM_DESC(pss_no_sound, "Configure sound compoents (0 - no, 1 - yes)"); +module_param(pss_keep_settings, bool, 0); +MODULE_PARM_DESC(pss_keep_settings, "Keep hardware setting at driver unloading (0 - no, 1 - yes)"); +module_param(pss_firmware, charp, 0); +MODULE_PARM_DESC(pss_firmware, "Location of the firmware file (default - /etc/sound/pss_synth)"); +module_param(pss_mixer, bool, 0); +MODULE_PARM_DESC(pss_mixer, "Enable (1) or disable (0) PSS mixer (controlling of output volume, bass, treble, synth volume). The mixer is not available on all PSS cards."); +MODULE_AUTHOR("Hannu Savolainen, Vladimir Michl"); +MODULE_DESCRIPTION("Module for PSS sound cards (based on AD1848, ADSP-2115 and ESC614). This module includes control of output amplifier and synth volume of the Beethoven ADSP-16 card (this may work with other PSS cards)."); +MODULE_LICENSE("GPL"); + + +static int fw_load = 0; +static int pssmpu = 0, pssmss = 0; + +/* + * Load a PSS sound card module + */ + +static int __init init_pss(void) +{ + + if(pss_no_sound) /* If configuring only nonsound components */ + { + cfg.io_base = pss_io; + if(!probe_pss(&cfg)) + return -ENODEV; + printk(KERN_INFO "ECHO-PSS Rev. %d\n", inw(REG(PSS_ID)) & 0x00ff); + printk(KERN_INFO "PSS: loading in no sound mode.\n"); + disable_all_emulations(); + configure_nonsound_components(); + release_region(pss_io, 0x10); + release_region(pss_io + 0x10, 0x9); + return 0; + } + + cfg.io_base = pss_io; + + cfg2.io_base = mss_io; + cfg2.irq = mss_irq; + cfg2.dma = mss_dma; + + cfg_mpu.io_base = mpu_io; + cfg_mpu.irq = mpu_irq; + + if (cfg.io_base == -1 || cfg2.io_base == -1 || cfg2.irq == -1 || cfg.dma == -1) { + printk(KERN_INFO "pss: mss_io, mss_dma, mss_irq and pss_io must be set.\n"); + return -EINVAL; + } + + if (!pss_synth) { + fw_load = 1; + pss_synthLen = mod_firmware_load(pss_firmware, (void *) &pss_synth); + } + if (!attach_pss(&cfg)) + return -ENODEV; + /* + * Attach stuff + */ + if (probe_pss_mpu(&cfg_mpu)) + pssmpu = 1; + + if (probe_pss_mss(&cfg2)) + pssmss = 1; + + return 0; +} + +static void __exit cleanup_pss(void) +{ + if(!pss_no_sound) + { + if(fw_load && pss_synth) + vfree(pss_synth); + if(pssmss) + unload_pss_mss(&cfg2); + if(pssmpu) + unload_pss_mpu(&cfg_mpu); + unload_pss(&cfg); + } + + if(!pss_keep_settings) /* Keep hardware settings if asked */ + { + disable_all_emulations(); + printk(KERN_INFO "Resetting PSS sound card configurations.\n"); + } +} + +module_init(init_pss); +module_exit(cleanup_pss); + +#ifndef MODULE +static int __init setup_pss(char *str) +{ + /* io, mss_io, mss_irq, mss_dma, mpu_io, mpu_irq */ + int ints[7]; + + str = get_options(str, ARRAY_SIZE(ints), ints); + + pss_io = ints[1]; + mss_io = ints[2]; + mss_irq = ints[3]; + mss_dma = ints[4]; + mpu_io = ints[5]; + mpu_irq = ints[6]; + + return 1; +} + +__setup("pss=", setup_pss); +#endif diff --git a/sound/oss/sb.h b/sound/oss/sb.h new file mode 100644 index 00000000..77e8891c --- /dev/null +++ b/sound/oss/sb.h @@ -0,0 +1,185 @@ +#define DSP_RESET (devc->base + 0x6) +#define DSP_READ (devc->base + 0xA) +#define DSP_WRITE (devc->base + 0xC) +#define DSP_COMMAND (devc->base + 0xC) +#define DSP_STATUS (devc->base + 0xC) +#define DSP_DATA_AVAIL (devc->base + 0xE) +#define DSP_DATA_AVL16 (devc->base + 0xF) +#define MIXER_ADDR (devc->base + 0x4) +#define MIXER_DATA (devc->base + 0x5) +#define OPL3_LEFT (devc->base + 0x0) +#define OPL3_RIGHT (devc->base + 0x2) +#define OPL3_BOTH (devc->base + 0x8) +/* DSP Commands */ + +#define DSP_CMD_SPKON 0xD1 +#define DSP_CMD_SPKOFF 0xD3 +#define DSP_CMD_DMAON 0xD0 +#define DSP_CMD_DMAOFF 0xD4 + +#define IMODE_NONE 0 +#define IMODE_OUTPUT PCM_ENABLE_OUTPUT +#define IMODE_INPUT PCM_ENABLE_INPUT +#define IMODE_INIT 3 +#define IMODE_MIDI 4 + +#define NORMAL_MIDI 0 +#define UART_MIDI 1 + + +/* + * Device models + */ +#define MDL_NONE 0 +#define MDL_SB1 1 /* SB1.0 or 1.5 */ +#define MDL_SB2 2 /* SB2.0 */ +#define MDL_SB201 3 /* SB2.01 */ +#define MDL_SBPRO 4 /* SB Pro */ +#define MDL_SB16 5 /* SB16/32/AWE */ +#define MDL_SBPNP 6 /* SB16/32/AWE PnP */ +#define MDL_JAZZ 10 /* Media Vision Jazz16 */ +#define MDL_SMW 11 /* Logitech SoundMan Wave (Jazz16) */ +#define MDL_ESS 12 /* ESS ES688 and ES1688 */ +#define MDL_AZTECH 13 /* Aztech Sound Galaxy family */ +#define MDL_ES1868MIDI 14 /* MIDI port of ESS1868 */ +#define MDL_AEDSP 15 /* Audio Excel DSP 16 */ +#define MDL_ESSPCI 16 /* ESS PCI card */ +#define MDL_YMPCI 17 /* Yamaha PCI sb in emulation */ + +#define SUBMDL_ALS007 42 /* ALS-007 differs from SB16 only in mixer */ + /* register assignment */ +#define SUBMDL_ALS100 43 /* ALS-100 allows sampling rates of up */ + /* to 48kHz */ + +/* + * Config flags + */ +#define SB_NO_MIDI 0x00000001 +#define SB_NO_MIXER 0x00000002 +#define SB_NO_AUDIO 0x00000004 +#define SB_NO_RECORDING 0x00000008 /* No audio recording */ +#define SB_MIDI_ONLY (SB_NO_AUDIO|SB_NO_MIXER) +#define SB_PCI_IRQ 0x00000010 /* PCI shared IRQ */ + +struct mixer_def { + unsigned int regno: 8; + unsigned int bitoffs:4; + unsigned int nbits:4; +}; + +typedef struct mixer_def mixer_tab[32][2]; +typedef struct mixer_def mixer_ent; + +struct sb_module_options +{ + int esstype; /* ESS chip type */ + int acer; /* Do acer notebook init? */ + int sm_games; /* Logitech soundman games? */ +}; + +typedef struct sb_devc { + int dev; + + /* Hardware parameters */ + int *osp; + int minor, major; + int type; + int model, submodel; + int caps; +# define SBCAP_STEREO 0x00000001 +# define SBCAP_16BITS 0x00000002 + + /* Hardware resources */ + int base; + int irq; + int dma8, dma16; + + int pcibase; /* For ESS Maestro etc */ + + /* State variables */ + int opened; + /* new audio fields for full duplex support */ + int fullduplex; + int duplex; + int speed, bits, channels; + volatile int irq_ok; + volatile int intr_active, irq_mode; + /* duplicate audio fields for full duplex support */ + volatile int intr_active_16, irq_mode_16; + + /* Mixer fields */ + int *levels; + mixer_tab *iomap; + size_t iomap_sz; /* number or records in the iomap table */ + int mixer_caps, recmask, outmask, supported_devices; + int supported_rec_devices, supported_out_devices; + int my_mixerdev; + int sbmixnum; + + /* Audio fields */ + unsigned long trg_buf; + int trigger_bits; + int trg_bytes; + int trg_intrflag; + int trg_restart; + /* duplicate audio fields for full duplex support */ + unsigned long trg_buf_16; + int trigger_bits_16; + int trg_bytes_16; + int trg_intrflag_16; + int trg_restart_16; + + unsigned char tconst; + + /* MIDI fields */ + int my_mididev; + int input_opened; + int midi_broken; + void (*midi_input_intr) (int dev, unsigned char data); + void *midi_irq_cookie; /* IRQ cookie for the midi */ + + spinlock_t lock; + + struct sb_module_options sbmo; /* Module options */ + + } sb_devc; + +/* + * PCI card types + */ + +#define SB_PCI_ESSMAESTRO 1 /* ESS Maestro Legacy */ +#define SB_PCI_YAMAHA 2 /* Yamaha Legacy */ + +/* + * Functions + */ + +int sb_dsp_command (sb_devc *devc, unsigned char val); +int sb_dsp_get_byte(sb_devc * devc); +int sb_dsp_reset (sb_devc *devc); +void sb_setmixer (sb_devc *devc, unsigned int port, unsigned int value); +unsigned int sb_getmixer (sb_devc *devc, unsigned int port); +int sb_dsp_detect (struct address_info *hw_config, int pci, int pciio, struct sb_module_options *sbmo); +int sb_dsp_init (struct address_info *hw_config, struct module *owner); +void sb_dsp_unload(struct address_info *hw_config, int sbmpu); +int sb_mixer_init(sb_devc *devc, struct module *owner); +void sb_mixer_unload(sb_devc *devc); +void sb_mixer_set_stereo (sb_devc *devc, int mode); +void smw_mixer_init(sb_devc *devc); +void sb_dsp_midi_init (sb_devc *devc, struct module *owner); +void sb_audio_init (sb_devc *devc, char *name, struct module *owner); +void sb_midi_interrupt (sb_devc *devc); +void sb_chgmixer (sb_devc * devc, unsigned int reg, unsigned int mask, unsigned int val); +int sb_common_mixer_set(sb_devc * devc, int dev, int left, int right); + +int sb_audio_open(int dev, int mode); +void sb_audio_close(int dev); + +/* From sb_common.c */ +void sb_dsp_disable_midi(int port); +int probe_sbmpu (struct address_info *hw_config, struct module *owner); +void unload_sbmpu (struct address_info *hw_config); + +void unload_sb16(struct address_info *hw_info); +void unload_sb16midi(struct address_info *hw_info); diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c new file mode 100644 index 00000000..733b014e --- /dev/null +++ b/sound/oss/sb_audio.c @@ -0,0 +1,1098 @@ +/* + * sound/oss/sb_audio.c + * + * Audio routines for Sound Blaster compatible cards. + * + * + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * + * Changes + * Alan Cox : Formatting and clean ups + * + * Status + * Mostly working. Weird uart bug causing irq storms + * + * Daniel J. Rodriksson: Changes to make sb16 work full duplex. + * Maybe other 16 bit cards in this code could behave + * the same. + * Chris Rankin: Use spinlocks instead of CLI/STI + */ + +#include <linux/spinlock.h> + +#include "sound_config.h" + +#include "sb_mixer.h" +#include "sb.h" + +#include "sb_ess.h" + +int sb_audio_open(int dev, int mode) +{ + sb_devc *devc = audio_devs[dev]->devc; + unsigned long flags; + + if (devc == NULL) + { + printk(KERN_ERR "Sound Blaster: incomplete initialization.\n"); + return -ENXIO; + } + if (devc->caps & SB_NO_RECORDING && mode & OPEN_READ) + { + if (mode == OPEN_READ) + return -EPERM; + } + spin_lock_irqsave(&devc->lock, flags); + if (devc->opened) + { + spin_unlock_irqrestore(&devc->lock, flags); + return -EBUSY; + } + if (devc->dma16 != -1 && devc->dma16 != devc->dma8 && !devc->duplex) + { + if (sound_open_dma(devc->dma16, "Sound Blaster 16 bit")) + { + spin_unlock_irqrestore(&devc->lock, flags); + return -EBUSY; + } + } + devc->opened = mode; + spin_unlock_irqrestore(&devc->lock, flags); + + devc->irq_mode = IMODE_NONE; + devc->irq_mode_16 = IMODE_NONE; + devc->fullduplex = devc->duplex && + ((mode & OPEN_READ) && (mode & OPEN_WRITE)); + sb_dsp_reset(devc); + + /* At first glance this check isn't enough, some ESS chips might not + * have a RECLEV. However if they don't common_mixer_set will refuse + * cause devc->iomap has no register mapping for RECLEV + */ + if (devc->model == MDL_ESS) ess_mixer_reload (devc, SOUND_MIXER_RECLEV); + + /* The ALS007 seems to require that the DSP be removed from the output */ + /* in order for recording to be activated properly. This is done by */ + /* setting the appropriate bits of the output control register 4ch to */ + /* zero. This code assumes that the output control registers are not */ + /* used anywhere else and therefore the DSP bits are *always* ON for */ + /* output and OFF for sampling. */ + + if (devc->submodel == SUBMDL_ALS007) + { + if (mode & OPEN_READ) + sb_setmixer(devc,ALS007_OUTPUT_CTRL2, + sb_getmixer(devc,ALS007_OUTPUT_CTRL2) & 0xf9); + else + sb_setmixer(devc,ALS007_OUTPUT_CTRL2, + sb_getmixer(devc,ALS007_OUTPUT_CTRL2) | 0x06); + } + return 0; +} + +void sb_audio_close(int dev) +{ + sb_devc *devc = audio_devs[dev]->devc; + + /* fix things if mmap turned off fullduplex */ + if(devc->duplex + && !devc->fullduplex + && (devc->opened & OPEN_READ) && (devc->opened & OPEN_WRITE)) + { + struct dma_buffparms *dmap_temp; + dmap_temp = audio_devs[dev]->dmap_out; + audio_devs[dev]->dmap_out = audio_devs[dev]->dmap_in; + audio_devs[dev]->dmap_in = dmap_temp; + } + audio_devs[dev]->dmap_out->dma = devc->dma8; + audio_devs[dev]->dmap_in->dma = ( devc->duplex ) ? + devc->dma16 : devc->dma8; + + if (devc->dma16 != -1 && devc->dma16 != devc->dma8 && !devc->duplex) + sound_close_dma(devc->dma16); + + /* For ALS007, turn DSP output back on if closing the device for read */ + + if ((devc->submodel == SUBMDL_ALS007) && (devc->opened & OPEN_READ)) + { + sb_setmixer(devc,ALS007_OUTPUT_CTRL2, + sb_getmixer(devc,ALS007_OUTPUT_CTRL2) | 0x06); + } + devc->opened = 0; +} + +static void sb_set_output_parms(int dev, unsigned long buf, int nr_bytes, + int intrflag) +{ + sb_devc *devc = audio_devs[dev]->devc; + + if (!devc->fullduplex || devc->bits == AFMT_S16_LE) + { + devc->trg_buf = buf; + devc->trg_bytes = nr_bytes; + devc->trg_intrflag = intrflag; + devc->irq_mode = IMODE_OUTPUT; + } + else + { + devc->trg_buf_16 = buf; + devc->trg_bytes_16 = nr_bytes; + devc->trg_intrflag_16 = intrflag; + devc->irq_mode_16 = IMODE_OUTPUT; + } +} + +static void sb_set_input_parms(int dev, unsigned long buf, int count, int intrflag) +{ + sb_devc *devc = audio_devs[dev]->devc; + + if (!devc->fullduplex || devc->bits != AFMT_S16_LE) + { + devc->trg_buf = buf; + devc->trg_bytes = count; + devc->trg_intrflag = intrflag; + devc->irq_mode = IMODE_INPUT; + } + else + { + devc->trg_buf_16 = buf; + devc->trg_bytes_16 = count; + devc->trg_intrflag_16 = intrflag; + devc->irq_mode_16 = IMODE_INPUT; + } +} + +/* + * SB1.x compatible routines + */ + +static void sb1_audio_output_block(int dev, unsigned long buf, int nr_bytes, int intrflag) +{ + unsigned long flags; + int count = nr_bytes; + sb_devc *devc = audio_devs[dev]->devc; + + /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_WRITE); */ + + if (audio_devs[dev]->dmap_out->dma > 3) + count >>= 1; + count--; + + devc->irq_mode = IMODE_OUTPUT; + + spin_lock_irqsave(&devc->lock, flags); + if (sb_dsp_command(devc, 0x14)) /* 8 bit DAC using DMA */ + { + sb_dsp_command(devc, (unsigned char) (count & 0xff)); + sb_dsp_command(devc, (unsigned char) ((count >> 8) & 0xff)); + } + else + printk(KERN_WARNING "Sound Blaster: unable to start DAC.\n"); + spin_unlock_irqrestore(&devc->lock, flags); + devc->intr_active = 1; +} + +static void sb1_audio_start_input(int dev, unsigned long buf, int nr_bytes, int intrflag) +{ + unsigned long flags; + int count = nr_bytes; + sb_devc *devc = audio_devs[dev]->devc; + + /* + * Start a DMA input to the buffer pointed by dmaqtail + */ + + /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_READ); */ + + if (audio_devs[dev]->dmap_out->dma > 3) + count >>= 1; + count--; + + devc->irq_mode = IMODE_INPUT; + + spin_lock_irqsave(&devc->lock, flags); + if (sb_dsp_command(devc, 0x24)) /* 8 bit ADC using DMA */ + { + sb_dsp_command(devc, (unsigned char) (count & 0xff)); + sb_dsp_command(devc, (unsigned char) ((count >> 8) & 0xff)); + } + else + printk(KERN_ERR "Sound Blaster: unable to start ADC.\n"); + spin_unlock_irqrestore(&devc->lock, flags); + + devc->intr_active = 1; +} + +static void sb1_audio_trigger(int dev, int bits) +{ + sb_devc *devc = audio_devs[dev]->devc; + + bits &= devc->irq_mode; + + if (!bits) + sb_dsp_command(devc, 0xd0); /* Halt DMA */ + else + { + switch (devc->irq_mode) + { + case IMODE_INPUT: + sb1_audio_start_input(dev, devc->trg_buf, devc->trg_bytes, + devc->trg_intrflag); + break; + + case IMODE_OUTPUT: + sb1_audio_output_block(dev, devc->trg_buf, devc->trg_bytes, + devc->trg_intrflag); + break; + } + } + devc->trigger_bits = bits; +} + +static int sb1_audio_prepare_for_input(int dev, int bsize, int bcount) +{ + sb_devc *devc = audio_devs[dev]->devc; + unsigned long flags; + + spin_lock_irqsave(&devc->lock, flags); + if (sb_dsp_command(devc, 0x40)) + sb_dsp_command(devc, devc->tconst); + sb_dsp_command(devc, DSP_CMD_SPKOFF); + spin_unlock_irqrestore(&devc->lock, flags); + + devc->trigger_bits = 0; + return 0; +} + +static int sb1_audio_prepare_for_output(int dev, int bsize, int bcount) +{ + sb_devc *devc = audio_devs[dev]->devc; + unsigned long flags; + + spin_lock_irqsave(&devc->lock, flags); + if (sb_dsp_command(devc, 0x40)) + sb_dsp_command(devc, devc->tconst); + sb_dsp_command(devc, DSP_CMD_SPKON); + spin_unlock_irqrestore(&devc->lock, flags); + devc->trigger_bits = 0; + return 0; +} + +static int sb1_audio_set_speed(int dev, int speed) +{ + int max_speed = 23000; + sb_devc *devc = audio_devs[dev]->devc; + int tmp; + + if (devc->opened & OPEN_READ) + max_speed = 13000; + + if (speed > 0) + { + if (speed < 4000) + speed = 4000; + + if (speed > max_speed) + speed = max_speed; + + devc->tconst = (256 - ((1000000 + speed / 2) / speed)) & 0xff; + tmp = 256 - devc->tconst; + speed = (1000000 + tmp / 2) / tmp; + + devc->speed = speed; + } + return devc->speed; +} + +static short sb1_audio_set_channels(int dev, short channels) +{ + sb_devc *devc = audio_devs[dev]->devc; + return devc->channels = 1; +} + +static unsigned int sb1_audio_set_bits(int dev, unsigned int bits) +{ + sb_devc *devc = audio_devs[dev]->devc; + return devc->bits = 8; +} + +static void sb1_audio_halt_xfer(int dev) +{ + unsigned long flags; + sb_devc *devc = audio_devs[dev]->devc; + + spin_lock_irqsave(&devc->lock, flags); + sb_dsp_reset(devc); + spin_unlock_irqrestore(&devc->lock, flags); +} + +/* + * SB 2.0 and SB 2.01 compatible routines + */ + +static void sb20_audio_output_block(int dev, unsigned long buf, int nr_bytes, + int intrflag) +{ + unsigned long flags; + int count = nr_bytes; + sb_devc *devc = audio_devs[dev]->devc; + unsigned char cmd; + + /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_WRITE); */ + + if (audio_devs[dev]->dmap_out->dma > 3) + count >>= 1; + count--; + + devc->irq_mode = IMODE_OUTPUT; + + spin_lock_irqsave(&devc->lock, flags); + if (sb_dsp_command(devc, 0x48)) /* DSP Block size */ + { + sb_dsp_command(devc, (unsigned char) (count & 0xff)); + sb_dsp_command(devc, (unsigned char) ((count >> 8) & 0xff)); + + if (devc->speed * devc->channels <= 23000) + cmd = 0x1c; /* 8 bit PCM output */ + else + cmd = 0x90; /* 8 bit high speed PCM output (SB2.01/Pro) */ + + if (!sb_dsp_command(devc, cmd)) + printk(KERN_ERR "Sound Blaster: unable to start DAC.\n"); + } + else + printk(KERN_ERR "Sound Blaster: unable to start DAC.\n"); + spin_unlock_irqrestore(&devc->lock, flags); + devc->intr_active = 1; +} + +static void sb20_audio_start_input(int dev, unsigned long buf, int nr_bytes, int intrflag) +{ + unsigned long flags; + int count = nr_bytes; + sb_devc *devc = audio_devs[dev]->devc; + unsigned char cmd; + + /* + * Start a DMA input to the buffer pointed by dmaqtail + */ + + /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_READ); */ + + if (audio_devs[dev]->dmap_out->dma > 3) + count >>= 1; + count--; + + devc->irq_mode = IMODE_INPUT; + + spin_lock_irqsave(&devc->lock, flags); + if (sb_dsp_command(devc, 0x48)) /* DSP Block size */ + { + sb_dsp_command(devc, (unsigned char) (count & 0xff)); + sb_dsp_command(devc, (unsigned char) ((count >> 8) & 0xff)); + + if (devc->speed * devc->channels <= (devc->major == 3 ? 23000 : 13000)) + cmd = 0x2c; /* 8 bit PCM input */ + else + cmd = 0x98; /* 8 bit high speed PCM input (SB2.01/Pro) */ + + if (!sb_dsp_command(devc, cmd)) + printk(KERN_ERR "Sound Blaster: unable to start ADC.\n"); + } + else + printk(KERN_ERR "Sound Blaster: unable to start ADC.\n"); + spin_unlock_irqrestore(&devc->lock, flags); + devc->intr_active = 1; +} + +static void sb20_audio_trigger(int dev, int bits) +{ + sb_devc *devc = audio_devs[dev]->devc; + bits &= devc->irq_mode; + + if (!bits) + sb_dsp_command(devc, 0xd0); /* Halt DMA */ + else + { + switch (devc->irq_mode) + { + case IMODE_INPUT: + sb20_audio_start_input(dev, devc->trg_buf, devc->trg_bytes, + devc->trg_intrflag); + break; + + case IMODE_OUTPUT: + sb20_audio_output_block(dev, devc->trg_buf, devc->trg_bytes, + devc->trg_intrflag); + break; + } + } + devc->trigger_bits = bits; +} + +/* + * SB2.01 specific speed setup + */ + +static int sb201_audio_set_speed(int dev, int speed) +{ + sb_devc *devc = audio_devs[dev]->devc; + int tmp; + int s = speed * devc->channels; + + if (speed > 0) + { + if (speed < 4000) + speed = 4000; + if (speed > 44100) + speed = 44100; + if (devc->opened & OPEN_READ && speed > 15000) + speed = 15000; + devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff; + tmp = 256 - devc->tconst; + speed = ((1000000 + tmp / 2) / tmp) / devc->channels; + + devc->speed = speed; + } + return devc->speed; +} + +/* + * SB Pro specific routines + */ + +static int sbpro_audio_prepare_for_input(int dev, int bsize, int bcount) +{ /* For SB Pro and Jazz16 */ + sb_devc *devc = audio_devs[dev]->devc; + unsigned long flags; + unsigned char bits = 0; + + if (devc->dma16 >= 0 && devc->dma16 != devc->dma8) + audio_devs[dev]->dmap_out->dma = audio_devs[dev]->dmap_in->dma = + devc->bits == 16 ? devc->dma16 : devc->dma8; + + if (devc->model == MDL_JAZZ || devc->model == MDL_SMW) + if (devc->bits == AFMT_S16_LE) + bits = 0x04; /* 16 bit mode */ + + spin_lock_irqsave(&devc->lock, flags); + if (sb_dsp_command(devc, 0x40)) + sb_dsp_command(devc, devc->tconst); + sb_dsp_command(devc, DSP_CMD_SPKOFF); + if (devc->channels == 1) + sb_dsp_command(devc, 0xa0 | bits); /* Mono input */ + else + sb_dsp_command(devc, 0xa8 | bits); /* Stereo input */ + spin_unlock_irqrestore(&devc->lock, flags); + + devc->trigger_bits = 0; + return 0; +} + +static int sbpro_audio_prepare_for_output(int dev, int bsize, int bcount) +{ /* For SB Pro and Jazz16 */ + sb_devc *devc = audio_devs[dev]->devc; + unsigned long flags; + unsigned char tmp; + unsigned char bits = 0; + + if (devc->dma16 >= 0 && devc->dma16 != devc->dma8) + audio_devs[dev]->dmap_out->dma = audio_devs[dev]->dmap_in->dma = devc->bits == 16 ? devc->dma16 : devc->dma8; + if (devc->model == MDL_SBPRO) + sb_mixer_set_stereo(devc, devc->channels == 2); + + spin_lock_irqsave(&devc->lock, flags); + if (sb_dsp_command(devc, 0x40)) + sb_dsp_command(devc, devc->tconst); + sb_dsp_command(devc, DSP_CMD_SPKON); + + if (devc->model == MDL_JAZZ || devc->model == MDL_SMW) + { + if (devc->bits == AFMT_S16_LE) + bits = 0x04; /* 16 bit mode */ + + if (devc->channels == 1) + sb_dsp_command(devc, 0xa0 | bits); /* Mono output */ + else + sb_dsp_command(devc, 0xa8 | bits); /* Stereo output */ + spin_unlock_irqrestore(&devc->lock, flags); + } + else + { + spin_unlock_irqrestore(&devc->lock, flags); + tmp = sb_getmixer(devc, 0x0e); + if (devc->channels == 1) + tmp &= ~0x02; + else + tmp |= 0x02; + sb_setmixer(devc, 0x0e, tmp); + } + devc->trigger_bits = 0; + return 0; +} + +static int sbpro_audio_set_speed(int dev, int speed) +{ + sb_devc *devc = audio_devs[dev]->devc; + + if (speed > 0) + { + if (speed < 4000) + speed = 4000; + if (speed > 44100) + speed = 44100; + if (devc->channels > 1 && speed > 22050) + speed = 22050; + sb201_audio_set_speed(dev, speed); + } + return devc->speed; +} + +static short sbpro_audio_set_channels(int dev, short channels) +{ + sb_devc *devc = audio_devs[dev]->devc; + + if (channels == 1 || channels == 2) + { + if (channels != devc->channels) + { + devc->channels = channels; + if (devc->model == MDL_SBPRO && devc->channels == 2) + sbpro_audio_set_speed(dev, devc->speed); + } + } + return devc->channels; +} + +static int jazz16_audio_set_speed(int dev, int speed) +{ + sb_devc *devc = audio_devs[dev]->devc; + + if (speed > 0) + { + int tmp; + int s = speed * devc->channels; + + if (speed < 5000) + speed = 5000; + if (speed > 44100) + speed = 44100; + + devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff; + + tmp = 256 - devc->tconst; + speed = ((1000000 + tmp / 2) / tmp) / devc->channels; + + devc->speed = speed; + } + return devc->speed; +} + +/* + * SB16 specific routines + */ + +static int sb16_audio_set_speed(int dev, int speed) +{ + sb_devc *devc = audio_devs[dev]->devc; + int max_speed = devc->submodel == SUBMDL_ALS100 ? 48000 : 44100; + + if (speed > 0) + { + if (speed < 5000) + speed = 5000; + + if (speed > max_speed) + speed = max_speed; + + devc->speed = speed; + } + return devc->speed; +} + +static unsigned int sb16_audio_set_bits(int dev, unsigned int bits) +{ + sb_devc *devc = audio_devs[dev]->devc; + + if (bits != 0) + { + if (bits == AFMT_U8 || bits == AFMT_S16_LE) + devc->bits = bits; + else + devc->bits = AFMT_U8; + } + + return devc->bits; +} + +static int sb16_audio_prepare_for_input(int dev, int bsize, int bcount) +{ + sb_devc *devc = audio_devs[dev]->devc; + + if (!devc->fullduplex) + { + audio_devs[dev]->dmap_out->dma = + audio_devs[dev]->dmap_in->dma = + devc->bits == AFMT_S16_LE ? + devc->dma16 : devc->dma8; + } + else if (devc->bits == AFMT_S16_LE) + { + audio_devs[dev]->dmap_out->dma = devc->dma8; + audio_devs[dev]->dmap_in->dma = devc->dma16; + } + else + { + audio_devs[dev]->dmap_out->dma = devc->dma16; + audio_devs[dev]->dmap_in->dma = devc->dma8; + } + + devc->trigger_bits = 0; + return 0; +} + +static int sb16_audio_prepare_for_output(int dev, int bsize, int bcount) +{ + sb_devc *devc = audio_devs[dev]->devc; + + if (!devc->fullduplex) + { + audio_devs[dev]->dmap_out->dma = + audio_devs[dev]->dmap_in->dma = + devc->bits == AFMT_S16_LE ? + devc->dma16 : devc->dma8; + } + else if (devc->bits == AFMT_S16_LE) + { + audio_devs[dev]->dmap_out->dma = devc->dma8; + audio_devs[dev]->dmap_in->dma = devc->dma16; + } + else + { + audio_devs[dev]->dmap_out->dma = devc->dma16; + audio_devs[dev]->dmap_in->dma = devc->dma8; + } + + devc->trigger_bits = 0; + return 0; +} + +static void sb16_audio_output_block(int dev, unsigned long buf, int count, + int intrflag) +{ + unsigned long flags, cnt; + sb_devc *devc = audio_devs[dev]->devc; + unsigned long bits; + + if (!devc->fullduplex || devc->bits == AFMT_S16_LE) + { + devc->irq_mode = IMODE_OUTPUT; + devc->intr_active = 1; + } + else + { + devc->irq_mode_16 = IMODE_OUTPUT; + devc->intr_active_16 = 1; + } + + /* save value */ + spin_lock_irqsave(&devc->lock, flags); + bits = devc->bits; + if (devc->fullduplex) + devc->bits = (devc->bits == AFMT_S16_LE) ? + AFMT_U8 : AFMT_S16_LE; + spin_unlock_irqrestore(&devc->lock, flags); + + cnt = count; + if (devc->bits == AFMT_S16_LE) + cnt >>= 1; + cnt--; + + spin_lock_irqsave(&devc->lock, flags); + + /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_WRITE); */ + + sb_dsp_command(devc, 0x41); + sb_dsp_command(devc, (unsigned char) ((devc->speed >> 8) & 0xff)); + sb_dsp_command(devc, (unsigned char) (devc->speed & 0xff)); + + sb_dsp_command(devc, (devc->bits == AFMT_S16_LE ? 0xb6 : 0xc6)); + sb_dsp_command(devc, ((devc->channels == 2 ? 0x20 : 0) + + (devc->bits == AFMT_S16_LE ? 0x10 : 0))); + sb_dsp_command(devc, (unsigned char) (cnt & 0xff)); + sb_dsp_command(devc, (unsigned char) (cnt >> 8)); + + /* restore real value after all programming */ + devc->bits = bits; + spin_unlock_irqrestore(&devc->lock, flags); +} + + +/* + * This fails on the Cyrix MediaGX. If you don't have the DMA enabled + * before the first sample arrives it locks up. However even if you + * do enable the DMA in time you just get DMA timeouts and missing + * interrupts and stuff, so for now I've not bothered fixing this either. + */ + +static void sb16_audio_start_input(int dev, unsigned long buf, int count, int intrflag) +{ + unsigned long flags, cnt; + sb_devc *devc = audio_devs[dev]->devc; + + if (!devc->fullduplex || devc->bits != AFMT_S16_LE) + { + devc->irq_mode = IMODE_INPUT; + devc->intr_active = 1; + } + else + { + devc->irq_mode_16 = IMODE_INPUT; + devc->intr_active_16 = 1; + } + + cnt = count; + if (devc->bits == AFMT_S16_LE) + cnt >>= 1; + cnt--; + + spin_lock_irqsave(&devc->lock, flags); + + /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_READ); */ + + sb_dsp_command(devc, 0x42); + sb_dsp_command(devc, (unsigned char) ((devc->speed >> 8) & 0xff)); + sb_dsp_command(devc, (unsigned char) (devc->speed & 0xff)); + + sb_dsp_command(devc, (devc->bits == AFMT_S16_LE ? 0xbe : 0xce)); + sb_dsp_command(devc, ((devc->channels == 2 ? 0x20 : 0) + + (devc->bits == AFMT_S16_LE ? 0x10 : 0))); + sb_dsp_command(devc, (unsigned char) (cnt & 0xff)); + sb_dsp_command(devc, (unsigned char) (cnt >> 8)); + + spin_unlock_irqrestore(&devc->lock, flags); +} + +static void sb16_audio_trigger(int dev, int bits) +{ + sb_devc *devc = audio_devs[dev]->devc; + + int bits_16 = bits & devc->irq_mode_16; + bits &= devc->irq_mode; + + if (!bits && !bits_16) + sb_dsp_command(devc, 0xd0); /* Halt DMA */ + else + { + if (bits) + { + switch (devc->irq_mode) + { + case IMODE_INPUT: + sb16_audio_start_input(dev, + devc->trg_buf, + devc->trg_bytes, + devc->trg_intrflag); + break; + + case IMODE_OUTPUT: + sb16_audio_output_block(dev, + devc->trg_buf, + devc->trg_bytes, + devc->trg_intrflag); + break; + } + } + if (bits_16) + { + switch (devc->irq_mode_16) + { + case IMODE_INPUT: + sb16_audio_start_input(dev, + devc->trg_buf_16, + devc->trg_bytes_16, + devc->trg_intrflag_16); + break; + + case IMODE_OUTPUT: + sb16_audio_output_block(dev, + devc->trg_buf_16, + devc->trg_bytes_16, + devc->trg_intrflag_16); + break; + } + } + } + + devc->trigger_bits = bits | bits_16; +} + +static unsigned char lbuf8[2048]; +static signed short *lbuf16 = (signed short *)lbuf8; +#define LBUFCOPYSIZE 1024 +static void +sb16_copy_from_user(int dev, + char *localbuf, int localoffs, + const char __user *userbuf, int useroffs, + int max_in, int max_out, + int *used, int *returned, + int len) +{ + sb_devc *devc = audio_devs[dev]->devc; + int i, c, p, locallen; + unsigned char *buf8; + signed short *buf16; + + /* if not duplex no conversion */ + if (!devc->fullduplex) + { + if (copy_from_user(localbuf + localoffs, + userbuf + useroffs, len)) + return; + *used = len; + *returned = len; + } + else if (devc->bits == AFMT_S16_LE) + { + /* 16 -> 8 */ + /* max_in >> 1, max number of samples in ( 16 bits ) */ + /* max_out, max number of samples out ( 8 bits ) */ + /* len, number of samples that will be taken ( 16 bits )*/ + /* c, count of samples remaining in buffer ( 16 bits )*/ + /* p, count of samples already processed ( 16 bits )*/ + len = ( (max_in >> 1) > max_out) ? max_out : (max_in >> 1); + c = len; + p = 0; + buf8 = (unsigned char *)(localbuf + localoffs); + while (c) + { + locallen = (c >= LBUFCOPYSIZE ? LBUFCOPYSIZE : c); + /* << 1 in order to get 16 bit samples */ + if (copy_from_user(lbuf16, + userbuf + useroffs + (p << 1), + locallen << 1)) + return; + for (i = 0; i < locallen; i++) + { + buf8[p+i] = ~((lbuf16[i] >> 8) & 0xff) ^ 0x80; + } + c -= locallen; p += locallen; + } + /* used = ( samples * 16 bits size ) */ + *used = max_in > ( max_out << 1) ? (max_out << 1) : max_in; + /* returned = ( samples * 8 bits size ) */ + *returned = len; + } + else + { + /* 8 -> 16 */ + /* max_in, max number of samples in ( 8 bits ) */ + /* max_out >> 1, max number of samples out ( 16 bits ) */ + /* len, number of samples that will be taken ( 8 bits )*/ + /* c, count of samples remaining in buffer ( 8 bits )*/ + /* p, count of samples already processed ( 8 bits )*/ + len = max_in > (max_out >> 1) ? (max_out >> 1) : max_in; + c = len; + p = 0; + buf16 = (signed short *)(localbuf + localoffs); + while (c) + { + locallen = (c >= LBUFCOPYSIZE ? LBUFCOPYSIZE : c); + if (copy_from_user(lbuf8, + userbuf+useroffs + p, + locallen)) + return; + for (i = 0; i < locallen; i++) + { + buf16[p+i] = (~lbuf8[i] ^ 0x80) << 8; + } + c -= locallen; p += locallen; + } + /* used = ( samples * 8 bits size ) */ + *used = len; + /* returned = ( samples * 16 bits size ) */ + *returned = len << 1; + } +} + +static void +sb16_audio_mmap(int dev) +{ + sb_devc *devc = audio_devs[dev]->devc; + devc->fullduplex = 0; +} + +static struct audio_driver sb1_audio_driver = /* SB1.x */ +{ + .owner = THIS_MODULE, + .open = sb_audio_open, + .close = sb_audio_close, + .output_block = sb_set_output_parms, + .start_input = sb_set_input_parms, + .prepare_for_input = sb1_audio_prepare_for_input, + .prepare_for_output = sb1_audio_prepare_for_output, + .halt_io = sb1_audio_halt_xfer, + .trigger = sb1_audio_trigger, + .set_speed = sb1_audio_set_speed, + .set_bits = sb1_audio_set_bits, + .set_channels = sb1_audio_set_channels +}; + +static struct audio_driver sb20_audio_driver = /* SB2.0 */ +{ + .owner = THIS_MODULE, + .open = sb_audio_open, + .close = sb_audio_close, + .output_block = sb_set_output_parms, + .start_input = sb_set_input_parms, + .prepare_for_input = sb1_audio_prepare_for_input, + .prepare_for_output = sb1_audio_prepare_for_output, + .halt_io = sb1_audio_halt_xfer, + .trigger = sb20_audio_trigger, + .set_speed = sb1_audio_set_speed, + .set_bits = sb1_audio_set_bits, + .set_channels = sb1_audio_set_channels +}; + +static struct audio_driver sb201_audio_driver = /* SB2.01 */ +{ + .owner = THIS_MODULE, + .open = sb_audio_open, + .close = sb_audio_close, + .output_block = sb_set_output_parms, + .start_input = sb_set_input_parms, + .prepare_for_input = sb1_audio_prepare_for_input, + .prepare_for_output = sb1_audio_prepare_for_output, + .halt_io = sb1_audio_halt_xfer, + .trigger = sb20_audio_trigger, + .set_speed = sb201_audio_set_speed, + .set_bits = sb1_audio_set_bits, + .set_channels = sb1_audio_set_channels +}; + +static struct audio_driver sbpro_audio_driver = /* SB Pro */ +{ + .owner = THIS_MODULE, + .open = sb_audio_open, + .close = sb_audio_close, + .output_block = sb_set_output_parms, + .start_input = sb_set_input_parms, + .prepare_for_input = sbpro_audio_prepare_for_input, + .prepare_for_output = sbpro_audio_prepare_for_output, + .halt_io = sb1_audio_halt_xfer, + .trigger = sb20_audio_trigger, + .set_speed = sbpro_audio_set_speed, + .set_bits = sb1_audio_set_bits, + .set_channels = sbpro_audio_set_channels +}; + +static struct audio_driver jazz16_audio_driver = /* Jazz16 and SM Wave */ +{ + .owner = THIS_MODULE, + .open = sb_audio_open, + .close = sb_audio_close, + .output_block = sb_set_output_parms, + .start_input = sb_set_input_parms, + .prepare_for_input = sbpro_audio_prepare_for_input, + .prepare_for_output = sbpro_audio_prepare_for_output, + .halt_io = sb1_audio_halt_xfer, + .trigger = sb20_audio_trigger, + .set_speed = jazz16_audio_set_speed, + .set_bits = sb16_audio_set_bits, + .set_channels = sbpro_audio_set_channels +}; + +static struct audio_driver sb16_audio_driver = /* SB16 */ +{ + .owner = THIS_MODULE, + .open = sb_audio_open, + .close = sb_audio_close, + .output_block = sb_set_output_parms, + .start_input = sb_set_input_parms, + .prepare_for_input = sb16_audio_prepare_for_input, + .prepare_for_output = sb16_audio_prepare_for_output, + .halt_io = sb1_audio_halt_xfer, + .copy_user = sb16_copy_from_user, + .trigger = sb16_audio_trigger, + .set_speed = sb16_audio_set_speed, + .set_bits = sb16_audio_set_bits, + .set_channels = sbpro_audio_set_channels, + .mmap = sb16_audio_mmap +}; + +void sb_audio_init(sb_devc * devc, char *name, struct module *owner) +{ + int audio_flags = 0; + int format_mask = AFMT_U8; + + struct audio_driver *driver = &sb1_audio_driver; + + switch (devc->model) + { + case MDL_SB1: /* SB1.0 or SB 1.5 */ + DDB(printk("Will use standard SB1.x driver\n")); + audio_flags = DMA_HARDSTOP; + break; + + case MDL_SB2: + DDB(printk("Will use SB2.0 driver\n")); + audio_flags = DMA_AUTOMODE; + driver = &sb20_audio_driver; + break; + + case MDL_SB201: + DDB(printk("Will use SB2.01 (high speed) driver\n")); + audio_flags = DMA_AUTOMODE; + driver = &sb201_audio_driver; + break; + + case MDL_JAZZ: + case MDL_SMW: + DDB(printk("Will use Jazz16 driver\n")); + audio_flags = DMA_AUTOMODE; + format_mask |= AFMT_S16_LE; + driver = &jazz16_audio_driver; + break; + + case MDL_ESS: + DDB(printk("Will use ESS ES688/1688 driver\n")); + driver = ess_audio_init (devc, &audio_flags, &format_mask); + break; + + case MDL_SB16: + DDB(printk("Will use SB16 driver\n")); + audio_flags = DMA_AUTOMODE; + format_mask |= AFMT_S16_LE; + if (devc->dma8 != devc->dma16 && devc->dma16 != -1) + { + audio_flags |= DMA_DUPLEX; + devc->duplex = 1; + } + driver = &sb16_audio_driver; + break; + + default: + DDB(printk("Will use SB Pro driver\n")); + audio_flags = DMA_AUTOMODE; + driver = &sbpro_audio_driver; + } + + if (owner) + driver->owner = owner; + + if ((devc->dev = sound_install_audiodrv(AUDIO_DRIVER_VERSION, + name,driver, sizeof(struct audio_driver), + audio_flags, format_mask, devc, + devc->dma8, + devc->duplex ? devc->dma16 : devc->dma8)) < 0) + { + printk(KERN_ERR "Sound Blaster: unable to install audio.\n"); + return; + } + audio_devs[devc->dev]->mixer_dev = devc->my_mixerdev; + audio_devs[devc->dev]->min_fragment = 5; +} diff --git a/sound/oss/sb_card.c b/sound/oss/sb_card.c new file mode 100644 index 00000000..fb5d7250 --- /dev/null +++ b/sound/oss/sb_card.c @@ -0,0 +1,354 @@ +/* + * sound/oss/sb_card.c + * + * Detection routine for the ISA Sound Blaster and compatible sound + * cards. + * + * This file is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this + * software for more info. + * + * This is a complete rewrite of the detection routines. This was + * prompted by the PnP API change during v2.5 and the ugly state the + * code was in. + * + * Copyright (C) by Paul Laufer 2002. Based on code originally by + * Hannu Savolainen which was modified by many others over the + * years. Authors specifically mentioned in the previous version were: + * Daniel Stone, Alessandro Zummo, Jeff Garzik, Arnaldo Carvalho de + * Melo, Daniel Church, and myself. + * + * 02-05-2003 Original Release, Paul Laufer <paul@laufernet.com> + * 02-07-2003 Bug made it into first release. Take two. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/slab.h> +#include <linux/init.h> +#include "sound_config.h" +#include "sb_mixer.h" +#include "sb.h" +#ifdef CONFIG_PNP +#include <linux/pnp.h> +#endif /* CONFIG_PNP */ +#include "sb_card.h" + +MODULE_DESCRIPTION("OSS Soundblaster ISA PnP and legacy sound driver"); +MODULE_LICENSE("GPL"); + +extern void *smw_free; + +static int __initdata mpu_io = 0; +static int __initdata io = -1; +static int __initdata irq = -1; +static int __initdata dma = -1; +static int __initdata dma16 = -1; +static int __initdata type = 0; /* Can set this to a specific card type */ +static int __initdata esstype = 0; /* ESS chip type */ +static int __initdata acer = 0; /* Do acer notebook init? */ +static int __initdata sm_games = 0; /* Logitech soundman games? */ + +static struct sb_card_config *legacy = NULL; + +#ifdef CONFIG_PNP +static int pnp_registered; +static int __initdata pnp = 1; +/* +static int __initdata uart401 = 0; +*/ +#else +static int __initdata pnp = 0; +#endif + +module_param(io, int, 000); +MODULE_PARM_DESC(io, "Soundblaster i/o base address (0x220,0x240,0x260,0x280)"); +module_param(irq, int, 000); +MODULE_PARM_DESC(irq, "IRQ (5,7,9,10)"); +module_param(dma, int, 000); +MODULE_PARM_DESC(dma, "8-bit DMA channel (0,1,3)"); +module_param(dma16, int, 000); +MODULE_PARM_DESC(dma16, "16-bit DMA channel (5,6,7)"); +module_param(mpu_io, int, 000); +MODULE_PARM_DESC(mpu_io, "MPU base address"); +module_param(type, int, 000); +MODULE_PARM_DESC(type, "You can set this to specific card type (doesn't " \ + "work with pnp)"); +module_param(sm_games, int, 000); +MODULE_PARM_DESC(sm_games, "Enable support for Logitech soundman games " \ + "(doesn't work with pnp)"); +module_param(esstype, int, 000); +MODULE_PARM_DESC(esstype, "ESS chip type (doesn't work with pnp)"); +module_param(acer, int, 000); +MODULE_PARM_DESC(acer, "Set this to detect cards in some ACER notebooks "\ + "(doesn't work with pnp)"); + +#ifdef CONFIG_PNP +module_param(pnp, int, 000); +MODULE_PARM_DESC(pnp, "Went set to 0 will disable detection using PnP. "\ + "Default is 1.\n"); +/* Not done yet.... */ +/* +module_param(uart401, int, 000); +MODULE_PARM_DESC(uart401, "When set to 1, will attempt to detect and enable"\ + "the mpu on some clones"); +*/ +#endif /* CONFIG_PNP */ + +/* OSS subsystem card registration shared by PnP and legacy routines */ +static int sb_register_oss(struct sb_card_config *scc, struct sb_module_options *sbmo) +{ + if (!request_region(scc->conf.io_base, 16, "soundblaster")) { + printk(KERN_ERR "sb: ports busy.\n"); + kfree(scc); + return -EBUSY; + } + + if (!sb_dsp_detect(&scc->conf, 0, 0, sbmo)) { + release_region(scc->conf.io_base, 16); + printk(KERN_ERR "sb: Failed DSP Detect.\n"); + kfree(scc); + return -ENODEV; + } + if(!sb_dsp_init(&scc->conf, THIS_MODULE)) { + printk(KERN_ERR "sb: Failed DSP init.\n"); + kfree(scc); + return -ENODEV; + } + if(scc->mpucnf.io_base > 0) { + scc->mpu = 1; + printk(KERN_INFO "sb: Turning on MPU\n"); + if(!probe_sbmpu(&scc->mpucnf, THIS_MODULE)) + scc->mpu = 0; + } + + return 1; +} + +static void sb_unload(struct sb_card_config *scc) +{ + sb_dsp_unload(&scc->conf, 0); + if(scc->mpu) + unload_sbmpu(&scc->mpucnf); + kfree(scc); +} + +/* Register legacy card with OSS subsystem */ +static int __init sb_init_legacy(void) +{ + struct sb_module_options sbmo = {0}; + + if((legacy = kzalloc(sizeof(struct sb_card_config), GFP_KERNEL)) == NULL) { + printk(KERN_ERR "sb: Error: Could not allocate memory\n"); + return -ENOMEM; + } + + legacy->conf.io_base = io; + legacy->conf.irq = irq; + legacy->conf.dma = dma; + legacy->conf.dma2 = dma16; + legacy->conf.card_subtype = type; + + legacy->mpucnf.io_base = mpu_io; + legacy->mpucnf.irq = -1; + legacy->mpucnf.dma = -1; + legacy->mpucnf.dma2 = -1; + + sbmo.esstype = esstype; + sbmo.sm_games = sm_games; + sbmo.acer = acer; + + return sb_register_oss(legacy, &sbmo); +} + +#ifdef CONFIG_PNP + +/* Populate the OSS subsystem structures with information from PnP */ +static void sb_dev2cfg(struct pnp_dev *dev, struct sb_card_config *scc) +{ + scc->conf.io_base = -1; + scc->conf.irq = -1; + scc->conf.dma = -1; + scc->conf.dma2 = -1; + scc->mpucnf.io_base = -1; + scc->mpucnf.irq = -1; + scc->mpucnf.dma = -1; + scc->mpucnf.dma2 = -1; + + /* All clones layout their PnP tables differently and some use + different logical devices for the MPU */ + if(!strncmp("CTL",scc->card_id,3)) { + scc->conf.io_base = pnp_port_start(dev,0); + scc->conf.irq = pnp_irq(dev,0); + scc->conf.dma = pnp_dma(dev,0); + scc->conf.dma2 = pnp_dma(dev,1); + scc->mpucnf.io_base = pnp_port_start(dev,1); + return; + } + if(!strncmp("tBA",scc->card_id,3)) { + scc->conf.io_base = pnp_port_start(dev,0); + scc->conf.irq = pnp_irq(dev,0); + scc->conf.dma = pnp_dma(dev,0); + scc->conf.dma2 = pnp_dma(dev,1); + return; + } + if(!strncmp("ESS",scc->card_id,3)) { + scc->conf.io_base = pnp_port_start(dev,0); + scc->conf.irq = pnp_irq(dev,0); + scc->conf.dma = pnp_dma(dev,0); + scc->conf.dma2 = pnp_dma(dev,1); + scc->mpucnf.io_base = pnp_port_start(dev,2); + return; + } + if(!strncmp("CMI",scc->card_id,3)) { + scc->conf.io_base = pnp_port_start(dev,0); + scc->conf.irq = pnp_irq(dev,0); + scc->conf.dma = pnp_dma(dev,0); + scc->conf.dma2 = pnp_dma(dev,1); + return; + } + if(!strncmp("RWB",scc->card_id,3)) { + scc->conf.io_base = pnp_port_start(dev,0); + scc->conf.irq = pnp_irq(dev,0); + scc->conf.dma = pnp_dma(dev,0); + return; + } + if(!strncmp("ALS",scc->card_id,3)) { + if(!strncmp("ALS0007",scc->card_id,7)) { + scc->conf.io_base = pnp_port_start(dev,0); + scc->conf.irq = pnp_irq(dev,0); + scc->conf.dma = pnp_dma(dev,0); + } else { + scc->conf.io_base = pnp_port_start(dev,0); + scc->conf.irq = pnp_irq(dev,0); + scc->conf.dma = pnp_dma(dev,1); + scc->conf.dma2 = pnp_dma(dev,0); + } + return; + } + if(!strncmp("RTL",scc->card_id,3)) { + scc->conf.io_base = pnp_port_start(dev,0); + scc->conf.irq = pnp_irq(dev,0); + scc->conf.dma = pnp_dma(dev,1); + scc->conf.dma2 = pnp_dma(dev,0); + } +} + +static unsigned int sb_pnp_devices; + +/* Probe callback function for the PnP API */ +static int sb_pnp_probe(struct pnp_card_link *card, const struct pnp_card_device_id *card_id) +{ + struct sb_card_config *scc; + struct sb_module_options sbmo = {0}; /* Default to 0 for PnP */ + struct pnp_dev *dev = pnp_request_card_device(card, card_id->devs[0].id, NULL); + + if(!dev){ + return -EBUSY; + } + + if((scc = kzalloc(sizeof(struct sb_card_config), GFP_KERNEL)) == NULL) { + printk(KERN_ERR "sb: Error: Could not allocate memory\n"); + return -ENOMEM; + } + + printk(KERN_INFO "sb: PnP: Found Card Named = \"%s\", Card PnP id = " \ + "%s, Device PnP id = %s\n", card->card->name, card_id->id, + dev->id->id); + + scc->card_id = card_id->id; + scc->dev_id = dev->id->id; + sb_dev2cfg(dev, scc); + + printk(KERN_INFO "sb: PnP: Detected at: io=0x%x, irq=%d, " \ + "dma=%d, dma16=%d\n", scc->conf.io_base, scc->conf.irq, + scc->conf.dma, scc->conf.dma2); + + pnp_set_card_drvdata(card, scc); + sb_pnp_devices++; + + return sb_register_oss(scc, &sbmo); +} + +static void sb_pnp_remove(struct pnp_card_link *card) +{ + struct sb_card_config *scc = pnp_get_card_drvdata(card); + + if(!scc) + return; + + printk(KERN_INFO "sb: PnP: Removing %s\n", scc->card_id); + + sb_unload(scc); +} + +static struct pnp_card_driver sb_pnp_driver = { + .name = "OSS SndBlstr", /* 16 character limit */ + .id_table = sb_pnp_card_table, + .probe = sb_pnp_probe, + .remove = sb_pnp_remove, +}; +MODULE_DEVICE_TABLE(pnp_card, sb_pnp_card_table); +#endif /* CONFIG_PNP */ + +static void sb_unregister_all(void) +{ +#ifdef CONFIG_PNP + if (pnp_registered) + pnp_unregister_card_driver(&sb_pnp_driver); +#endif +} + +static int __init sb_init(void) +{ + int lres = 0; + int pres = 0; + + printk(KERN_INFO "sb: Init: Starting Probe...\n"); + + if(io != -1 && irq != -1 && dma != -1) { + printk(KERN_INFO "sb: Probing legacy card with io=%x, "\ + "irq=%d, dma=%d, dma16=%d\n",io, irq, dma, dma16); + lres = sb_init_legacy(); + } else if((io != -1 || irq != -1 || dma != -1) || + (!pnp && (io == -1 && irq == -1 && dma == -1))) + printk(KERN_ERR "sb: Error: At least io, irq, and dma "\ + "must be set for legacy cards.\n"); + +#ifdef CONFIG_PNP + if(pnp) { + int err = pnp_register_card_driver(&sb_pnp_driver); + if (!err) + pnp_registered = 1; + pres = sb_pnp_devices; + } +#endif + printk(KERN_INFO "sb: Init: Done\n"); + + /* If either PnP or Legacy registered a card then return + * success */ + if (pres == 0 && lres <= 0) { + sb_unregister_all(); + return -ENODEV; + } + return 0; +} + +static void __exit sb_exit(void) +{ + printk(KERN_INFO "sb: Unloading...\n"); + + /* Unload legacy card */ + if (legacy) { + printk (KERN_INFO "sb: Unloading legacy card\n"); + sb_unload(legacy); + } + + sb_unregister_all(); + + vfree(smw_free); + smw_free = NULL; +} + +module_init(sb_init); +module_exit(sb_exit); diff --git a/sound/oss/sb_card.h b/sound/oss/sb_card.h new file mode 100644 index 00000000..5535cff8 --- /dev/null +++ b/sound/oss/sb_card.h @@ -0,0 +1,149 @@ +/* + * sound/oss/sb_card.h + * + * This file is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this + * software for more info. + * + * 02-05-2002 Original Release, Paul Laufer <paul@laufernet.com> + */ + +struct sb_card_config { + struct address_info conf; + struct address_info mpucnf; + const char *card_id; + const char *dev_id; + int mpu; +}; + +#ifdef CONFIG_PNP + +/* + * SoundBlaster PnP tables and structures. + */ + +/* Card PnP ID Table */ +static struct pnp_card_device_id sb_pnp_card_table[] = { + /* Sound Blaster 16 */ + {.id = "CTL0024", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster 16 */ + {.id = "CTL0025", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster 16 */ + {.id = "CTL0026", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster 16 */ + {.id = "CTL0027", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster 16 */ + {.id = "CTL0028", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster 16 */ + {.id = "CTL0029", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster 16 */ + {.id = "CTL002a", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster 16 */ + {.id = "CTL002b", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster 16 */ + {.id = "CTL002c", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster 16 */ + {.id = "CTL00ed", .driver_data = 0, .devs = { {.id="CTL0041"}, } }, + /* Sound Blaster 16 */ + {.id = "CTL0086", .driver_data = 0, .devs = { {.id="CTL0041"}, } }, + /* Sound Blaster Vibra16S */ + {.id = "CTL0051", .driver_data = 0, .devs = { {.id="CTL0001"}, } }, + /* Sound Blaster Vibra16C */ + {.id = "CTL0070", .driver_data = 0, .devs = { {.id="CTL0001"}, } }, + /* Sound Blaster Vibra16CL */ + {.id = "CTL0080", .driver_data = 0, .devs = { {.id="CTL0041"}, } }, + /* Sound Blaster Vibra16CL */ + {.id = "CTL00F0", .driver_data = 0, .devs = { {.id="CTL0043"}, } }, + /* Sound Blaster AWE 32 */ + {.id = "CTL0039", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster AWE 32 */ + {.id = "CTL0042", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster AWE 32 */ + {.id = "CTL0043", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster AWE 32 */ + {.id = "CTL0044", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster AWE 32 */ + {.id = "CTL0045", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster AWE 32 */ + {.id = "CTL0046", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster AWE 32 */ + {.id = "CTL0047", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster AWE 32 */ + {.id = "CTL0048", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster AWE 32 */ + {.id = "CTL0054", .driver_data = 0, .devs = { {.id="CTL0031"}, } }, + /* Sound Blaster AWE 32 */ + {.id = "CTL009C", .driver_data = 0, .devs = { {.id="CTL0041"}, } }, + /* Createive SB32 PnP */ + {.id = "CTL009F", .driver_data = 0, .devs = { {.id="CTL0041"}, } }, + /* Sound Blaster AWE 64 */ + {.id = "CTL009D", .driver_data = 0, .devs = { {.id="CTL0042"}, } }, + /* Sound Blaster AWE 64 Gold */ + {.id = "CTL009E", .driver_data = 0, .devs = { {.id="CTL0044"}, } }, + /* Sound Blaster AWE 64 Gold */ + {.id = "CTL00B2", .driver_data = 0, .devs = { {.id="CTL0044"}, } }, + /* Sound Blaster AWE 64 */ + {.id = "CTL00C1", .driver_data = 0, .devs = { {.id="CTL0042"}, } }, + /* Sound Blaster AWE 64 */ + {.id = "CTL00C3", .driver_data = 0, .devs = { {.id="CTL0045"}, } }, + /* Sound Blaster AWE 64 */ + {.id = "CTL00C5", .driver_data = 0, .devs = { {.id="CTL0045"}, } }, + /* Sound Blaster AWE 64 */ + {.id = "CTL00C7", .driver_data = 0, .devs = { {.id="CTL0045"}, } }, + /* Sound Blaster AWE 64 */ + {.id = "CTL00E4", .driver_data = 0, .devs = { {.id="CTL0045"}, } }, + /* Sound Blaster AWE 64 */ + {.id = "CTL00E9", .driver_data = 0, .devs = { {.id="CTL0045"}, } }, + /* ESS 1868 */ + {.id = "ESS0968", .driver_data = 0, .devs = { {.id="ESS0968"}, } }, + /* ESS 1868 */ + {.id = "ESS1868", .driver_data = 0, .devs = { {.id="ESS1868"}, } }, + /* ESS 1868 */ + {.id = "ESS1868", .driver_data = 0, .devs = { {.id="ESS8611"}, } }, + /* ESS 1869 PnP AudioDrive */ + {.id = "ESS0003", .driver_data = 0, .devs = { {.id="ESS1869"}, } }, + /* ESS 1869 */ + {.id = "ESS1869", .driver_data = 0, .devs = { {.id="ESS1869"}, } }, + /* ESS 1878 */ + {.id = "ESS1878", .driver_data = 0, .devs = { {.id="ESS1878"}, } }, + /* ESS 1879 */ + {.id = "ESS1879", .driver_data = 0, .devs = { {.id="ESS1879"}, } }, + /* CMI 8330 SoundPRO */ + {.id = "CMI0001", .driver_data = 0, .devs = { {.id="@X@0001"}, + {.id="@H@0001"}, + {.id="@@@0001"}, } }, + /* Diamond DT0197H */ + {.id = "RWR1688", .driver_data = 0, .devs = { {.id="@@@0001"}, + {.id="@X@0001"}, + {.id="@H@0001"}, } }, + /* ALS007 */ + {.id = "ALS0007", .driver_data = 0, .devs = { {.id="@@@0001"}, + {.id="@X@0001"}, + {.id="@H@0001"}, } }, + /* ALS100 */ + {.id = "ALS0001", .driver_data = 0, .devs = { {.id="@@@0001"}, + {.id="@X@0001"}, + {.id="@H@0001"}, } }, + /* ALS110 */ + {.id = "ALS0110", .driver_data = 0, .devs = { {.id="@@@1001"}, + {.id="@X@1001"}, + {.id="@H@0001"}, } }, + /* ALS120 */ + {.id = "ALS0120", .driver_data = 0, .devs = { {.id="@@@2001"}, + {.id="@X@2001"}, + {.id="@H@0001"}, } }, + /* ALS200 */ + {.id = "ALS0200", .driver_data = 0, .devs = { {.id="@@@0020"}, + {.id="@X@0030"}, + {.id="@H@0001"}, } }, + /* ALS200 */ + {.id = "RTL3000", .driver_data = 0, .devs = { {.id="@@@2001"}, + {.id="@X@2001"}, + {.id="@H@0001"}, } }, + /* Sound Blaster 16 (Virtual PC 2004) */ + {.id = "tBA03b0", .driver_data = 0, .devs = { {.id="PNPb003"}, } }, + /* -end- */ + {.id = "", } +}; + +#endif diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c new file mode 100644 index 00000000..7d42c541 --- /dev/null +++ b/sound/oss/sb_common.c @@ -0,0 +1,1292 @@ +/* + * sound/oss/sb_common.c + * + * Common routines for Sound Blaster compatible cards. + * + * + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * + * + * Daniel J. Rodriksson: Modified sbintr to handle 8 and 16 bit interrupts + * for full duplex support ( only sb16 by now ) + * Rolf Fokkens: Added (BETA?) support for ES1887 chips. + * (fokkensr@vertis.nl) Which means: You can adjust the recording levels. + * + * 2000/01/18 - separated sb_card and sb_common - + * Jeff Garzik <jgarzik@pobox.com> + * + * 2000/09/18 - got rid of attach_uart401 + * Arnaldo Carvalho de Melo <acme@conectiva.com.br> + * + * 2001/01/26 - replaced CLI/STI with spinlocks + * Chris Rankin <rankinc@zipworld.com.au> + */ + +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/spinlock.h> +#include <linux/slab.h> + +#include "sound_config.h" +#include "sound_firmware.h" + +#include "mpu401.h" + +#include "sb_mixer.h" +#include "sb.h" +#include "sb_ess.h" + +/* + * global module flag + */ + +int sb_be_quiet; + +static sb_devc *detected_devc; /* For communication from probe to init */ +static sb_devc *last_devc; /* For MPU401 initialization */ + +static unsigned char jazz_irq_bits[] = { + 0, 0, 2, 3, 0, 1, 0, 4, 0, 2, 5, 0, 0, 0, 0, 6 +}; + +static unsigned char jazz_dma_bits[] = { + 0, 1, 0, 2, 0, 3, 0, 4 +}; + +void *smw_free; + +/* + * Jazz16 chipset specific control variables + */ + +static int jazz16_base; /* Not detected */ +static unsigned char jazz16_bits; /* I/O relocation bits */ +static DEFINE_SPINLOCK(jazz16_lock); + +/* + * Logitech Soundman Wave specific initialization code + */ + +#ifdef SMW_MIDI0001_INCLUDED +#include "smw-midi0001.h" +#else +static unsigned char *smw_ucode; +static int smw_ucodeLen; + +#endif + +static sb_devc *last_sb; /* Last sb loaded */ + +int sb_dsp_command(sb_devc * devc, unsigned char val) +{ + int i; + unsigned long limit; + + limit = jiffies + HZ / 10; /* Timeout */ + + /* + * Note! the i<500000 is an emergency exit. The sb_dsp_command() is sometimes + * called while interrupts are disabled. This means that the timer is + * disabled also. However the timeout situation is a abnormal condition. + * Normally the DSP should be ready to accept commands after just couple of + * loops. + */ + + for (i = 0; i < 500000 && (limit-jiffies)>0; i++) + { + if ((inb(DSP_STATUS) & 0x80) == 0) + { + outb((val), DSP_COMMAND); + return 1; + } + } + printk(KERN_WARNING "Sound Blaster: DSP command(%x) timeout.\n", val); + return 0; +} + +int sb_dsp_get_byte(sb_devc * devc) +{ + int i; + + for (i = 1000; i; i--) + { + if (inb(DSP_DATA_AVAIL) & 0x80) + return inb(DSP_READ); + } + return 0xffff; +} + +static void sb_intr (sb_devc *devc) +{ + int status; + unsigned char src = 0xff; + + if (devc->model == MDL_SB16) + { + src = sb_getmixer(devc, IRQ_STAT); /* Interrupt source register */ + + if (src & 4) /* MPU401 interrupt */ + if(devc->midi_irq_cookie) + uart401intr(devc->irq, devc->midi_irq_cookie); + + if (!(src & 3)) + return; /* Not a DSP interrupt */ + } + if (devc->intr_active && (!devc->fullduplex || (src & 0x01))) + { + switch (devc->irq_mode) + { + case IMODE_OUTPUT: + DMAbuf_outputintr(devc->dev, 1); + break; + + case IMODE_INPUT: + DMAbuf_inputintr(devc->dev); + break; + + case IMODE_INIT: + break; + + case IMODE_MIDI: + sb_midi_interrupt(devc); + break; + + default: + /* printk(KERN_WARNING "Sound Blaster: Unexpected interrupt\n"); */ + ; + } + } + else if (devc->intr_active_16 && (src & 0x02)) + { + switch (devc->irq_mode_16) + { + case IMODE_OUTPUT: + DMAbuf_outputintr(devc->dev, 1); + break; + + case IMODE_INPUT: + DMAbuf_inputintr(devc->dev); + break; + + case IMODE_INIT: + break; + + default: + /* printk(KERN_WARNING "Sound Blaster: Unexpected interrupt\n"); */ + ; + } + } + /* + * Acknowledge interrupts + */ + + if (src & 0x01) + status = inb(DSP_DATA_AVAIL); + + if (devc->model == MDL_SB16 && src & 0x02) + status = inb(DSP_DATA_AVL16); +} + +static void pci_intr(sb_devc *devc) +{ + int src = inb(devc->pcibase+0x1A); + src&=3; + if(src) + sb_intr(devc); +} + +static irqreturn_t sbintr(int irq, void *dev_id) +{ + sb_devc *devc = dev_id; + + devc->irq_ok = 1; + + switch (devc->model) { + case MDL_ESSPCI: + pci_intr (devc); + break; + + case MDL_ESS: + ess_intr (devc); + break; + default: + sb_intr (devc); + break; + } + return IRQ_HANDLED; +} + +int sb_dsp_reset(sb_devc * devc) +{ + int loopc; + + DEB(printk("Entered sb_dsp_reset()\n")); + + if (devc->model == MDL_ESS) return ess_dsp_reset (devc); + + /* This is only for non-ESS chips */ + + outb(1, DSP_RESET); + + udelay(10); + outb(0, DSP_RESET); + udelay(30); + + for (loopc = 0; loopc < 1000 && !(inb(DSP_DATA_AVAIL) & 0x80); loopc++); + + if (inb(DSP_READ) != 0xAA) + { + DDB(printk("sb: No response to RESET\n")); + return 0; /* Sorry */ + } + + DEB(printk("sb_dsp_reset() OK\n")); + + return 1; +} + +static void dsp_get_vers(sb_devc * devc) +{ + int i; + + unsigned long flags; + + DDB(printk("Entered dsp_get_vers()\n")); + spin_lock_irqsave(&devc->lock, flags); + devc->major = devc->minor = 0; + sb_dsp_command(devc, 0xe1); /* Get version */ + + for (i = 100000; i; i--) + { + if (inb(DSP_DATA_AVAIL) & 0x80) + { + if (devc->major == 0) + devc->major = inb(DSP_READ); + else + { + devc->minor = inb(DSP_READ); + break; + } + } + } + spin_unlock_irqrestore(&devc->lock, flags); + DDB(printk("DSP version %d.%02d\n", devc->major, devc->minor)); +} + +static int sb16_set_dma_hw(sb_devc * devc) +{ + int bits; + + if (devc->dma8 != 0 && devc->dma8 != 1 && devc->dma8 != 3) + { + printk(KERN_ERR "SB16: Invalid 8 bit DMA (%d)\n", devc->dma8); + return 0; + } + bits = (1 << devc->dma8); + + if (devc->dma16 >= 5 && devc->dma16 <= 7) + bits |= (1 << devc->dma16); + + sb_setmixer(devc, DMA_NR, bits); + return 1; +} + +static void sb16_set_mpu_port(sb_devc * devc, struct address_info *hw_config) +{ + /* + * This routine initializes new MIDI port setup register of SB Vibra (CT2502). + */ + unsigned char bits = sb_getmixer(devc, 0x84) & ~0x06; + + switch (hw_config->io_base) + { + case 0x300: + sb_setmixer(devc, 0x84, bits | 0x04); + break; + + case 0x330: + sb_setmixer(devc, 0x84, bits | 0x00); + break; + + default: + sb_setmixer(devc, 0x84, bits | 0x02); /* Disable MPU */ + printk(KERN_ERR "SB16: Invalid MIDI I/O port %x\n", hw_config->io_base); + } +} + +static int sb16_set_irq_hw(sb_devc * devc, int level) +{ + int ival; + + switch (level) + { + case 5: + ival = 2; + break; + case 7: + ival = 4; + break; + case 9: + ival = 1; + break; + case 10: + ival = 8; + break; + default: + printk(KERN_ERR "SB16: Invalid IRQ%d\n", level); + return 0; + } + sb_setmixer(devc, IRQ_NR, ival); + return 1; +} + +static void relocate_Jazz16(sb_devc * devc, struct address_info *hw_config) +{ + unsigned char bits = 0; + unsigned long flags; + + if (jazz16_base != 0 && jazz16_base != hw_config->io_base) + return; + + switch (hw_config->io_base) + { + case 0x220: + bits = 1; + break; + case 0x240: + bits = 2; + break; + case 0x260: + bits = 3; + break; + default: + return; + } + bits = jazz16_bits = bits << 5; + jazz16_base = hw_config->io_base; + + /* + * Magic wake up sequence by writing to 0x201 (aka Joystick port) + */ + spin_lock_irqsave(&jazz16_lock, flags); + outb((0xAF), 0x201); + outb((0x50), 0x201); + outb((bits), 0x201); + spin_unlock_irqrestore(&jazz16_lock, flags); +} + +static int init_Jazz16(sb_devc * devc, struct address_info *hw_config) +{ + char name[100]; + /* + * First try to check that the card has Jazz16 chip. It identifies itself + * by returning 0x12 as response to DSP command 0xfa. + */ + + if (!sb_dsp_command(devc, 0xfa)) + return 0; + + if (sb_dsp_get_byte(devc) != 0x12) + return 0; + + /* + * OK so far. Now configure the IRQ and DMA channel used by the card. + */ + if (hw_config->irq < 1 || hw_config->irq > 15 || jazz_irq_bits[hw_config->irq] == 0) + { + printk(KERN_ERR "Jazz16: Invalid interrupt (IRQ%d)\n", hw_config->irq); + return 0; + } + if (hw_config->dma < 0 || hw_config->dma > 3 || jazz_dma_bits[hw_config->dma] == 0) + { + printk(KERN_ERR "Jazz16: Invalid 8 bit DMA (DMA%d)\n", hw_config->dma); + return 0; + } + if (hw_config->dma2 < 0) + { + printk(KERN_ERR "Jazz16: No 16 bit DMA channel defined\n"); + return 0; + } + if (hw_config->dma2 < 5 || hw_config->dma2 > 7 || jazz_dma_bits[hw_config->dma2] == 0) + { + printk(KERN_ERR "Jazz16: Invalid 16 bit DMA (DMA%d)\n", hw_config->dma2); + return 0; + } + devc->dma16 = hw_config->dma2; + + if (!sb_dsp_command(devc, 0xfb)) + return 0; + + if (!sb_dsp_command(devc, jazz_dma_bits[hw_config->dma] | + (jazz_dma_bits[hw_config->dma2] << 4))) + return 0; + + if (!sb_dsp_command(devc, jazz_irq_bits[hw_config->irq])) + return 0; + + /* + * Now we have configured a standard Jazz16 device. + */ + devc->model = MDL_JAZZ; + strcpy(name, "Jazz16"); + + hw_config->name = "Jazz16"; + devc->caps |= SB_NO_MIDI; + return 1; +} + +static void relocate_ess1688(sb_devc * devc) +{ + unsigned char bits; + + switch (devc->base) + { + case 0x220: + bits = 0x04; + break; + case 0x230: + bits = 0x05; + break; + case 0x240: + bits = 0x06; + break; + case 0x250: + bits = 0x07; + break; + default: + return; /* Wrong port */ + } + + DDB(printk("Doing ESS1688 address selection\n")); + + /* + * ES1688 supports two alternative ways for software address config. + * First try the so called Read-Sequence-Key method. + */ + + /* Reset the sequence logic */ + inb(0x229); + inb(0x229); + inb(0x229); + + /* Perform the read sequence */ + inb(0x22b); + inb(0x229); + inb(0x22b); + inb(0x229); + inb(0x229); + inb(0x22b); + inb(0x229); + + /* Select the base address by reading from it. Then probe using the port. */ + inb(devc->base); + if (sb_dsp_reset(devc)) /* Bingo */ + return; + +#if 0 /* This causes system lockups (Nokia 386/25 at least) */ + /* + * The last resort is the system control register method. + */ + + outb((0x00), 0xfb); /* 0xFB is the unlock register */ + outb((0x00), 0xe0); /* Select index 0 */ + outb((bits), 0xe1); /* Write the config bits */ + outb((0x00), 0xf9); /* 0xFB is the lock register */ +#endif +} + +int sb_dsp_detect(struct address_info *hw_config, int pci, int pciio, struct sb_module_options *sbmo) +{ + sb_devc sb_info; + sb_devc *devc = &sb_info; + + memset((char *) &sb_info, 0, sizeof(sb_info)); /* Zero everything */ + + /* Copy module options in place */ + if(sbmo) memcpy(&devc->sbmo, sbmo, sizeof(struct sb_module_options)); + + sb_info.my_mididev = -1; + sb_info.my_mixerdev = -1; + sb_info.dev = -1; + + /* + * Initialize variables + */ + + DDB(printk("sb_dsp_detect(%x) entered\n", hw_config->io_base)); + + spin_lock_init(&devc->lock); + devc->type = hw_config->card_subtype; + + devc->base = hw_config->io_base; + devc->irq = hw_config->irq; + devc->dma8 = hw_config->dma; + + devc->dma16 = -1; + devc->pcibase = pciio; + + if(pci == SB_PCI_ESSMAESTRO) + { + devc->model = MDL_ESSPCI; + devc->caps |= SB_PCI_IRQ; + hw_config->driver_use_1 |= SB_PCI_IRQ; + hw_config->card_subtype = MDL_ESSPCI; + } + + if(pci == SB_PCI_YAMAHA) + { + devc->model = MDL_YMPCI; + devc->caps |= SB_PCI_IRQ; + hw_config->driver_use_1 |= SB_PCI_IRQ; + hw_config->card_subtype = MDL_YMPCI; + + printk("Yamaha PCI mode.\n"); + } + + if (devc->sbmo.acer) + { + unsigned long flags; + + spin_lock_irqsave(&devc->lock, flags); + inb(devc->base + 0x09); + inb(devc->base + 0x09); + inb(devc->base + 0x09); + inb(devc->base + 0x0b); + inb(devc->base + 0x09); + inb(devc->base + 0x0b); + inb(devc->base + 0x09); + inb(devc->base + 0x09); + inb(devc->base + 0x0b); + inb(devc->base + 0x09); + inb(devc->base + 0x00); + spin_unlock_irqrestore(&devc->lock, flags); + } + /* + * Detect the device + */ + + if (sb_dsp_reset(devc)) + dsp_get_vers(devc); + else + devc->major = 0; + + if (devc->type == 0 || devc->type == MDL_JAZZ || devc->type == MDL_SMW) + if (devc->major == 0 || (devc->major == 3 && devc->minor == 1)) + relocate_Jazz16(devc, hw_config); + + if (devc->major == 0 && (devc->type == MDL_ESS || devc->type == 0)) + relocate_ess1688(devc); + + if (!sb_dsp_reset(devc)) + { + DDB(printk("SB reset failed\n")); +#ifdef MODULE + printk(KERN_INFO "sb: dsp reset failed.\n"); +#endif + return 0; + } + if (devc->major == 0) + dsp_get_vers(devc); + + if (devc->major == 3 && devc->minor == 1) + { + if (devc->type == MDL_AZTECH) /* SG Washington? */ + { + if (sb_dsp_command(devc, 0x09)) + if (sb_dsp_command(devc, 0x00)) /* Enter WSS mode */ + { + int i; + + /* Have some delay */ + for (i = 0; i < 10000; i++) + inb(DSP_DATA_AVAIL); + devc->caps = SB_NO_AUDIO | SB_NO_MIDI; /* Mixer only */ + devc->model = MDL_AZTECH; + } + } + } + + if(devc->type == MDL_ESSPCI) + devc->model = MDL_ESSPCI; + + if(devc->type == MDL_YMPCI) + { + printk("YMPCI selected\n"); + devc->model = MDL_YMPCI; + } + + /* + * Save device information for sb_dsp_init() + */ + + + detected_devc = kmalloc(sizeof(sb_devc), GFP_KERNEL); + if (detected_devc == NULL) + { + printk(KERN_ERR "sb: Can't allocate memory for device information\n"); + return 0; + } + memcpy(detected_devc, devc, sizeof(sb_devc)); + MDB(printk(KERN_INFO "SB %d.%02d detected OK (%x)\n", devc->major, devc->minor, hw_config->io_base)); + return 1; +} + +int sb_dsp_init(struct address_info *hw_config, struct module *owner) +{ + sb_devc *devc; + char name[100]; + extern int sb_be_quiet; + int mixer22, mixer30; + +/* + * Check if we had detected a SB device earlier + */ + DDB(printk("sb_dsp_init(%x) entered\n", hw_config->io_base)); + name[0] = 0; + + if (detected_devc == NULL) + { + MDB(printk("No detected device\n")); + return 0; + } + devc = detected_devc; + detected_devc = NULL; + + if (devc->base != hw_config->io_base) + { + DDB(printk("I/O port mismatch\n")); + release_region(devc->base, 16); + return 0; + } + /* + * Now continue initialization of the device + */ + + devc->caps = hw_config->driver_use_1; + + if (!((devc->caps & SB_NO_AUDIO) && (devc->caps & SB_NO_MIDI)) && hw_config->irq > 0) + { /* IRQ setup */ + + /* + * ESS PCI cards do shared PCI IRQ stuff. Since they + * will get shared PCI irq lines we must cope. + */ + + int i=(devc->caps&SB_PCI_IRQ)?IRQF_SHARED:0; + + if (request_irq(hw_config->irq, sbintr, i, "soundblaster", devc) < 0) + { + printk(KERN_ERR "SB: Can't allocate IRQ%d\n", hw_config->irq); + release_region(devc->base, 16); + return 0; + } + devc->irq_ok = 0; + + if (devc->major == 4) + if (!sb16_set_irq_hw(devc, devc->irq)) /* Unsupported IRQ */ + { + free_irq(devc->irq, devc); + release_region(devc->base, 16); + return 0; + } + if ((devc->type == 0 || devc->type == MDL_ESS) && + devc->major == 3 && devc->minor == 1) + { /* Handle various chipsets which claim they are SB Pro compatible */ + if ((devc->type != 0 && devc->type != MDL_ESS) || + !ess_init(devc, hw_config)) + { + if ((devc->type != 0 && devc->type != MDL_JAZZ && + devc->type != MDL_SMW) || !init_Jazz16(devc, hw_config)) + { + DDB(printk("This is a genuine SB Pro\n")); + } + } + } + if (devc->major == 4 && devc->minor <= 11 ) /* Won't work */ + devc->irq_ok = 1; + else + { + int n; + + for (n = 0; n < 3 && devc->irq_ok == 0; n++) + { + if (sb_dsp_command(devc, 0xf2)) /* Cause interrupt immediately */ + { + int i; + + for (i = 0; !devc->irq_ok && i < 10000; i++); + } + } + if (!devc->irq_ok) + printk(KERN_WARNING "sb: Interrupt test on IRQ%d failed - Probable IRQ conflict\n", devc->irq); + else + { + DDB(printk("IRQ test OK (IRQ%d)\n", devc->irq)); + } + } + } /* IRQ setup */ + + last_sb = devc; + + switch (devc->major) + { + case 1: /* SB 1.0 or 1.5 */ + devc->model = hw_config->card_subtype = MDL_SB1; + break; + + case 2: /* SB 2.x */ + if (devc->minor == 0) + devc->model = hw_config->card_subtype = MDL_SB2; + else + devc->model = hw_config->card_subtype = MDL_SB201; + break; + + case 3: /* SB Pro and most clones */ + switch (devc->model) { + case 0: + devc->model = hw_config->card_subtype = MDL_SBPRO; + if (hw_config->name == NULL) + hw_config->name = "Sound Blaster Pro (8 BIT ONLY)"; + break; + case MDL_ESS: + ess_dsp_init(devc, hw_config); + break; + } + break; + + case 4: + devc->model = hw_config->card_subtype = MDL_SB16; + /* + * ALS007 and ALS100 return DSP version 4.2 and have 2 post-reset !=0 + * registers at 0x3c and 0x4c (output ctrl registers on ALS007) whereas + * a "standard" SB16 doesn't have a register at 0x4c. ALS100 actively + * updates register 0x22 whenever 0x30 changes, as per the SB16 spec. + * Since ALS007 doesn't, this can be used to differentiate the 2 cards. + */ + if ((devc->minor == 2) && sb_getmixer(devc,0x3c) && sb_getmixer(devc,0x4c)) + { + mixer30 = sb_getmixer(devc,0x30); + sb_setmixer(devc,0x22,(mixer22=sb_getmixer(devc,0x22)) & 0x0f); + sb_setmixer(devc,0x30,0xff); + /* ALS100 will force 0x30 to 0xf8 like SB16; ALS007 will allow 0xff. */ + /* Register 0x22 & 0xf0 on ALS100 == 0xf0; on ALS007 it == 0x10. */ + if ((sb_getmixer(devc,0x30) != 0xff) || ((sb_getmixer(devc,0x22) & 0xf0) != 0x10)) + { + devc->submodel = SUBMDL_ALS100; + if (hw_config->name == NULL) + hw_config->name = "Sound Blaster 16 (ALS-100)"; + } + else + { + sb_setmixer(devc,0x3c,0x1f); /* Enable all inputs */ + sb_setmixer(devc,0x4c,0x1f); + sb_setmixer(devc,0x22,mixer22); /* Restore 0x22 to original value */ + devc->submodel = SUBMDL_ALS007; + if (hw_config->name == NULL) + hw_config->name = "Sound Blaster 16 (ALS-007)"; + } + sb_setmixer(devc,0x30,mixer30); + } + else if (hw_config->name == NULL) + hw_config->name = "Sound Blaster 16"; + + if (hw_config->dma2 == -1) + devc->dma16 = devc->dma8; + else if (hw_config->dma2 < 5 || hw_config->dma2 > 7) + { + printk(KERN_WARNING "SB16: Bad or missing 16 bit DMA channel\n"); + devc->dma16 = devc->dma8; + } + else + devc->dma16 = hw_config->dma2; + + if(!sb16_set_dma_hw(devc)) { + free_irq(devc->irq, devc); + release_region(hw_config->io_base, 16); + return 0; + } + + devc->caps |= SB_NO_MIDI; + } + + if (!(devc->caps & SB_NO_MIXER)) + if (devc->major == 3 || devc->major == 4) + sb_mixer_init(devc, owner); + + if (!(devc->caps & SB_NO_MIDI)) + sb_dsp_midi_init(devc, owner); + + if (hw_config->name == NULL) + hw_config->name = "Sound Blaster (8 BIT/MONO ONLY)"; + + sprintf(name, "%s (%d.%02d)", hw_config->name, devc->major, devc->minor); + conf_printf(name, hw_config); + + /* + * Assuming that a sound card is Sound Blaster (compatible) is the most common + * configuration error and the mother of all problems. Usually sound cards + * emulate SB Pro but in addition they have a 16 bit native mode which should be + * used in Unix. See Readme.cards for more information about configuring OSS/Free + * properly. + */ + if (devc->model <= MDL_SBPRO) + { + if (devc->major == 3 && devc->minor != 1) /* "True" SB Pro should have v3.1 (rare ones may have 3.2). */ + { + printk(KERN_INFO "This sound card may not be fully Sound Blaster Pro compatible.\n"); + printk(KERN_INFO "In many cases there is another way to configure OSS so that\n"); + printk(KERN_INFO "it works properly with OSS (for example in 16 bit mode).\n"); + printk(KERN_INFO "Please ignore this message if you _really_ have a SB Pro.\n"); + } + else if (!sb_be_quiet && devc->model == MDL_SBPRO) + { + printk(KERN_INFO "SB DSP version is just %d.%02d which means that your card is\n", devc->major, devc->minor); + printk(KERN_INFO "several years old (8 bit only device) or alternatively the sound driver\n"); + printk(KERN_INFO "is incorrectly configured.\n"); + } + } + hw_config->card_subtype = devc->model; + hw_config->slots[0]=devc->dev; + last_devc = devc; /* For SB MPU detection */ + + if (!(devc->caps & SB_NO_AUDIO) && devc->dma8 >= 0) + { + if (sound_alloc_dma(devc->dma8, "SoundBlaster8")) + { + printk(KERN_WARNING "Sound Blaster: Can't allocate 8 bit DMA channel %d\n", devc->dma8); + } + if (devc->dma16 >= 0 && devc->dma16 != devc->dma8) + { + if (sound_alloc_dma(devc->dma16, "SoundBlaster16")) + printk(KERN_WARNING "Sound Blaster: can't allocate 16 bit DMA channel %d.\n", devc->dma16); + } + sb_audio_init(devc, name, owner); + hw_config->slots[0]=devc->dev; + } + else + { + MDB(printk("Sound Blaster: no audio devices found.\n")); + } + return 1; +} + +/* if (sbmpu) below we allow mpu401 to manage the midi devs + otherwise we have to unload them. (Andrzej Krzysztofowicz) */ + +void sb_dsp_unload(struct address_info *hw_config, int sbmpu) +{ + sb_devc *devc; + + devc = audio_devs[hw_config->slots[0]]->devc; + + if (devc && devc->base == hw_config->io_base) + { + if ((devc->model & MDL_ESS) && devc->pcibase) + release_region(devc->pcibase, 8); + + release_region(devc->base, 16); + + if (!(devc->caps & SB_NO_AUDIO)) + { + sound_free_dma(devc->dma8); + if (devc->dma16 >= 0) + sound_free_dma(devc->dma16); + } + if (!(devc->caps & SB_NO_AUDIO && devc->caps & SB_NO_MIDI)) + { + if (devc->irq > 0) + free_irq(devc->irq, devc); + + sb_mixer_unload(devc); + /* We don't have to do this bit any more the UART401 is its own + master -- Krzysztof Halasa */ + /* But we have to do it, if UART401 is not detected */ + if (!sbmpu) + sound_unload_mididev(devc->my_mididev); + sound_unload_audiodev(devc->dev); + } + kfree(devc); + } + else + release_region(hw_config->io_base, 16); + + kfree(detected_devc); +} + +/* + * Mixer access routines + * + * ES1887 modifications: some mixer registers reside in the + * range above 0xa0. These must be accessed in another way. + */ + +void sb_setmixer(sb_devc * devc, unsigned int port, unsigned int value) +{ + unsigned long flags; + + if (devc->model == MDL_ESS) { + ess_setmixer (devc, port, value); + return; + } + + spin_lock_irqsave(&devc->lock, flags); + + outb(((unsigned char) (port & 0xff)), MIXER_ADDR); + udelay(20); + outb(((unsigned char) (value & 0xff)), MIXER_DATA); + udelay(20); + + spin_unlock_irqrestore(&devc->lock, flags); +} + +unsigned int sb_getmixer(sb_devc * devc, unsigned int port) +{ + unsigned int val; + unsigned long flags; + + if (devc->model == MDL_ESS) return ess_getmixer (devc, port); + + spin_lock_irqsave(&devc->lock, flags); + + outb(((unsigned char) (port & 0xff)), MIXER_ADDR); + udelay(20); + val = inb(MIXER_DATA); + udelay(20); + + spin_unlock_irqrestore(&devc->lock, flags); + + return val; +} + +void sb_chgmixer + (sb_devc * devc, unsigned int reg, unsigned int mask, unsigned int val) +{ + int value; + + value = sb_getmixer(devc, reg); + value = (value & ~mask) | (val & mask); + sb_setmixer(devc, reg, value); +} + +/* + * MPU401 MIDI initialization. + */ + +static void smw_putmem(sb_devc * devc, int base, int addr, unsigned char val) +{ + unsigned long flags; + + spin_lock_irqsave(&jazz16_lock, flags); /* NOT the SB card? */ + + outb((addr & 0xff), base + 1); /* Low address bits */ + outb((addr >> 8), base + 2); /* High address bits */ + outb((val), base); /* Data */ + + spin_unlock_irqrestore(&jazz16_lock, flags); +} + +static unsigned char smw_getmem(sb_devc * devc, int base, int addr) +{ + unsigned long flags; + unsigned char val; + + spin_lock_irqsave(&jazz16_lock, flags); /* NOT the SB card? */ + + outb((addr & 0xff), base + 1); /* Low address bits */ + outb((addr >> 8), base + 2); /* High address bits */ + val = inb(base); /* Data */ + + spin_unlock_irqrestore(&jazz16_lock, flags); + return val; +} + +static int smw_midi_init(sb_devc * devc, struct address_info *hw_config) +{ + int mpu_base = hw_config->io_base; + int mp_base = mpu_base + 4; /* Microcontroller base */ + int i; + unsigned char control; + + + /* + * Reset the microcontroller so that the RAM can be accessed + */ + + control = inb(mpu_base + 7); + outb((control | 3), mpu_base + 7); /* Set last two bits to 1 (?) */ + outb(((control & 0xfe) | 2), mpu_base + 7); /* xxxxxxx0 resets the mc */ + + mdelay(3); /* Wait at least 1ms */ + + outb((control & 0xfc), mpu_base + 7); /* xxxxxx00 enables RAM */ + + /* + * Detect microcontroller by probing the 8k RAM area + */ + smw_putmem(devc, mp_base, 0, 0x00); + smw_putmem(devc, mp_base, 1, 0xff); + udelay(10); + + if (smw_getmem(devc, mp_base, 0) != 0x00 || smw_getmem(devc, mp_base, 1) != 0xff) + { + DDB(printk("SM Wave: No microcontroller RAM detected (%02x, %02x)\n", smw_getmem(devc, mp_base, 0), smw_getmem(devc, mp_base, 1))); + return 0; /* No RAM */ + } + /* + * There is RAM so assume it's really a SM Wave + */ + + devc->model = MDL_SMW; + smw_mixer_init(devc); + +#ifdef MODULE + if (!smw_ucode) + { + smw_ucodeLen = mod_firmware_load("/etc/sound/midi0001.bin", (void *) &smw_ucode); + smw_free = smw_ucode; + } +#endif + if (smw_ucodeLen > 0) + { + if (smw_ucodeLen != 8192) + { + printk(KERN_ERR "SM Wave: Invalid microcode (MIDI0001.BIN) length\n"); + return 1; + } + /* + * Download microcode + */ + + for (i = 0; i < 8192; i++) + smw_putmem(devc, mp_base, i, smw_ucode[i]); + + /* + * Verify microcode + */ + + for (i = 0; i < 8192; i++) + if (smw_getmem(devc, mp_base, i) != smw_ucode[i]) + { + printk(KERN_ERR "SM Wave: Microcode verification failed\n"); + return 0; + } + } + control = 0; +#ifdef SMW_SCSI_IRQ + /* + * Set the SCSI interrupt (IRQ2/9, IRQ3 or IRQ10). The SCSI interrupt + * is disabled by default. + * + * FIXME - make this a module option + * + * BTW the Zilog 5380 SCSI controller is located at MPU base + 0x10. + */ + { + static unsigned char scsi_irq_bits[] = { + 0, 0, 3, 1, 0, 0, 0, 0, 0, 3, 2, 0, 0, 0, 0, 0 + }; + control |= scsi_irq_bits[SMW_SCSI_IRQ] << 6; + } +#endif + +#ifdef SMW_OPL4_ENABLE + /* + * Make the OPL4 chip visible on the PC bus at 0x380. + * + * There is no need to enable this feature since this driver + * doesn't support OPL4 yet. Also there is no RAM in SM Wave so + * enabling OPL4 is pretty useless. + */ + control |= 0x10; /* Uses IRQ12 if bit 0x20 == 0 */ + /* control |= 0x20; Uncomment this if you want to use IRQ7 */ +#endif + outb((control | 0x03), mpu_base + 7); /* xxxxxx11 restarts */ + hw_config->name = "SoundMan Wave"; + return 1; +} + +static int init_Jazz16_midi(sb_devc * devc, struct address_info *hw_config) +{ + int mpu_base = hw_config->io_base; + int sb_base = devc->base; + int irq = hw_config->irq; + + unsigned char bits = 0; + unsigned long flags; + + if (irq < 0) + irq *= -1; + + if (irq < 1 || irq > 15 || + jazz_irq_bits[irq] == 0) + { + printk(KERN_ERR "Jazz16: Invalid MIDI interrupt (IRQ%d)\n", irq); + return 0; + } + switch (sb_base) + { + case 0x220: + bits = 1; + break; + case 0x240: + bits = 2; + break; + case 0x260: + bits = 3; + break; + default: + return 0; + } + bits = jazz16_bits = bits << 5; + switch (mpu_base) + { + case 0x310: + bits |= 1; + break; + case 0x320: + bits |= 2; + break; + case 0x330: + bits |= 3; + break; + default: + printk(KERN_ERR "Jazz16: Invalid MIDI I/O port %x\n", mpu_base); + return 0; + } + /* + * Magic wake up sequence by writing to 0x201 (aka Joystick port) + */ + spin_lock_irqsave(&jazz16_lock, flags); + outb(0xAF, 0x201); + outb(0x50, 0x201); + outb(bits, 0x201); + spin_unlock_irqrestore(&jazz16_lock, flags); + + hw_config->name = "Jazz16"; + smw_midi_init(devc, hw_config); + + if (!sb_dsp_command(devc, 0xfb)) + return 0; + + if (!sb_dsp_command(devc, jazz_dma_bits[devc->dma8] | + (jazz_dma_bits[devc->dma16] << 4))) + return 0; + + if (!sb_dsp_command(devc, jazz_irq_bits[devc->irq] | + (jazz_irq_bits[irq] << 4))) + return 0; + + return 1; +} + +int probe_sbmpu(struct address_info *hw_config, struct module *owner) +{ + sb_devc *devc = last_devc; + int ret; + + if (last_devc == NULL) + return 0; + + last_devc = NULL; + + if (hw_config->io_base <= 0) + { + /* The real vibra16 is fine about this, but we have to go + wipe up after Cyrix again */ + + if(devc->model == MDL_SB16 && devc->minor >= 12) + { + unsigned char bits = sb_getmixer(devc, 0x84) & ~0x06; + sb_setmixer(devc, 0x84, bits | 0x02); /* Disable MPU */ + } + return 0; + } + +#if defined(CONFIG_SOUND_MPU401) + if (devc->model == MDL_ESS) + { + struct resource *ports; + ports = request_region(hw_config->io_base, 2, "mpu401"); + if (!ports) { + printk(KERN_ERR "sbmpu: I/O port conflict (%x)\n", hw_config->io_base); + return 0; + } + if (!ess_midi_init(devc, hw_config)) { + release_region(hw_config->io_base, 2); + return 0; + } + hw_config->name = "ESS1xxx MPU"; + devc->midi_irq_cookie = NULL; + if (!probe_mpu401(hw_config, ports)) { + release_region(hw_config->io_base, 2); + return 0; + } + attach_mpu401(hw_config, owner); + if (last_sb->irq == -hw_config->irq) + last_sb->midi_irq_cookie = + (void *)(long) hw_config->slots[1]; + return 1; + } +#endif + + switch (devc->model) + { + case MDL_SB16: + if (hw_config->io_base != 0x300 && hw_config->io_base != 0x330) + { + printk(KERN_ERR "SB16: Invalid MIDI port %x\n", hw_config->io_base); + return 0; + } + hw_config->name = "Sound Blaster 16"; + if (hw_config->irq < 3 || hw_config->irq == devc->irq) + hw_config->irq = -devc->irq; + if (devc->minor > 12) /* What is Vibra's version??? */ + sb16_set_mpu_port(devc, hw_config); + break; + + case MDL_JAZZ: + if (hw_config->irq < 3 || hw_config->irq == devc->irq) + hw_config->irq = -devc->irq; + if (!init_Jazz16_midi(devc, hw_config)) + return 0; + break; + + case MDL_YMPCI: + hw_config->name = "Yamaha PCI Legacy"; + printk("Yamaha PCI legacy UART401 check.\n"); + break; + default: + return 0; + } + + ret = probe_uart401(hw_config, owner); + if (ret) + last_sb->midi_irq_cookie=midi_devs[hw_config->slots[4]]->devc; + return ret; +} + +void unload_sbmpu(struct address_info *hw_config) +{ +#if defined(CONFIG_SOUND_MPU401) + if (!strcmp (hw_config->name, "ESS1xxx MPU")) { + unload_mpu401(hw_config); + return; + } +#endif + unload_uart401(hw_config); +} + +EXPORT_SYMBOL(sb_dsp_init); +EXPORT_SYMBOL(sb_dsp_detect); +EXPORT_SYMBOL(sb_dsp_unload); +EXPORT_SYMBOL(sb_be_quiet); +EXPORT_SYMBOL(probe_sbmpu); +EXPORT_SYMBOL(unload_sbmpu); +EXPORT_SYMBOL(smw_free); +MODULE_LICENSE("GPL"); diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c new file mode 100644 index 00000000..5c773dff --- /dev/null +++ b/sound/oss/sb_ess.c @@ -0,0 +1,1831 @@ +#undef FKS_LOGGING +#undef FKS_TEST + +/* + * tabs should be 4 spaces, in vi(m): set tabstop=4 + * + * TODO: consistency speed calculations!! + * cleanup! + * ????: Did I break MIDI support? + * + * History: + * + * Rolf Fokkens (Dec 20 1998): ES188x recording level support on a per + * fokkensr@vertis.nl input basis. + * (Dec 24 1998): Recognition of ES1788, ES1887, ES1888, + * ES1868, ES1869 and ES1878. Could be used for + * specific handling in the future. All except + * ES1887 and ES1888 and ES688 are handled like + * ES1688. + * (Dec 27 1998): RECLEV for all (?) ES1688+ chips. ES188x now + * have the "Dec 20" support + RECLEV + * (Jan 2 1999): Preparation for Full Duplex. This means + * Audio 2 is now used for playback when dma16 + * is specified. The next step would be to use + * Audio 1 and Audio 2 at the same time. + * (Jan 9 1999): Put all ESS stuff into sb_ess.[ch], this + * includes both the ESS stuff that has been in + * sb_*[ch] before I touched it and the ESS support + * I added later + * (Jan 23 1999): Full Duplex seems to work. I wrote a small + * test proggy which works OK. Haven't found + * any applications to test it though. So why did + * I bother to create it anyway?? :) Just for + * fun. + * (May 2 1999): I tried to be too smart by "introducing" + * ess_calc_best_speed (). The idea was that two + * dividers could be used to setup a samplerate, + * ess_calc_best_speed () would choose the best. + * This works for playback, but results in + * recording problems for high samplerates. I + * fixed this by removing ess_calc_best_speed () + * and just doing what the documentation says. + * Andy Sloane (Jun 4 1999): Stole some code from ALSA to fix the playback + * andy@guildsoftware.com speed on ES1869, ES1879, ES1887, and ES1888. + * 1879's were previously ignored by this driver; + * added (untested) support for those. + * Cvetan Ivanov (Oct 27 1999): Fixed ess_dsp_init to call ess_set_dma_hw for + * zezo@inet.bg _ALL_ ESS models, not only ES1887 + * + * This files contains ESS chip specifics. It's based on the existing ESS + * handling as it resided in sb_common.c, sb_mixer.c and sb_audio.c. This + * file adds features like: + * - Chip Identification (as shown in /proc/sound) + * - RECLEV support for ES1688 and later + * - 6 bits playback level support chips later than ES1688 + * - Recording level support on a per-device basis for ES1887 + * - Full-Duplex for ES1887 + * + * Full duplex is enabled by specifying dma16. While the normal dma must + * be one of 0, 1 or 3, dma16 can be one of 0, 1, 3 or 5. DMA 5 is a 16 bit + * DMA channel, while the others are 8 bit.. + * + * ESS detection isn't full proof (yet). If it fails an additional module + * parameter esstype can be specified to be one of the following: + * -1, 0, 688, 1688, 1868, 1869, 1788, 1887, 1888 + * -1 means: mimic 2.0 behaviour, + * 0 means: auto detect. + * others: explicitly specify chip + * -1 is default, cause auto detect still doesn't work. + */ + +/* + * About the documentation + * + * I don't know if the chips all are OK, but the documentation is buggy. 'cause + * I don't have all the cips myself, there's a lot I cannot verify. I'll try to + * keep track of my latest insights about his here. If you have additional info, + * please enlighten me (fokkensr@vertis.nl)! + * + * I had the impression that ES1688 also has 6 bit master volume control. The + * documentation about ES1888 (rev C, october '95) claims that ES1888 has + * the following features ES1688 doesn't have: + * - 6 bit master volume + * - Full Duplex + * So ES1688 apparently doesn't have 6 bit master volume control, but the + * ES1688 does have RECLEV control. Makes me wonder: does ES688 have it too? + * Without RECLEV ES688 won't be much fun I guess. + * + * From the ES1888 (rev C, october '95) documentation I got the impression + * that registers 0x68 to 0x6e don't exist which means: no recording volume + * controls. To my surprise the ES888 documentation (1/14/96) claims that + * ES888 does have these record mixer registers, but that ES1888 doesn't have + * 0x69 and 0x6b. So the rest should be there. + * + * I'm trying to get ES1887 Full Duplex. Audio 2 is playback only, while Audio 2 + * is both record and playback. I think I should use Audio 2 for all playback. + * + * The documentation is an adventure: it's close but not fully accurate. I + * found out that after a reset some registers are *NOT* reset, though the + * docs say the would be. Interesting ones are 0x7f, 0x7d and 0x7a. They are + * related to the Audio 2 channel. I also was surprised about the consequences + * of writing 0x00 to 0x7f (which should be done by reset): The ES1887 moves + * into ES1888 mode. This means that it claims IRQ 11, which happens to be my + * ISDN adapter. Needless to say it no longer worked. I now understand why + * after rebooting 0x7f already was 0x05, the value of my choice: the BIOS + * did it. + * + * Oh, and this is another trap: in ES1887 docs mixer register 0x70 is + * described as if it's exactly the same as register 0xa1. This is *NOT* true. + * The description of 0x70 in ES1869 docs is accurate however. + * Well, the assumption about ES1869 was wrong: register 0x70 is very much + * like register 0xa1, except that bit 7 is always 1, whatever you want + * it to be. + * + * When using audio 2 mixer register 0x72 seems te be meaningless. Only 0xa2 + * has effect. + * + * Software reset not being able to reset all registers is great! Especially + * the fact that register 0x78 isn't reset is great when you wanna change back + * to single dma operation (simplex): audio 2 is still operational, and uses + * the same dma as audio 1: your ess changes into a funny echo machine. + * + * Received the news that ES1688 is detected as a ES1788. Did some thinking: + * the ES1887 detection scheme suggests in step 2 to try if bit 3 of register + * 0x64 can be changed. This is inaccurate, first I inverted the * check: "If + * can be modified, it's a 1688", which lead to a correct detection + * of my ES1887. It resulted however in bad detection of 1688 (reported by mail) + * and 1868 (if no PnP detection first): they result in a 1788 being detected. + * I don't have docs on 1688, but I do have docs on 1868: The documentation is + * probably inaccurate in the fact that I should check bit 2, not bit 3. This + * is what I do now. + */ + +/* + * About recognition of ESS chips + * + * The distinction of ES688, ES1688, ES1788, ES1887 and ES1888 is described in + * a (preliminary ??) datasheet on ES1887. Its aim is to identify ES1887, but + * during detection the text claims that "this chip may be ..." when a step + * fails. This scheme is used to distinct between the above chips. + * It appears however that some PnP chips like ES1868 are recognized as ES1788 + * by the ES1887 detection scheme. These PnP chips can be detected in another + * way however: ES1868, ES1869 and ES1878 can be recognized (full proof I think) + * by repeatedly reading mixer register 0x40. This is done by ess_identify in + * sb_common.c. + * This results in the following detection steps: + * - distinct between ES688 and ES1688+ (as always done in this driver) + * if ES688 we're ready + * - try to detect ES1868, ES1869 or ES1878 + * if successful we're ready + * - try to detect ES1888, ES1887 or ES1788 + * if successful we're ready + * - Dunno. Must be 1688. Will do in general + * + * About RECLEV support: + * + * The existing ES1688 support didn't take care of the ES1688+ recording + * levels very well. Whenever a device was selected (recmask) for recording + * its recording level was loud, and it couldn't be changed. The fact that + * internal register 0xb4 could take care of RECLEV, didn't work meaning until + * its value was restored every time the chip was reset; this reset the + * value of 0xb4 too. I guess that's what 4front also had (have?) trouble with. + * + * About ES1887 support: + * + * The ES1887 has separate registers to control the recording levels, for all + * inputs. The ES1887 specific software makes these levels the same as their + * corresponding playback levels, unless recmask says they aren't recorded. In + * the latter case the recording volumes are 0. + * Now recording levels of inputs can be controlled, by changing the playback + * levels. Furthermore several devices can be recorded together (which is not + * possible with the ES1688). + * Besides the separate recording level control for each input, the common + * recording level can also be controlled by RECLEV as described above. + * + * Not only ES1887 have this recording mixer. I know the following from the + * documentation: + * ES688 no + * ES1688 no + * ES1868 no + * ES1869 yes + * ES1878 no + * ES1879 yes + * ES1888 no/yes Contradicting documentation; most recent: yes + * ES1946 yes This is a PCI chip; not handled by this driver + */ + +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/spinlock.h> + +#include "sound_config.h" +#include "sb_mixer.h" +#include "sb.h" + +#include "sb_ess.h" + +#define ESSTYPE_LIKE20 -1 /* Mimic 2.0 behaviour */ +#define ESSTYPE_DETECT 0 /* Mimic 2.0 behaviour */ + +#define SUBMDL_ES1788 0x10 /* Subtype ES1788 for specific handling */ +#define SUBMDL_ES1868 0x11 /* Subtype ES1868 for specific handling */ +#define SUBMDL_ES1869 0x12 /* Subtype ES1869 for specific handling */ +#define SUBMDL_ES1878 0x13 /* Subtype ES1878 for specific handling */ +#define SUBMDL_ES1879 0x16 /* ES1879 was initially forgotten */ +#define SUBMDL_ES1887 0x14 /* Subtype ES1887 for specific handling */ +#define SUBMDL_ES1888 0x15 /* Subtype ES1888 for specific handling */ + +#define SB_CAP_ES18XX_RATE 0x100 + +#define ES1688_CLOCK1 795444 /* 128 - div */ +#define ES1688_CLOCK2 397722 /* 256 - div */ +#define ES18XX_CLOCK1 793800 /* 128 - div */ +#define ES18XX_CLOCK2 768000 /* 256 - div */ + +#ifdef FKS_LOGGING +static void ess_show_mixerregs (sb_devc *devc); +#endif +static int ess_read (sb_devc * devc, unsigned char reg); +static int ess_write (sb_devc * devc, unsigned char reg, unsigned char data); +static void ess_chgmixer + (sb_devc * devc, unsigned int reg, unsigned int mask, unsigned int val); + +/**************************************************************************** + * * + * ESS audio * + * * + ****************************************************************************/ + +struct ess_command {short cmd; short data;}; + +/* + * Commands for initializing Audio 1 for input (record) + */ +static struct ess_command ess_i08m[] = /* input 8 bit mono */ + { {0xb7, 0x51}, {0xb7, 0xd0}, {-1, 0} }; +static struct ess_command ess_i16m[] = /* input 16 bit mono */ + { {0xb7, 0x71}, {0xb7, 0xf4}, {-1, 0} }; +static struct ess_command ess_i08s[] = /* input 8 bit stereo */ + { {0xb7, 0x51}, {0xb7, 0x98}, {-1, 0} }; +static struct ess_command ess_i16s[] = /* input 16 bit stereo */ + { {0xb7, 0x71}, {0xb7, 0xbc}, {-1, 0} }; + +static struct ess_command *ess_inp_cmds[] = + { ess_i08m, ess_i16m, ess_i08s, ess_i16s }; + + +/* + * Commands for initializing Audio 1 for output (playback) + */ +static struct ess_command ess_o08m[] = /* output 8 bit mono */ + { {0xb6, 0x80}, {0xb7, 0x51}, {0xb7, 0xd0}, {-1, 0} }; +static struct ess_command ess_o16m[] = /* output 16 bit mono */ + { {0xb6, 0x00}, {0xb7, 0x71}, {0xb7, 0xf4}, {-1, 0} }; +static struct ess_command ess_o08s[] = /* output 8 bit stereo */ + { {0xb6, 0x80}, {0xb7, 0x51}, {0xb7, 0x98}, {-1, 0} }; +static struct ess_command ess_o16s[] = /* output 16 bit stereo */ + { {0xb6, 0x00}, {0xb7, 0x71}, {0xb7, 0xbc}, {-1, 0} }; + +static struct ess_command *ess_out_cmds[] = + { ess_o08m, ess_o16m, ess_o08s, ess_o16s }; + +static void ess_exec_commands + (sb_devc *devc, struct ess_command *cmdtab[]) +{ + struct ess_command *cmd; + + cmd = cmdtab [ ((devc->channels != 1) << 1) + (devc->bits != AFMT_U8) ]; + + while (cmd->cmd != -1) { + ess_write (devc, cmd->cmd, cmd->data); + cmd++; + } +} + +static void ess_change + (sb_devc *devc, unsigned int reg, unsigned int mask, unsigned int val) +{ + int value; + + value = ess_read (devc, reg); + value = (value & ~mask) | (val & mask); + ess_write (devc, reg, value); +} + +static void ess_set_output_parms + (int dev, unsigned long buf, int nr_bytes, int intrflag) +{ + sb_devc *devc = audio_devs[dev]->devc; + + if (devc->duplex) { + devc->trg_buf_16 = buf; + devc->trg_bytes_16 = nr_bytes; + devc->trg_intrflag_16 = intrflag; + devc->irq_mode_16 = IMODE_OUTPUT; + } else { + devc->trg_buf = buf; + devc->trg_bytes = nr_bytes; + devc->trg_intrflag = intrflag; + devc->irq_mode = IMODE_OUTPUT; + } +} + +static void ess_set_input_parms + (int dev, unsigned long buf, int count, int intrflag) +{ + sb_devc *devc = audio_devs[dev]->devc; + + devc->trg_buf = buf; + devc->trg_bytes = count; + devc->trg_intrflag = intrflag; + devc->irq_mode = IMODE_INPUT; +} + +static int ess_calc_div (int clock, int revert, int *speedp, int *diffp) +{ + int divider; + int speed, diff; + int retval; + + speed = *speedp; + divider = (clock + speed / 2) / speed; + retval = revert - divider; + if (retval > revert - 1) { + retval = revert - 1; + divider = revert - retval; + } + /* This line is suggested. Must be wrong I think + *speedp = (clock + divider / 2) / divider; + So I chose the next one */ + + *speedp = clock / divider; + diff = speed - *speedp; + if (diff < 0) diff =-diff; + *diffp = diff; + + return retval; +} + +static int ess_calc_best_speed + (int clock1, int rev1, int clock2, int rev2, int *divp, int *speedp) +{ + int speed1 = *speedp, speed2 = *speedp; + int div1, div2; + int diff1, diff2; + int retval; + + div1 = ess_calc_div (clock1, rev1, &speed1, &diff1); + div2 = ess_calc_div (clock2, rev2, &speed2, &diff2); + + if (diff1 < diff2) { + *divp = div1; + *speedp = speed1; + retval = 1; + } else { + /* *divp = div2; */ + *divp = 0x80 | div2; + *speedp = speed2; + retval = 2; + } + + return retval; +} + +/* + * Depending on the audiochannel ESS devices can + * have different clock settings. These are made consistent for duplex + * however. + * callers of ess_speed only do an audionum suggestion, which means + * input suggests 1, output suggests 2. This suggestion is only true + * however when doing duplex. + */ +static void ess_common_speed (sb_devc *devc, int *speedp, int *divp) +{ + int diff = 0, div; + + if (devc->duplex) { + /* + * The 0x80 is important for the first audio channel + */ + if (devc->submodel == SUBMDL_ES1888) { + div = 0x80 | ess_calc_div (795500, 256, speedp, &diff); + } else { + div = 0x80 | ess_calc_div (795500, 128, speedp, &diff); + } + } else if(devc->caps & SB_CAP_ES18XX_RATE) { + if (devc->submodel == SUBMDL_ES1888) { + ess_calc_best_speed(397700, 128, 795500, 256, + &div, speedp); + } else { + ess_calc_best_speed(ES18XX_CLOCK1, 128, ES18XX_CLOCK2, 256, + &div, speedp); + } + } else { + if (*speedp > 22000) { + div = 0x80 | ess_calc_div (ES1688_CLOCK1, 256, speedp, &diff); + } else { + div = 0x00 | ess_calc_div (ES1688_CLOCK2, 128, speedp, &diff); + } + } + *divp = div; +} + +static void ess_speed (sb_devc *devc, int audionum) +{ + int speed; + int div, div2; + + ess_common_speed (devc, &(devc->speed), &div); + +#ifdef FKS_REG_LOGGING +printk (KERN_INFO "FKS: ess_speed (%d) b speed = %d, div=%x\n", audionum, devc->speed, div); +#endif + + /* Set filter roll-off to 90% of speed/2 */ + speed = (devc->speed * 9) / 20; + + div2 = 256 - 7160000 / (speed * 82); + + if (!devc->duplex) audionum = 1; + + if (audionum == 1) { + /* Change behaviour of register A1 * + sb_chg_mixer(devc, 0x71, 0x20, 0x20) + * For ES1869 only??? */ + ess_write (devc, 0xa1, div); + ess_write (devc, 0xa2, div2); + } else { + ess_setmixer (devc, 0x70, div); + /* + * FKS: fascinating: 0x72 doesn't seem to work. + */ + ess_write (devc, 0xa2, div2); + ess_setmixer (devc, 0x72, div2); + } +} + +static int ess_audio_prepare_for_input(int dev, int bsize, int bcount) +{ + sb_devc *devc = audio_devs[dev]->devc; + + ess_speed(devc, 1); + + sb_dsp_command(devc, DSP_CMD_SPKOFF); + + ess_write (devc, 0xb8, 0x0e); /* Auto init DMA mode */ + ess_change (devc, 0xa8, 0x03, 3 - devc->channels); /* Mono/stereo */ + ess_write (devc, 0xb9, 2); /* Demand mode (4 bytes/DMA request) */ + + ess_exec_commands (devc, ess_inp_cmds); + + ess_change (devc, 0xb1, 0xf0, 0x50); + ess_change (devc, 0xb2, 0xf0, 0x50); + + devc->trigger_bits = 0; + return 0; +} + +static int ess_audio_prepare_for_output_audio1 (int dev, int bsize, int bcount) +{ + sb_devc *devc = audio_devs[dev]->devc; + + sb_dsp_reset(devc); + ess_speed(devc, 1); + ess_write (devc, 0xb8, 4); /* Auto init DMA mode */ + ess_change (devc, 0xa8, 0x03, 3 - devc->channels); /* Mono/stereo */ + ess_write (devc, 0xb9, 2); /* Demand mode (4 bytes/request) */ + + ess_exec_commands (devc, ess_out_cmds); + + ess_change (devc, 0xb1, 0xf0, 0x50); /* Enable DMA */ + ess_change (devc, 0xb2, 0xf0, 0x50); /* Enable IRQ */ + + sb_dsp_command(devc, DSP_CMD_SPKON); /* There be sound! */ + + devc->trigger_bits = 0; + return 0; +} + +static int ess_audio_prepare_for_output_audio2 (int dev, int bsize, int bcount) +{ + sb_devc *devc = audio_devs[dev]->devc; + unsigned char bits; + +/* FKS: qqq + sb_dsp_reset(devc); +*/ + + /* + * Auto-Initialize: + * DMA mode + demand mode (8 bytes/request, yes I want it all!) + * But leave 16-bit DMA bit untouched! + */ + ess_chgmixer (devc, 0x78, 0xd0, 0xd0); + + ess_speed(devc, 2); + + /* bits 4:3 on ES1887 represent recording source. Keep them! */ + bits = ess_getmixer (devc, 0x7a) & 0x18; + + /* Set stereo/mono */ + if (devc->channels != 1) bits |= 0x02; + + /* Init DACs; UNSIGNED mode for 8 bit; SIGNED mode for 16 bit */ + if (devc->bits != AFMT_U8) bits |= 0x05; /* 16 bit */ + + /* Enable DMA, IRQ will be shared (hopefully)*/ + bits |= 0x60; + + ess_setmixer (devc, 0x7a, bits); + + ess_mixer_reload (devc, SOUND_MIXER_PCM); /* There be sound! */ + + devc->trigger_bits = 0; + return 0; +} + +static int ess_audio_prepare_for_output(int dev, int bsize, int bcount) +{ + sb_devc *devc = audio_devs[dev]->devc; + +#ifdef FKS_REG_LOGGING +printk(KERN_INFO "ess_audio_prepare_for_output: dma_out=%d,dma_in=%d\n" +, audio_devs[dev]->dmap_out->dma, audio_devs[dev]->dmap_in->dma); +#endif + + if (devc->duplex) { + return ess_audio_prepare_for_output_audio2 (dev, bsize, bcount); + } else { + return ess_audio_prepare_for_output_audio1 (dev, bsize, bcount); + } +} + +static void ess_audio_halt_xfer(int dev) +{ + unsigned long flags; + sb_devc *devc = audio_devs[dev]->devc; + + spin_lock_irqsave(&devc->lock, flags); + sb_dsp_reset(devc); + spin_unlock_irqrestore(&devc->lock, flags); + + /* + * Audio 2 may still be operational! Creates awful sounds! + */ + if (devc->duplex) ess_chgmixer(devc, 0x78, 0x03, 0x00); +} + +static void ess_audio_start_input + (int dev, unsigned long buf, int nr_bytes, int intrflag) +{ + int count = nr_bytes; + sb_devc *devc = audio_devs[dev]->devc; + short c = -nr_bytes; + + /* + * Start a DMA input to the buffer pointed by dmaqtail + */ + + if (audio_devs[dev]->dmap_in->dma > 3) count >>= 1; + count--; + + devc->irq_mode = IMODE_INPUT; + + ess_write (devc, 0xa4, (unsigned char) ((unsigned short) c & 0xff)); + ess_write (devc, 0xa5, (unsigned char) (((unsigned short) c >> 8) & 0xff)); + + ess_change (devc, 0xb8, 0x0f, 0x0f); /* Go */ + devc->intr_active = 1; +} + +static void ess_audio_output_block_audio1 + (int dev, unsigned long buf, int nr_bytes, int intrflag) +{ + int count = nr_bytes; + sb_devc *devc = audio_devs[dev]->devc; + short c = -nr_bytes; + + if (audio_devs[dev]->dmap_out->dma > 3) + count >>= 1; + count--; + + devc->irq_mode = IMODE_OUTPUT; + + ess_write (devc, 0xa4, (unsigned char) ((unsigned short) c & 0xff)); + ess_write (devc, 0xa5, (unsigned char) (((unsigned short) c >> 8) & 0xff)); + + ess_change (devc, 0xb8, 0x05, 0x05); /* Go */ + devc->intr_active = 1; +} + +static void ess_audio_output_block_audio2 + (int dev, unsigned long buf, int nr_bytes, int intrflag) +{ + int count = nr_bytes; + sb_devc *devc = audio_devs[dev]->devc; + short c = -nr_bytes; + + if (audio_devs[dev]->dmap_out->dma > 3) count >>= 1; + count--; + + ess_setmixer (devc, 0x74, (unsigned char) ((unsigned short) c & 0xff)); + ess_setmixer (devc, 0x76, (unsigned char) (((unsigned short) c >> 8) & 0xff)); + ess_chgmixer (devc, 0x78, 0x03, 0x03); /* Go */ + + devc->irq_mode_16 = IMODE_OUTPUT; + devc->intr_active_16 = 1; +} + +static void ess_audio_output_block + (int dev, unsigned long buf, int nr_bytes, int intrflag) +{ + sb_devc *devc = audio_devs[dev]->devc; + + if (devc->duplex) { + ess_audio_output_block_audio2 (dev, buf, nr_bytes, intrflag); + } else { + ess_audio_output_block_audio1 (dev, buf, nr_bytes, intrflag); + } +} + +/* + * FKS: the if-statements for both bits and bits_16 are quite alike. + * Combine this... + */ +static void ess_audio_trigger(int dev, int bits) +{ + sb_devc *devc = audio_devs[dev]->devc; + + int bits_16 = bits & devc->irq_mode_16; + bits &= devc->irq_mode; + + if (!bits && !bits_16) { + /* FKS oh oh.... wrong?? for dma 16? */ + sb_dsp_command(devc, 0xd0); /* Halt DMA */ + } + + if (bits) { + switch (devc->irq_mode) + { + case IMODE_INPUT: + ess_audio_start_input(dev, devc->trg_buf, devc->trg_bytes, + devc->trg_intrflag); + break; + + case IMODE_OUTPUT: + ess_audio_output_block(dev, devc->trg_buf, devc->trg_bytes, + devc->trg_intrflag); + break; + } + } + + if (bits_16) { + switch (devc->irq_mode_16) { + case IMODE_INPUT: + ess_audio_start_input(dev, devc->trg_buf_16, devc->trg_bytes_16, + devc->trg_intrflag_16); + break; + + case IMODE_OUTPUT: + ess_audio_output_block(dev, devc->trg_buf_16, devc->trg_bytes_16, + devc->trg_intrflag_16); + break; + } + } + + devc->trigger_bits = bits | bits_16; +} + +static int ess_audio_set_speed(int dev, int speed) +{ + sb_devc *devc = audio_devs[dev]->devc; + int minspeed, maxspeed, dummydiv; + + if (speed > 0) { + minspeed = (devc->duplex ? 6215 : 5000 ); + maxspeed = (devc->duplex ? 44100 : 48000); + if (speed < minspeed) speed = minspeed; + if (speed > maxspeed) speed = maxspeed; + + ess_common_speed (devc, &speed, &dummydiv); + + devc->speed = speed; + } + return devc->speed; +} + +/* + * FKS: This is a one-on-one copy of sb1_audio_set_bits + */ +static unsigned int ess_audio_set_bits(int dev, unsigned int bits) +{ + sb_devc *devc = audio_devs[dev]->devc; + + if (bits != 0) { + if (bits == AFMT_U8 || bits == AFMT_S16_LE) { + devc->bits = bits; + } else { + devc->bits = AFMT_U8; + } + } + + return devc->bits; +} + +/* + * FKS: This is a one-on-one copy of sbpro_audio_set_channels + * (*) Modified it!! + */ +static short ess_audio_set_channels(int dev, short channels) +{ + sb_devc *devc = audio_devs[dev]->devc; + + if (channels == 1 || channels == 2) devc->channels = channels; + + return devc->channels; +} + +static struct audio_driver ess_audio_driver = /* ESS ES688/1688 */ +{ + .owner = THIS_MODULE, + .open = sb_audio_open, + .close = sb_audio_close, + .output_block = ess_set_output_parms, + .start_input = ess_set_input_parms, + .prepare_for_input = ess_audio_prepare_for_input, + .prepare_for_output = ess_audio_prepare_for_output, + .halt_io = ess_audio_halt_xfer, + .trigger = ess_audio_trigger, + .set_speed = ess_audio_set_speed, + .set_bits = ess_audio_set_bits, + .set_channels = ess_audio_set_channels +}; + +/* + * ess_audio_init must be called from sb_audio_init + */ +struct audio_driver *ess_audio_init + (sb_devc *devc, int *audio_flags, int *format_mask) +{ + *audio_flags = DMA_AUTOMODE; + *format_mask |= AFMT_S16_LE; + + if (devc->duplex) { + int tmp_dma; + /* + * sb_audio_init thinks dma8 is for playback and + * dma16 is for record. Not now! So swap them. + */ + tmp_dma = devc->dma16; + devc->dma16 = devc->dma8; + devc->dma8 = tmp_dma; + + *audio_flags |= DMA_DUPLEX; + } + + return &ess_audio_driver; +} + +/**************************************************************************** + * * + * ESS common * + * * + ****************************************************************************/ +static void ess_handle_channel + (char *channel, int dev, int intr_active, unsigned char flag, int irq_mode) +{ + if (!intr_active || !flag) return; +#ifdef FKS_REG_LOGGING +printk(KERN_INFO "FKS: ess_handle_channel %s irq_mode=%d\n", channel, irq_mode); +#endif + switch (irq_mode) { + case IMODE_OUTPUT: + DMAbuf_outputintr (dev, 1); + break; + + case IMODE_INPUT: + DMAbuf_inputintr (dev); + break; + + case IMODE_INIT: + break; + + default:; + /* printk(KERN_WARNING "ESS: Unexpected interrupt\n"); */ + } +} + +/* + * FKS: TODO!!! Finish this! + * + * I think midi stuff uses uart401, without interrupts. + * So IMODE_MIDI isn't a value for devc->irq_mode. + */ +void ess_intr (sb_devc *devc) +{ + int status; + unsigned char src; + + if (devc->submodel == SUBMDL_ES1887) { + src = ess_getmixer (devc, 0x7f) >> 4; + } else { + src = 0xff; + } + +#ifdef FKS_REG_LOGGING +printk(KERN_INFO "FKS: sbintr src=%x\n",(int)src); +#endif + ess_handle_channel + ( "Audio 1" + , devc->dev, devc->intr_active , src & 0x01, devc->irq_mode ); + ess_handle_channel + ( "Audio 2" + , devc->dev, devc->intr_active_16, src & 0x02, devc->irq_mode_16); + /* + * Acknowledge interrupts + */ + if (devc->submodel == SUBMDL_ES1887 && (src & 0x02)) { + ess_chgmixer (devc, 0x7a, 0x80, 0x00); + } + + if (src & 0x01) { + status = inb(DSP_DATA_AVAIL); + } +} + +static void ess_extended (sb_devc * devc) +{ + /* Enable extended mode */ + + sb_dsp_command(devc, 0xc6); +} + +static int ess_write (sb_devc * devc, unsigned char reg, unsigned char data) +{ +#ifdef FKS_REG_LOGGING +printk(KERN_INFO "FKS: write reg %x: %x\n", reg, data); +#endif + /* Write a byte to an extended mode register of ES1688 */ + + if (!sb_dsp_command(devc, reg)) + return 0; + + return sb_dsp_command(devc, data); +} + +static int ess_read (sb_devc * devc, unsigned char reg) +{ + /* Read a byte from an extended mode register of ES1688 */ + + /* Read register command */ + if (!sb_dsp_command(devc, 0xc0)) return -1; + + if (!sb_dsp_command(devc, reg )) return -1; + + return sb_dsp_get_byte(devc); +} + +int ess_dsp_reset(sb_devc * devc) +{ + int loopc; + +#ifdef FKS_REG_LOGGING +printk(KERN_INFO "FKS: ess_dsp_reset 1\n"); +ess_show_mixerregs (devc); +#endif + + DEB(printk("Entered ess_dsp_reset()\n")); + + outb(3, DSP_RESET); /* Reset FIFO too */ + + udelay(10); + outb(0, DSP_RESET); + udelay(30); + + for (loopc = 0; loopc < 1000 && !(inb(DSP_DATA_AVAIL) & 0x80); loopc++); + + if (inb(DSP_READ) != 0xAA) { + DDB(printk("sb: No response to RESET\n")); + return 0; /* Sorry */ + } + ess_extended (devc); + + DEB(printk("sb_dsp_reset() OK\n")); + +#ifdef FKS_LOGGING +printk(KERN_INFO "FKS: dsp_reset 2\n"); +ess_show_mixerregs (devc); +#endif + + return 1; +} + +static int ess_irq_bits (int irq) +{ + switch (irq) { + case 2: + case 9: + return 0; + + case 5: + return 1; + + case 7: + return 2; + + case 10: + return 3; + + default: + printk(KERN_ERR "ESS1688: Invalid IRQ %d\n", irq); + return -1; + } +} + +/* + * Set IRQ configuration register for all ESS models + */ +static int ess_common_set_irq_hw (sb_devc * devc) +{ + int irq_bits; + + if ((irq_bits = ess_irq_bits (devc->irq)) == -1) return 0; + + if (!ess_write (devc, 0xb1, 0x50 | (irq_bits << 2))) { + printk(KERN_ERR "ES1688: Failed to write to IRQ config register\n"); + return 0; + } + return 1; +} + +/* + * I wanna use modern ES1887 mixer irq handling. Funny is the + * fact that my BIOS wants the same. But suppose someone's BIOS + * doesn't do this! + * This is independent of duplex. If there's a 1887 this will + * prevent it from going into 1888 mode. + */ +static void ess_es1887_set_irq_hw (sb_devc * devc) +{ + int irq_bits; + + if ((irq_bits = ess_irq_bits (devc->irq)) == -1) return; + + ess_chgmixer (devc, 0x7f, 0x0f, 0x01 | ((irq_bits + 1) << 1)); +} + +static int ess_set_irq_hw (sb_devc * devc) +{ + if (devc->submodel == SUBMDL_ES1887) ess_es1887_set_irq_hw (devc); + + return ess_common_set_irq_hw (devc); +} + +#ifdef FKS_TEST + +/* + * FKS_test: + * for ES1887: 00, 18, non wr bits: 0001 1000 + * for ES1868: 00, b8, non wr bits: 1011 1000 + * for ES1888: 00, f8, non wr bits: 1111 1000 + * for ES1688: 00, f8, non wr bits: 1111 1000 + * + ES968 + */ + +static void FKS_test (sb_devc * devc) +{ + int val1, val2; + val1 = ess_getmixer (devc, 0x64); + ess_setmixer (devc, 0x64, ~val1); + val2 = ess_getmixer (devc, 0x64) ^ ~val1; + ess_setmixer (devc, 0x64, val1); + val1 ^= ess_getmixer (devc, 0x64); +printk (KERN_INFO "FKS: FKS_test %02x, %02x\n", (val1 & 0x0ff), (val2 & 0x0ff)); +}; +#endif + +static unsigned int ess_identify (sb_devc * devc) +{ + unsigned int val; + unsigned long flags; + + spin_lock_irqsave(&devc->lock, flags); + outb(((unsigned char) (0x40 & 0xff)), MIXER_ADDR); + + udelay(20); + val = inb(MIXER_DATA) << 8; + udelay(20); + val |= inb(MIXER_DATA); + udelay(20); + spin_unlock_irqrestore(&devc->lock, flags); + + return val; +} + +/* + * ESS technology describes a detection scheme in their docs. It involves + * fiddling with the bits in certain mixer registers. ess_probe is supposed + * to help. + * + * FKS: tracing shows ess_probe writes wrong value to 0x64. Bit 3 reads 1, but + * should be written 0 only. Check this. + */ +static int ess_probe (sb_devc * devc, int reg, int xorval) +{ + int val1, val2, val3; + + val1 = ess_getmixer (devc, reg); + val2 = val1 ^ xorval; + ess_setmixer (devc, reg, val2); + val3 = ess_getmixer (devc, reg); + ess_setmixer (devc, reg, val1); + + return (val2 == val3); +} + +int ess_init(sb_devc * devc, struct address_info *hw_config) +{ + unsigned char cfg; + int ess_major = 0, ess_minor = 0; + int i; + static char name[100], modelname[10]; + + /* + * Try to detect ESS chips. + */ + + sb_dsp_command(devc, 0xe7); /* Return identification */ + + for (i = 1000; i; i--) { + if (inb(DSP_DATA_AVAIL) & 0x80) { + if (ess_major == 0) { + ess_major = inb(DSP_READ); + } else { + ess_minor = inb(DSP_READ); + break; + } + } + } + + if (ess_major == 0) return 0; + + if (ess_major == 0x48 && (ess_minor & 0xf0) == 0x80) { + sprintf(name, "ESS ES488 AudioDrive (rev %d)", + ess_minor & 0x0f); + hw_config->name = name; + devc->model = MDL_SBPRO; + return 1; + } + + /* + * This the detection heuristic of ESS technology, though somewhat + * changed to actually make it work. + * This results in the following detection steps: + * - distinct between ES688 and ES1688+ (as always done in this driver) + * if ES688 we're ready + * - try to detect ES1868, ES1869 or ES1878 (ess_identify) + * if successful we're ready + * - try to detect ES1888, ES1887 or ES1788 (aim: detect ES1887) + * if successful we're ready + * - Dunno. Must be 1688. Will do in general + * + * This is the most BETA part of the software: Will the detection + * always work? + */ + devc->model = MDL_ESS; + devc->submodel = ess_minor & 0x0f; + + if (ess_major == 0x68 && (ess_minor & 0xf0) == 0x80) { + char *chip = NULL; + int submodel = -1; + + switch (devc->sbmo.esstype) { + case ESSTYPE_DETECT: + case ESSTYPE_LIKE20: + break; + case 688: + submodel = 0x00; + break; + case 1688: + submodel = 0x08; + break; + case 1868: + submodel = SUBMDL_ES1868; + break; + case 1869: + submodel = SUBMDL_ES1869; + break; + case 1788: + submodel = SUBMDL_ES1788; + break; + case 1878: + submodel = SUBMDL_ES1878; + break; + case 1879: + submodel = SUBMDL_ES1879; + break; + case 1887: + submodel = SUBMDL_ES1887; + break; + case 1888: + submodel = SUBMDL_ES1888; + break; + default: + printk (KERN_ERR "Invalid esstype=%d specified\n", devc->sbmo.esstype); + return 0; + }; + if (submodel != -1) { + devc->submodel = submodel; + sprintf (modelname, "ES%d", devc->sbmo.esstype); + chip = modelname; + }; + if (chip == NULL && (ess_minor & 0x0f) < 8) { + chip = "ES688"; + }; +#ifdef FKS_TEST +FKS_test (devc); +#endif + /* + * If Nothing detected yet, and we want 2.0 behaviour... + * Then let's assume it's ES1688. + */ + if (chip == NULL && devc->sbmo.esstype == ESSTYPE_LIKE20) { + chip = "ES1688"; + }; + + if (chip == NULL) { + int type; + + type = ess_identify (devc); + + switch (type) { + case 0x1868: + chip = "ES1868"; + devc->submodel = SUBMDL_ES1868; + break; + case 0x1869: + chip = "ES1869"; + devc->submodel = SUBMDL_ES1869; + break; + case 0x1878: + chip = "ES1878"; + devc->submodel = SUBMDL_ES1878; + break; + case 0x1879: + chip = "ES1879"; + devc->submodel = SUBMDL_ES1879; + break; + default: + if ((type & 0x00ff) != ((type >> 8) & 0x00ff)) { + printk ("ess_init: Unrecognized %04x\n", type); + } + }; + }; +#if 0 + /* + * this one failed: + * the probing of bit 4 is another thought: from ES1788 and up, all + * chips seem to have hardware volume control. Bit 4 is readonly to + * check if a hardware volume interrupt has fired. + * Cause ES688/ES1688 don't have this feature, bit 4 might be writeable + * for these chips. + */ + if (chip == NULL && !ess_probe(devc, 0x64, (1 << 4))) { +#endif + /* + * the probing of bit 2 is my idea. The ES1887 docs want me to probe + * bit 3. This results in ES1688 being detected as ES1788. + * Bit 2 is for "Enable HWV IRQE", but as ES(1)688 chips don't have + * HardWare Volume, I think they don't have this IRQE. + */ + if (chip == NULL && ess_probe(devc, 0x64, (1 << 2))) { + if (ess_probe (devc, 0x70, 0x7f)) { + if (ess_probe (devc, 0x64, (1 << 5))) { + chip = "ES1887"; + devc->submodel = SUBMDL_ES1887; + } else { + chip = "ES1888"; + devc->submodel = SUBMDL_ES1888; + } + } else { + chip = "ES1788"; + devc->submodel = SUBMDL_ES1788; + } + }; + if (chip == NULL) { + chip = "ES1688"; + }; + + printk ( KERN_INFO "ESS chip %s %s%s\n" + , chip + , ( devc->sbmo.esstype == ESSTYPE_DETECT || devc->sbmo.esstype == ESSTYPE_LIKE20 + ? "detected" + : "specified" + ) + , ( devc->sbmo.esstype == ESSTYPE_LIKE20 + ? " (kernel 2.0 compatible)" + : "" + ) + ); + + sprintf(name,"ESS %s AudioDrive (rev %d)", chip, ess_minor & 0x0f); + } else { + strcpy(name, "Jazz16"); + } + + /* AAS: info stolen from ALSA: these boards have different clocks */ + switch(devc->submodel) { +/* APPARENTLY NOT 1869 AND 1887 + case SUBMDL_ES1869: + case SUBMDL_ES1887: +*/ + case SUBMDL_ES1888: + devc->caps |= SB_CAP_ES18XX_RATE; + break; + } + + hw_config->name = name; + /* FKS: sb_dsp_reset to enable extended mode???? */ + sb_dsp_reset(devc); /* Turn on extended mode */ + + /* + * Enable joystick and OPL3 + */ + cfg = ess_getmixer (devc, 0x40); + ess_setmixer (devc, 0x40, cfg | 0x03); + if (devc->submodel >= 8) { /* ES1688 */ + devc->caps |= SB_NO_MIDI; /* ES1688 uses MPU401 MIDI mode */ + } + sb_dsp_reset (devc); + + /* + * This is important! If it's not done, the IRQ probe in sb_dsp_init + * may fail. + */ + return ess_set_irq_hw (devc); +} + +static int ess_set_dma_hw(sb_devc * devc) +{ + unsigned char cfg, dma_bits = 0, dma16_bits; + int dma; + +#ifdef FKS_LOGGING +printk(KERN_INFO "ess_set_dma_hw: dma8=%d,dma16=%d,dup=%d\n" +, devc->dma8, devc->dma16, devc->duplex); +#endif + + /* + * FKS: It seems as if this duplex flag isn't set yet. Check it. + */ + dma = devc->dma8; + + if (dma > 3 || dma < 0 || dma == 2) { + dma_bits = 0; + printk(KERN_ERR "ESS1688: Invalid DMA8 %d\n", dma); + return 0; + } else { + /* Extended mode DMA enable */ + cfg = 0x50; + + if (dma == 3) { + dma_bits = 3; + } else { + dma_bits = dma + 1; + } + } + + if (!ess_write (devc, 0xb2, cfg | (dma_bits << 2))) { + printk(KERN_ERR "ESS1688: Failed to write to DMA config register\n"); + return 0; + } + + if (devc->duplex) { + dma = devc->dma16; + dma16_bits = 0; + + if (dma >= 0) { + switch (dma) { + case 0: + dma_bits = 0x04; + break; + case 1: + dma_bits = 0x05; + break; + case 3: + dma_bits = 0x06; + break; + case 5: + dma_bits = 0x07; + dma16_bits = 0x20; + break; + default: + printk(KERN_ERR "ESS1887: Invalid DMA16 %d\n", dma); + return 0; + }; + ess_chgmixer (devc, 0x78, 0x20, dma16_bits); + ess_chgmixer (devc, 0x7d, 0x07, dma_bits); + } + } + return 1; +} + +/* + * This one is called from sb_dsp_init. + * + * Return values: + * 0: Failed + * 1: Succeeded or doesn't apply (not SUBMDL_ES1887) + */ +int ess_dsp_init (sb_devc *devc, struct address_info *hw_config) +{ + /* + * Caller also checks this, but anyway + */ + if (devc->model != MDL_ESS) { + printk (KERN_INFO "ess_dsp_init for non ESS chip\n"); + return 1; + } + /* + * This for ES1887 to run Full Duplex. Actually ES1888 + * is allowed to do so too. I have no idea yet if this + * will work for ES1888 however. + * + * For SB16 having both dma8 and dma16 means enable + * Full Duplex. Let's try this for ES1887 too + * + */ + if (devc->submodel == SUBMDL_ES1887) { + if (hw_config->dma2 != -1) { + devc->dma16 = hw_config->dma2; + } + /* + * devc->duplex initialization is put here, cause + * ess_set_dma_hw needs it. + */ + if (devc->dma8 != devc->dma16 && devc->dma16 != -1) { + devc->duplex = 1; + } + } + if (!ess_set_dma_hw (devc)) { + free_irq(devc->irq, devc); + return 0; + } + return 1; +} + +/**************************************************************************** + * * + * ESS mixer * + * * + ****************************************************************************/ + +#define ES688_RECORDING_DEVICES \ + ( SOUND_MASK_LINE | SOUND_MASK_MIC | SOUND_MASK_CD ) +#define ES688_MIXER_DEVICES \ + ( SOUND_MASK_SYNTH | SOUND_MASK_PCM | SOUND_MASK_LINE \ + | SOUND_MASK_MIC | SOUND_MASK_CD | SOUND_MASK_VOLUME \ + | SOUND_MASK_LINE2 | SOUND_MASK_SPEAKER ) + +#define ES1688_RECORDING_DEVICES \ + ( ES688_RECORDING_DEVICES ) +#define ES1688_MIXER_DEVICES \ + ( ES688_MIXER_DEVICES | SOUND_MASK_RECLEV ) + +#define ES1887_RECORDING_DEVICES \ + ( ES1688_RECORDING_DEVICES | SOUND_MASK_LINE2 | SOUND_MASK_SYNTH) +#define ES1887_MIXER_DEVICES \ + ( ES1688_MIXER_DEVICES ) + +/* + * Mixer registers of ES1887 + * + * These registers specifically take care of recording levels. To make the + * mapping from playback devices to recording devices every recording + * devices = playback device + ES_REC_MIXER_RECDIFF + */ +#define ES_REC_MIXER_RECBASE (SOUND_MIXER_LINE3 + 1) +#define ES_REC_MIXER_RECDIFF (ES_REC_MIXER_RECBASE - SOUND_MIXER_SYNTH) + +#define ES_REC_MIXER_RECSYNTH (SOUND_MIXER_SYNTH + ES_REC_MIXER_RECDIFF) +#define ES_REC_MIXER_RECPCM (SOUND_MIXER_PCM + ES_REC_MIXER_RECDIFF) +#define ES_REC_MIXER_RECSPEAKER (SOUND_MIXER_SPEAKER + ES_REC_MIXER_RECDIFF) +#define ES_REC_MIXER_RECLINE (SOUND_MIXER_LINE + ES_REC_MIXER_RECDIFF) +#define ES_REC_MIXER_RECMIC (SOUND_MIXER_MIC + ES_REC_MIXER_RECDIFF) +#define ES_REC_MIXER_RECCD (SOUND_MIXER_CD + ES_REC_MIXER_RECDIFF) +#define ES_REC_MIXER_RECIMIX (SOUND_MIXER_IMIX + ES_REC_MIXER_RECDIFF) +#define ES_REC_MIXER_RECALTPCM (SOUND_MIXER_ALTPCM + ES_REC_MIXER_RECDIFF) +#define ES_REC_MIXER_RECRECLEV (SOUND_MIXER_RECLEV + ES_REC_MIXER_RECDIFF) +#define ES_REC_MIXER_RECIGAIN (SOUND_MIXER_IGAIN + ES_REC_MIXER_RECDIFF) +#define ES_REC_MIXER_RECOGAIN (SOUND_MIXER_OGAIN + ES_REC_MIXER_RECDIFF) +#define ES_REC_MIXER_RECLINE1 (SOUND_MIXER_LINE1 + ES_REC_MIXER_RECDIFF) +#define ES_REC_MIXER_RECLINE2 (SOUND_MIXER_LINE2 + ES_REC_MIXER_RECDIFF) +#define ES_REC_MIXER_RECLINE3 (SOUND_MIXER_LINE3 + ES_REC_MIXER_RECDIFF) + +static mixer_tab es688_mix = { +MIX_ENT(SOUND_MIXER_VOLUME, 0x32, 7, 4, 0x32, 3, 4), +MIX_ENT(SOUND_MIXER_BASS, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_TREBLE, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_SYNTH, 0x36, 7, 4, 0x36, 3, 4), +MIX_ENT(SOUND_MIXER_PCM, 0x14, 7, 4, 0x14, 3, 4), +MIX_ENT(SOUND_MIXER_SPEAKER, 0x3c, 2, 3, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE, 0x3e, 7, 4, 0x3e, 3, 4), +MIX_ENT(SOUND_MIXER_MIC, 0x1a, 7, 4, 0x1a, 3, 4), +MIX_ENT(SOUND_MIXER_CD, 0x38, 7, 4, 0x38, 3, 4), +MIX_ENT(SOUND_MIXER_IMIX, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_RECLEV, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_IGAIN, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_OGAIN, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE1, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE2, 0x3a, 7, 4, 0x3a, 3, 4), +MIX_ENT(SOUND_MIXER_LINE3, 0x00, 0, 0, 0x00, 0, 0) +}; + +/* + * The ES1688 specifics... hopefully correct... + * - 6 bit master volume + * I was wrong, ES1888 docs say ES1688 didn't have it. + * - RECLEV control + * These may apply to ES688 too. I have no idea. + */ +static mixer_tab es1688_mix = { +MIX_ENT(SOUND_MIXER_VOLUME, 0x32, 7, 4, 0x32, 3, 4), +MIX_ENT(SOUND_MIXER_BASS, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_TREBLE, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_SYNTH, 0x36, 7, 4, 0x36, 3, 4), +MIX_ENT(SOUND_MIXER_PCM, 0x14, 7, 4, 0x14, 3, 4), +MIX_ENT(SOUND_MIXER_SPEAKER, 0x3c, 2, 3, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE, 0x3e, 7, 4, 0x3e, 3, 4), +MIX_ENT(SOUND_MIXER_MIC, 0x1a, 7, 4, 0x1a, 3, 4), +MIX_ENT(SOUND_MIXER_CD, 0x38, 7, 4, 0x38, 3, 4), +MIX_ENT(SOUND_MIXER_IMIX, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_RECLEV, 0xb4, 7, 4, 0xb4, 3, 4), +MIX_ENT(SOUND_MIXER_IGAIN, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_OGAIN, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE1, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE2, 0x3a, 7, 4, 0x3a, 3, 4), +MIX_ENT(SOUND_MIXER_LINE3, 0x00, 0, 0, 0x00, 0, 0) +}; + +static mixer_tab es1688later_mix = { +MIX_ENT(SOUND_MIXER_VOLUME, 0x60, 5, 6, 0x62, 5, 6), +MIX_ENT(SOUND_MIXER_BASS, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_TREBLE, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_SYNTH, 0x36, 7, 4, 0x36, 3, 4), +MIX_ENT(SOUND_MIXER_PCM, 0x14, 7, 4, 0x14, 3, 4), +MIX_ENT(SOUND_MIXER_SPEAKER, 0x3c, 2, 3, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE, 0x3e, 7, 4, 0x3e, 3, 4), +MIX_ENT(SOUND_MIXER_MIC, 0x1a, 7, 4, 0x1a, 3, 4), +MIX_ENT(SOUND_MIXER_CD, 0x38, 7, 4, 0x38, 3, 4), +MIX_ENT(SOUND_MIXER_IMIX, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_RECLEV, 0xb4, 7, 4, 0xb4, 3, 4), +MIX_ENT(SOUND_MIXER_IGAIN, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_OGAIN, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE1, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE2, 0x3a, 7, 4, 0x3a, 3, 4), +MIX_ENT(SOUND_MIXER_LINE3, 0x00, 0, 0, 0x00, 0, 0) +}; + +/* + * This one is for all ESS chips with a record mixer. + * It's not used (yet) however + */ +static mixer_tab es_rec_mix = { +MIX_ENT(SOUND_MIXER_VOLUME, 0x60, 5, 6, 0x62, 5, 6), +MIX_ENT(SOUND_MIXER_BASS, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_TREBLE, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_SYNTH, 0x36, 7, 4, 0x36, 3, 4), +MIX_ENT(SOUND_MIXER_PCM, 0x14, 7, 4, 0x14, 3, 4), +MIX_ENT(SOUND_MIXER_SPEAKER, 0x3c, 2, 3, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE, 0x3e, 7, 4, 0x3e, 3, 4), +MIX_ENT(SOUND_MIXER_MIC, 0x1a, 7, 4, 0x1a, 3, 4), +MIX_ENT(SOUND_MIXER_CD, 0x38, 7, 4, 0x38, 3, 4), +MIX_ENT(SOUND_MIXER_IMIX, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_RECLEV, 0xb4, 7, 4, 0xb4, 3, 4), +MIX_ENT(SOUND_MIXER_IGAIN, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_OGAIN, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE1, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE2, 0x3a, 7, 4, 0x3a, 3, 4), +MIX_ENT(SOUND_MIXER_LINE3, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECSYNTH, 0x6b, 7, 4, 0x6b, 3, 4), +MIX_ENT(ES_REC_MIXER_RECPCM, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECSPEAKER, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECLINE, 0x6e, 7, 4, 0x6e, 3, 4), +MIX_ENT(ES_REC_MIXER_RECMIC, 0x68, 7, 4, 0x68, 3, 4), +MIX_ENT(ES_REC_MIXER_RECCD, 0x6a, 7, 4, 0x6a, 3, 4), +MIX_ENT(ES_REC_MIXER_RECIMIX, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECALTPCM, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECRECLEV, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECIGAIN, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECOGAIN, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECLINE1, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECLINE2, 0x6c, 7, 4, 0x6c, 3, 4), +MIX_ENT(ES_REC_MIXER_RECLINE3, 0x00, 0, 0, 0x00, 0, 0) +}; + +/* + * This one is for ES1887. It's little different from es_rec_mix: it + * has 0x7c for PCM playback level. This is because ES1887 uses + * Audio 2 for playback. + */ +static mixer_tab es1887_mix = { +MIX_ENT(SOUND_MIXER_VOLUME, 0x60, 5, 6, 0x62, 5, 6), +MIX_ENT(SOUND_MIXER_BASS, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_TREBLE, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_SYNTH, 0x36, 7, 4, 0x36, 3, 4), +MIX_ENT(SOUND_MIXER_PCM, 0x7c, 7, 4, 0x7c, 3, 4), +MIX_ENT(SOUND_MIXER_SPEAKER, 0x3c, 2, 3, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE, 0x3e, 7, 4, 0x3e, 3, 4), +MIX_ENT(SOUND_MIXER_MIC, 0x1a, 7, 4, 0x1a, 3, 4), +MIX_ENT(SOUND_MIXER_CD, 0x38, 7, 4, 0x38, 3, 4), +MIX_ENT(SOUND_MIXER_IMIX, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_RECLEV, 0xb4, 7, 4, 0xb4, 3, 4), +MIX_ENT(SOUND_MIXER_IGAIN, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_OGAIN, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE1, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE2, 0x3a, 7, 4, 0x3a, 3, 4), +MIX_ENT(SOUND_MIXER_LINE3, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECSYNTH, 0x6b, 7, 4, 0x6b, 3, 4), +MIX_ENT(ES_REC_MIXER_RECPCM, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECSPEAKER, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECLINE, 0x6e, 7, 4, 0x6e, 3, 4), +MIX_ENT(ES_REC_MIXER_RECMIC, 0x68, 7, 4, 0x68, 3, 4), +MIX_ENT(ES_REC_MIXER_RECCD, 0x6a, 7, 4, 0x6a, 3, 4), +MIX_ENT(ES_REC_MIXER_RECIMIX, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECALTPCM, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECRECLEV, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECIGAIN, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECOGAIN, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECLINE1, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(ES_REC_MIXER_RECLINE2, 0x6c, 7, 4, 0x6c, 3, 4), +MIX_ENT(ES_REC_MIXER_RECLINE3, 0x00, 0, 0, 0x00, 0, 0) +}; + +static int ess_has_rec_mixer (int submodel) +{ + switch (submodel) { + case SUBMDL_ES1887: + return 1; + default: + return 0; + }; +}; + +#ifdef FKS_LOGGING +static int ess_mixer_mon_regs[] + = { 0x70, 0x71, 0x72, 0x74, 0x76, 0x78, 0x7a, 0x7c, 0x7d, 0x7f + , 0xa1, 0xa2, 0xa4, 0xa5, 0xa8, 0xa9 + , 0xb1, 0xb2, 0xb4, 0xb5, 0xb6, 0xb7, 0xb9 + , 0x00}; + +static void ess_show_mixerregs (sb_devc *devc) +{ + int *mp = ess_mixer_mon_regs; + +return; + + while (*mp != 0) { + printk (KERN_INFO "res (%x)=%x\n", *mp, (int)(ess_getmixer (devc, *mp))); + mp++; + } +} +#endif + +void ess_setmixer (sb_devc * devc, unsigned int port, unsigned int value) +{ + unsigned long flags; + +#ifdef FKS_LOGGING +printk(KERN_INFO "FKS: write mixer %x: %x\n", port, value); +#endif + + spin_lock_irqsave(&devc->lock, flags); + if (port >= 0xa0) { + ess_write (devc, port, value); + } else { + outb(((unsigned char) (port & 0xff)), MIXER_ADDR); + + udelay(20); + outb(((unsigned char) (value & 0xff)), MIXER_DATA); + udelay(20); + }; + spin_unlock_irqrestore(&devc->lock, flags); +} + +unsigned int ess_getmixer (sb_devc * devc, unsigned int port) +{ + unsigned int val; + unsigned long flags; + + spin_lock_irqsave(&devc->lock, flags); + + if (port >= 0xa0) { + val = ess_read (devc, port); + } else { + outb(((unsigned char) (port & 0xff)), MIXER_ADDR); + + udelay(20); + val = inb(MIXER_DATA); + udelay(20); + } + spin_unlock_irqrestore(&devc->lock, flags); + + return val; +} + +static void ess_chgmixer + (sb_devc * devc, unsigned int reg, unsigned int mask, unsigned int val) +{ + int value; + + value = ess_getmixer (devc, reg); + value = (value & ~mask) | (val & mask); + ess_setmixer (devc, reg, value); +} + +/* + * ess_mixer_init must be called from sb_mixer_init + */ +void ess_mixer_init (sb_devc * devc) +{ + devc->mixer_caps = SOUND_CAP_EXCL_INPUT; + + /* + * Take care of ES1887 specifics... + */ + switch (devc->submodel) { + case SUBMDL_ES1887: + devc->supported_devices = ES1887_MIXER_DEVICES; + devc->supported_rec_devices = ES1887_RECORDING_DEVICES; +#ifdef FKS_LOGGING +printk (KERN_INFO "FKS: ess_mixer_init dup = %d\n", devc->duplex); +#endif + if (devc->duplex) { + devc->iomap = &es1887_mix; + devc->iomap_sz = ARRAY_SIZE(es1887_mix); + } else { + devc->iomap = &es_rec_mix; + devc->iomap_sz = ARRAY_SIZE(es_rec_mix); + } + break; + default: + if (devc->submodel < 8) { + devc->supported_devices = ES688_MIXER_DEVICES; + devc->supported_rec_devices = ES688_RECORDING_DEVICES; + devc->iomap = &es688_mix; + devc->iomap_sz = ARRAY_SIZE(es688_mix); + } else { + /* + * es1688 has 4 bits master vol. + * later chips have 6 bits (?) + */ + devc->supported_devices = ES1688_MIXER_DEVICES; + devc->supported_rec_devices = ES1688_RECORDING_DEVICES; + if (devc->submodel < 0x10) { + devc->iomap = &es1688_mix; + devc->iomap_sz = ARRAY_SIZE(es688_mix); + } else { + devc->iomap = &es1688later_mix; + devc->iomap_sz = ARRAY_SIZE(es1688later_mix); + } + } + } +} + +/* + * Changing playback levels at an ESS chip with record mixer means having to + * take care of recording levels of recorded inputs (devc->recmask) too! + */ +int ess_mixer_set(sb_devc *devc, int dev, int left, int right) +{ + if (ess_has_rec_mixer (devc->submodel) && (devc->recmask & (1 << dev))) { + sb_common_mixer_set (devc, dev + ES_REC_MIXER_RECDIFF, left, right); + } + return sb_common_mixer_set (devc, dev, left, right); +} + +/* + * After a sb_dsp_reset extended register 0xb4 (RECLEV) is reset too. After + * sb_dsp_reset RECLEV has to be restored. This is where ess_mixer_reload + * helps. + */ +void ess_mixer_reload (sb_devc *devc, int dev) +{ + int left, right, value; + + value = devc->levels[dev]; + left = value & 0x000000ff; + right = (value & 0x0000ff00) >> 8; + + sb_common_mixer_set(devc, dev, left, right); +} + +static int es_rec_set_recmask(sb_devc * devc, int mask) +{ + int i, i_mask, cur_mask, diff_mask; + int value, left, right; + +#ifdef FKS_LOGGING +printk (KERN_INFO "FKS: es_rec_set_recmask mask = %x\n", mask); +#endif + /* + * Changing the recmask on an ESS chip with recording mixer means: + * (1) Find the differences + * (2) For "turned-on" inputs: make the recording level the playback level + * (3) For "turned-off" inputs: make the recording level zero + */ + cur_mask = devc->recmask; + diff_mask = (cur_mask ^ mask); + + for (i = 0; i < 32; i++) { + i_mask = (1 << i); + if (diff_mask & i_mask) { /* Difference? (1) */ + if (mask & i_mask) { /* Turn it on (2) */ + value = devc->levels[i]; + left = value & 0x000000ff; + right = (value & 0x0000ff00) >> 8; + } else { /* Turn it off (3) */ + left = 0; + right = 0; + } + sb_common_mixer_set(devc, i + ES_REC_MIXER_RECDIFF, left, right); + } + } + return mask; +} + +int ess_set_recmask(sb_devc * devc, int *mask) +{ + /* This applies to ESS chips with record mixers only! */ + + if (ess_has_rec_mixer (devc->submodel)) { + *mask = es_rec_set_recmask (devc, *mask); + return 1; /* Applied */ + } else { + return 0; /* Not applied */ + } +} + +/* + * ess_mixer_reset must be called from sb_mixer_reset + */ +int ess_mixer_reset (sb_devc * devc) +{ + /* + * Separate actions for ESS chips with a record mixer: + */ + if (ess_has_rec_mixer (devc->submodel)) { + switch (devc->submodel) { + case SUBMDL_ES1887: + /* + * Separate actions for ES1887: + * Change registers 7a and 1c to make the record mixer the + * actual recording source. + */ + ess_chgmixer(devc, 0x7a, 0x18, 0x08); + ess_chgmixer(devc, 0x1c, 0x07, 0x07); + break; + }; + /* + * Call set_recmask for proper initialization + */ + devc->recmask = devc->supported_rec_devices; + es_rec_set_recmask(devc, 0); + devc->recmask = 0; + + return 1; /* We took care of recmask. */ + } else { + return 0; /* We didn't take care; caller do it */ + } +} + +/**************************************************************************** + * * + * ESS midi * + * * + ****************************************************************************/ + +/* + * FKS: IRQ may be shared. Hm. And if so? Then What? + */ +int ess_midi_init(sb_devc * devc, struct address_info *hw_config) +{ + unsigned char cfg, tmp; + + cfg = ess_getmixer (devc, 0x40) & 0x03; + + if (devc->submodel < 8) { + ess_setmixer (devc, 0x40, cfg | 0x03); /* Enable OPL3 & joystick */ + return 0; /* ES688 doesn't support MPU401 mode */ + } + tmp = (hw_config->io_base & 0x0f0) >> 4; + + if (tmp > 3) { + ess_setmixer (devc, 0x40, cfg); + return 0; + } + cfg |= tmp << 3; + + tmp = 1; /* MPU enabled without interrupts */ + + /* May be shared: if so the value is -ve */ + + switch (abs(hw_config->irq)) { + case 9: + tmp = 0x4; + break; + case 5: + tmp = 0x5; + break; + case 7: + tmp = 0x6; + break; + case 10: + tmp = 0x7; + break; + default: + return 0; + } + + cfg |= tmp << 5; + ess_setmixer (devc, 0x40, cfg | 0x03); + + return 1; +} + diff --git a/sound/oss/sb_ess.h b/sound/oss/sb_ess.h new file mode 100644 index 00000000..38aa072e --- /dev/null +++ b/sound/oss/sb_ess.h @@ -0,0 +1,34 @@ +/* + * Created: 9-Jan-1999 Rolf Fokkens + */ + +extern void ess_intr + (sb_devc *devc); +extern int ess_dsp_init + (sb_devc *devc, struct address_info *hw_config); + +extern struct audio_driver *ess_audio_init + (sb_devc *devc, int *audio_flags, int *format_mask); +extern int ess_midi_init + (sb_devc *devc, struct address_info *hw_config); +extern void ess_mixer_init + (sb_devc *devc); + +extern int ess_init + (sb_devc *devc, struct address_info *hw_config); +extern int ess_dsp_reset + (sb_devc *devc); + +extern void ess_setmixer + (sb_devc *devc, unsigned int port, unsigned int value); +extern unsigned int ess_getmixer + (sb_devc *devc, unsigned int port); +extern int ess_mixer_set + (sb_devc *devc, int dev, int left, int right); +extern int ess_mixer_reset + (sb_devc *devc); +extern void ess_mixer_reload + (sb_devc * devc, int dev); +extern int ess_set_recmask + (sb_devc *devc, int *mask); + diff --git a/sound/oss/sb_midi.c b/sound/oss/sb_midi.c new file mode 100644 index 00000000..f139028e --- /dev/null +++ b/sound/oss/sb_midi.c @@ -0,0 +1,206 @@ +/* + * sound/oss/sb_midi.c + * + * The low level driver for the Sound Blaster DS chips. + * + * + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + */ + +#include <linux/spinlock.h> +#include <linux/slab.h> + +#include "sound_config.h" + +#include "sb.h" +#undef SB_TEST_IRQ + +/* + * The DSP channel can be used either for input or output. Variable + * 'sb_irq_mode' will be set when the program calls read or write first time + * after open. Current version doesn't support mode changes without closing + * and reopening the device. Support for this feature may be implemented in a + * future version of this driver. + */ + + +static int sb_midi_open(int dev, int mode, + void (*input) (int dev, unsigned char data), + void (*output) (int dev) +) +{ + sb_devc *devc = midi_devs[dev]->devc; + unsigned long flags; + + if (devc == NULL) + return -ENXIO; + + spin_lock_irqsave(&devc->lock, flags); + if (devc->opened) + { + spin_unlock_irqrestore(&devc->lock, flags); + return -EBUSY; + } + devc->opened = 1; + spin_unlock_irqrestore(&devc->lock, flags); + + devc->irq_mode = IMODE_MIDI; + devc->midi_broken = 0; + + sb_dsp_reset(devc); + + if (!sb_dsp_command(devc, 0x35)) /* Start MIDI UART mode */ + { + devc->opened = 0; + return -EIO; + } + devc->intr_active = 1; + + if (mode & OPEN_READ) + { + devc->input_opened = 1; + devc->midi_input_intr = input; + } + return 0; +} + +static void sb_midi_close(int dev) +{ + sb_devc *devc = midi_devs[dev]->devc; + unsigned long flags; + + if (devc == NULL) + return; + + spin_lock_irqsave(&devc->lock, flags); + sb_dsp_reset(devc); + devc->intr_active = 0; + devc->input_opened = 0; + devc->opened = 0; + spin_unlock_irqrestore(&devc->lock, flags); +} + +static int sb_midi_out(int dev, unsigned char midi_byte) +{ + sb_devc *devc = midi_devs[dev]->devc; + + if (devc == NULL) + return 1; + + if (devc->midi_broken) + return 1; + + if (!sb_dsp_command(devc, midi_byte)) + { + devc->midi_broken = 1; + return 1; + } + return 1; +} + +static int sb_midi_start_read(int dev) +{ + return 0; +} + +static int sb_midi_end_read(int dev) +{ + sb_devc *devc = midi_devs[dev]->devc; + + if (devc == NULL) + return -ENXIO; + + sb_dsp_reset(devc); + devc->intr_active = 0; + return 0; +} + +static int sb_midi_ioctl(int dev, unsigned cmd, void __user *arg) +{ + return -EINVAL; +} + +void sb_midi_interrupt(sb_devc * devc) +{ + unsigned long flags; + unsigned char data; + + if (devc == NULL) + return; + + spin_lock_irqsave(&devc->lock, flags); + + data = inb(DSP_READ); + if (devc->input_opened) + devc->midi_input_intr(devc->my_mididev, data); + + spin_unlock_irqrestore(&devc->lock, flags); +} + +#define MIDI_SYNTH_NAME "Sound Blaster Midi" +#define MIDI_SYNTH_CAPS 0 +#include "midi_synth.h" + +static struct midi_operations sb_midi_operations = +{ + .owner = THIS_MODULE, + .info = {"Sound Blaster", 0, 0, SNDCARD_SB}, + .converter = &std_midi_synth, + .in_info = {0}, + .open = sb_midi_open, + .close = sb_midi_close, + .ioctl = sb_midi_ioctl, + .outputc = sb_midi_out, + .start_read = sb_midi_start_read, + .end_read = sb_midi_end_read, +}; + +void sb_dsp_midi_init(sb_devc * devc, struct module *owner) +{ + int dev; + + if (devc->model < 2) /* No MIDI support for SB 1.x */ + return; + + dev = sound_alloc_mididev(); + + if (dev == -1) + { + printk(KERN_ERR "sb_midi: too many MIDI devices detected\n"); + return; + } + std_midi_synth.midi_dev = devc->my_mididev = dev; + midi_devs[dev] = kmalloc(sizeof(struct midi_operations), GFP_KERNEL); + if (midi_devs[dev] == NULL) + { + printk(KERN_WARNING "Sound Blaster: failed to allocate MIDI memory.\n"); + sound_unload_mididev(dev); + return; + } + memcpy((char *) midi_devs[dev], (char *) &sb_midi_operations, + sizeof(struct midi_operations)); + + if (owner) + midi_devs[dev]->owner = owner; + + midi_devs[dev]->devc = devc; + + + midi_devs[dev]->converter = kmalloc(sizeof(struct synth_operations), GFP_KERNEL); + if (midi_devs[dev]->converter == NULL) + { + printk(KERN_WARNING "Sound Blaster: failed to allocate MIDI memory.\n"); + kfree(midi_devs[dev]); + sound_unload_mididev(dev); + return; + } + memcpy((char *) midi_devs[dev]->converter, (char *) &std_midi_synth, + sizeof(struct synth_operations)); + + midi_devs[dev]->converter->id = "SBMIDI"; + sequencer_init(); +} diff --git a/sound/oss/sb_mixer.c b/sound/oss/sb_mixer.c new file mode 100644 index 00000000..2039d31b --- /dev/null +++ b/sound/oss/sb_mixer.c @@ -0,0 +1,770 @@ +/* + * sound/oss/sb_mixer.c + * + * The low level mixer driver for the Sound Blaster compatible cards. + */ +/* + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * + * + * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed) + * Rolf Fokkens (Dec 20 1998) : Moved ESS stuff into sb_ess.[ch] + * Stanislav Voronyi <stas@esc.kharkov.com> : Support for AWE 3DSE device (Jun 7 1999) + */ + +#include <linux/slab.h> + +#include "sound_config.h" + +#define __SB_MIXER_C__ + +#include "sb.h" +#include "sb_mixer.h" + +#include "sb_ess.h" + +#define SBPRO_RECORDING_DEVICES (SOUND_MASK_LINE | SOUND_MASK_MIC | SOUND_MASK_CD) + +/* Same as SB Pro, unless I find otherwise */ +#define SGNXPRO_RECORDING_DEVICES SBPRO_RECORDING_DEVICES + +#define SBPRO_MIXER_DEVICES (SOUND_MASK_SYNTH | SOUND_MASK_PCM | SOUND_MASK_LINE | SOUND_MASK_MIC | \ + SOUND_MASK_CD | SOUND_MASK_VOLUME) + +/* SG NX Pro has treble and bass settings on the mixer. The 'speaker' + * channel is the COVOX/DisneySoundSource emulation volume control + * on the mixer. It does NOT control speaker volume. Should have own + * mask eventually? + */ +#define SGNXPRO_MIXER_DEVICES (SBPRO_MIXER_DEVICES|SOUND_MASK_BASS| \ + SOUND_MASK_TREBLE|SOUND_MASK_SPEAKER ) + +#define SB16_RECORDING_DEVICES (SOUND_MASK_SYNTH | SOUND_MASK_LINE | SOUND_MASK_MIC | \ + SOUND_MASK_CD) + +#define SB16_OUTFILTER_DEVICES (SOUND_MASK_LINE | SOUND_MASK_MIC | \ + SOUND_MASK_CD) + +#define SB16_MIXER_DEVICES (SOUND_MASK_SYNTH | SOUND_MASK_PCM | SOUND_MASK_SPEAKER | SOUND_MASK_LINE | SOUND_MASK_MIC | \ + SOUND_MASK_CD | \ + SOUND_MASK_IGAIN | SOUND_MASK_OGAIN | \ + SOUND_MASK_VOLUME | SOUND_MASK_BASS | SOUND_MASK_TREBLE | \ + SOUND_MASK_IMIX) + +/* These are the only devices that are working at the moment. Others could + * be added once they are identified and a method is found to control them. + */ +#define ALS007_MIXER_DEVICES (SOUND_MASK_SYNTH | SOUND_MASK_LINE | \ + SOUND_MASK_PCM | SOUND_MASK_MIC | \ + SOUND_MASK_CD | \ + SOUND_MASK_VOLUME) + +static mixer_tab sbpro_mix = { +MIX_ENT(SOUND_MIXER_VOLUME, 0x22, 7, 4, 0x22, 3, 4), +MIX_ENT(SOUND_MIXER_BASS, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_TREBLE, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_SYNTH, 0x26, 7, 4, 0x26, 3, 4), +MIX_ENT(SOUND_MIXER_PCM, 0x04, 7, 4, 0x04, 3, 4), +MIX_ENT(SOUND_MIXER_SPEAKER, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE, 0x2e, 7, 4, 0x2e, 3, 4), +MIX_ENT(SOUND_MIXER_MIC, 0x0a, 2, 3, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_CD, 0x28, 7, 4, 0x28, 3, 4), +MIX_ENT(SOUND_MIXER_IMIX, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_RECLEV, 0x00, 0, 0, 0x00, 0, 0) +}; + +static mixer_tab sb16_mix = { +MIX_ENT(SOUND_MIXER_VOLUME, 0x30, 7, 5, 0x31, 7, 5), +MIX_ENT(SOUND_MIXER_BASS, 0x46, 7, 4, 0x47, 7, 4), +MIX_ENT(SOUND_MIXER_TREBLE, 0x44, 7, 4, 0x45, 7, 4), +MIX_ENT(SOUND_MIXER_SYNTH, 0x34, 7, 5, 0x35, 7, 5), +MIX_ENT(SOUND_MIXER_PCM, 0x32, 7, 5, 0x33, 7, 5), +MIX_ENT(SOUND_MIXER_SPEAKER, 0x3b, 7, 2, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE, 0x38, 7, 5, 0x39, 7, 5), +MIX_ENT(SOUND_MIXER_MIC, 0x3a, 7, 5, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_CD, 0x36, 7, 5, 0x37, 7, 5), +MIX_ENT(SOUND_MIXER_IMIX, 0x3c, 0, 1, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_RECLEV, 0x3f, 7, 2, 0x40, 7, 2), /* Obsolete. Use IGAIN */ +MIX_ENT(SOUND_MIXER_IGAIN, 0x3f, 7, 2, 0x40, 7, 2), +MIX_ENT(SOUND_MIXER_OGAIN, 0x41, 7, 2, 0x42, 7, 2) +}; + +static mixer_tab als007_mix = +{ +MIX_ENT(SOUND_MIXER_VOLUME, 0x62, 7, 4, 0x62, 3, 4), +MIX_ENT(SOUND_MIXER_BASS, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_TREBLE, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_SYNTH, 0x66, 7, 4, 0x66, 3, 4), +MIX_ENT(SOUND_MIXER_PCM, 0x64, 7, 4, 0x64, 3, 4), +MIX_ENT(SOUND_MIXER_SPEAKER, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_LINE, 0x6e, 7, 4, 0x6e, 3, 4), +MIX_ENT(SOUND_MIXER_MIC, 0x6a, 2, 3, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_CD, 0x68, 7, 4, 0x68, 3, 4), +MIX_ENT(SOUND_MIXER_IMIX, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_RECLEV, 0x00, 0, 0, 0x00, 0, 0), /* Obsolete. Use IGAIN */ +MIX_ENT(SOUND_MIXER_IGAIN, 0x00, 0, 0, 0x00, 0, 0), +MIX_ENT(SOUND_MIXER_OGAIN, 0x00, 0, 0, 0x00, 0, 0) +}; + + +/* SM_GAMES Master volume is lower and PCM & FM volumes + higher than with SB Pro. This improves the + sound quality */ + +static int smg_default_levels[32] = +{ + 0x2020, /* Master Volume */ + 0x4b4b, /* Bass */ + 0x4b4b, /* Treble */ + 0x6464, /* FM */ + 0x6464, /* PCM */ + 0x4b4b, /* PC Speaker */ + 0x4b4b, /* Ext Line */ + 0x0000, /* Mic */ + 0x4b4b, /* CD */ + 0x4b4b, /* Recording monitor */ + 0x4b4b, /* SB PCM */ + 0x4b4b, /* Recording level */ + 0x4b4b, /* Input gain */ + 0x4b4b, /* Output gain */ + 0x4040, /* Line1 */ + 0x4040, /* Line2 */ + 0x1515 /* Line3 */ +}; + +static int sb_default_levels[32] = +{ + 0x5a5a, /* Master Volume */ + 0x4b4b, /* Bass */ + 0x4b4b, /* Treble */ + 0x4b4b, /* FM */ + 0x4b4b, /* PCM */ + 0x4b4b, /* PC Speaker */ + 0x4b4b, /* Ext Line */ + 0x1010, /* Mic */ + 0x4b4b, /* CD */ + 0x0000, /* Recording monitor */ + 0x4b4b, /* SB PCM */ + 0x4b4b, /* Recording level */ + 0x4b4b, /* Input gain */ + 0x4b4b, /* Output gain */ + 0x4040, /* Line1 */ + 0x4040, /* Line2 */ + 0x1515 /* Line3 */ +}; + +static unsigned char sb16_recmasks_L[SOUND_MIXER_NRDEVICES] = +{ + 0x00, /* SOUND_MIXER_VOLUME */ + 0x00, /* SOUND_MIXER_BASS */ + 0x00, /* SOUND_MIXER_TREBLE */ + 0x40, /* SOUND_MIXER_SYNTH */ + 0x00, /* SOUND_MIXER_PCM */ + 0x00, /* SOUND_MIXER_SPEAKER */ + 0x10, /* SOUND_MIXER_LINE */ + 0x01, /* SOUND_MIXER_MIC */ + 0x04, /* SOUND_MIXER_CD */ + 0x00, /* SOUND_MIXER_IMIX */ + 0x00, /* SOUND_MIXER_ALTPCM */ + 0x00, /* SOUND_MIXER_RECLEV */ + 0x00, /* SOUND_MIXER_IGAIN */ + 0x00 /* SOUND_MIXER_OGAIN */ +}; + +static unsigned char sb16_recmasks_R[SOUND_MIXER_NRDEVICES] = +{ + 0x00, /* SOUND_MIXER_VOLUME */ + 0x00, /* SOUND_MIXER_BASS */ + 0x00, /* SOUND_MIXER_TREBLE */ + 0x20, /* SOUND_MIXER_SYNTH */ + 0x00, /* SOUND_MIXER_PCM */ + 0x00, /* SOUND_MIXER_SPEAKER */ + 0x08, /* SOUND_MIXER_LINE */ + 0x01, /* SOUND_MIXER_MIC */ + 0x02, /* SOUND_MIXER_CD */ + 0x00, /* SOUND_MIXER_IMIX */ + 0x00, /* SOUND_MIXER_ALTPCM */ + 0x00, /* SOUND_MIXER_RECLEV */ + 0x00, /* SOUND_MIXER_IGAIN */ + 0x00 /* SOUND_MIXER_OGAIN */ +}; + +static char smw_mix_regs[] = /* Left mixer registers */ +{ + 0x0b, /* SOUND_MIXER_VOLUME */ + 0x0d, /* SOUND_MIXER_BASS */ + 0x0d, /* SOUND_MIXER_TREBLE */ + 0x05, /* SOUND_MIXER_SYNTH */ + 0x09, /* SOUND_MIXER_PCM */ + 0x00, /* SOUND_MIXER_SPEAKER */ + 0x03, /* SOUND_MIXER_LINE */ + 0x01, /* SOUND_MIXER_MIC */ + 0x07, /* SOUND_MIXER_CD */ + 0x00, /* SOUND_MIXER_IMIX */ + 0x00, /* SOUND_MIXER_ALTPCM */ + 0x00, /* SOUND_MIXER_RECLEV */ + 0x00, /* SOUND_MIXER_IGAIN */ + 0x00, /* SOUND_MIXER_OGAIN */ + 0x00, /* SOUND_MIXER_LINE1 */ + 0x00, /* SOUND_MIXER_LINE2 */ + 0x00 /* SOUND_MIXER_LINE3 */ +}; + +static int sbmixnum = 1; + +static void sb_mixer_reset(sb_devc * devc); + +void sb_mixer_set_stereo(sb_devc * devc, int mode) +{ + sb_chgmixer(devc, OUT_FILTER, STEREO_DAC, (mode ? STEREO_DAC : MONO_DAC)); +} + +static int detect_mixer(sb_devc * devc) +{ + /* Just trust the mixer is there */ + return 1; +} + +static void change_bits(sb_devc * devc, unsigned char *regval, int dev, int chn, int newval) +{ + unsigned char mask; + int shift; + + mask = (1 << (*devc->iomap)[dev][chn].nbits) - 1; + newval = (int) ((newval * mask) + 50) / 100; /* Scale */ + + shift = (*devc->iomap)[dev][chn].bitoffs - (*devc->iomap)[dev][LEFT_CHN].nbits + 1; + + *regval &= ~(mask << shift); /* Mask out previous value */ + *regval |= (newval & mask) << shift; /* Set the new value */ +} + +static int sb_mixer_get(sb_devc * devc, int dev) +{ + if (!((1 << dev) & devc->supported_devices)) + return -EINVAL; + return devc->levels[dev]; +} + +void smw_mixer_init(sb_devc * devc) +{ + int i; + + sb_setmixer(devc, 0x00, 0x18); /* Mute unused (Telephone) line */ + sb_setmixer(devc, 0x10, 0x38); /* Config register 2 */ + + devc->supported_devices = 0; + for (i = 0; i < sizeof(smw_mix_regs); i++) + if (smw_mix_regs[i] != 0) + devc->supported_devices |= (1 << i); + + devc->supported_rec_devices = devc->supported_devices & + ~(SOUND_MASK_BASS | SOUND_MASK_TREBLE | SOUND_MASK_PCM | SOUND_MASK_VOLUME); + sb_mixer_reset(devc); +} + +int sb_common_mixer_set(sb_devc * devc, int dev, int left, int right) +{ + int regoffs; + unsigned char val; + + if ((dev < 0) || (dev >= devc->iomap_sz)) + return -EINVAL; + + regoffs = (*devc->iomap)[dev][LEFT_CHN].regno; + + if (regoffs == 0) + return -EINVAL; + + val = sb_getmixer(devc, regoffs); + change_bits(devc, &val, dev, LEFT_CHN, left); + + if ((*devc->iomap)[dev][RIGHT_CHN].regno != regoffs) /* + * Change register + */ + { + sb_setmixer(devc, regoffs, val); /* + * Save the old one + */ + regoffs = (*devc->iomap)[dev][RIGHT_CHN].regno; + + if (regoffs == 0) + return left | (left << 8); /* + * Just left channel present + */ + + val = sb_getmixer(devc, regoffs); /* + * Read the new one + */ + } + change_bits(devc, &val, dev, RIGHT_CHN, right); + + sb_setmixer(devc, regoffs, val); + + return left | (right << 8); +} + +static int smw_mixer_set(sb_devc * devc, int dev, int left, int right) +{ + int reg, val; + + switch (dev) + { + case SOUND_MIXER_VOLUME: + sb_setmixer(devc, 0x0b, 96 - (96 * left / 100)); /* 96=mute, 0=max */ + sb_setmixer(devc, 0x0c, 96 - (96 * right / 100)); + break; + + case SOUND_MIXER_BASS: + case SOUND_MIXER_TREBLE: + devc->levels[dev] = left | (right << 8); + /* Set left bass and treble values */ + val = ((devc->levels[SOUND_MIXER_TREBLE] & 0xff) * 16 / (unsigned) 100) << 4; + val |= ((devc->levels[SOUND_MIXER_BASS] & 0xff) * 16 / (unsigned) 100) & 0x0f; + sb_setmixer(devc, 0x0d, val); + + /* Set right bass and treble values */ + val = (((devc->levels[SOUND_MIXER_TREBLE] >> 8) & 0xff) * 16 / (unsigned) 100) << 4; + val |= (((devc->levels[SOUND_MIXER_BASS] >> 8) & 0xff) * 16 / (unsigned) 100) & 0x0f; + sb_setmixer(devc, 0x0e, val); + + break; + + default: + /* bounds check */ + if (dev < 0 || dev >= ARRAY_SIZE(smw_mix_regs)) + return -EINVAL; + reg = smw_mix_regs[dev]; + if (reg == 0) + return -EINVAL; + sb_setmixer(devc, reg, (24 - (24 * left / 100)) | 0x20); /* 24=mute, 0=max */ + sb_setmixer(devc, reg + 1, (24 - (24 * right / 100)) | 0x40); + } + + devc->levels[dev] = left | (right << 8); + return left | (right << 8); +} + +static int sb_mixer_set(sb_devc * devc, int dev, int value) +{ + int left = value & 0x000000ff; + int right = (value & 0x0000ff00) >> 8; + int retval; + + if (left > 100) + left = 100; + if (right > 100) + right = 100; + + if ((dev < 0) || (dev > 31)) + return -EINVAL; + + if (!(devc->supported_devices & (1 << dev))) /* + * Not supported + */ + return -EINVAL; + + /* Differentiate depending on the chipsets */ + switch (devc->model) { + case MDL_SMW: + retval = smw_mixer_set(devc, dev, left, right); + break; + case MDL_ESS: + retval = ess_mixer_set(devc, dev, left, right); + break; + default: + retval = sb_common_mixer_set(devc, dev, left, right); + } + if (retval >= 0) devc->levels[dev] = retval; + + return retval; +} + +/* + * set_recsrc doesn't apply to ES188x + */ +static void set_recsrc(sb_devc * devc, int src) +{ + sb_setmixer(devc, RECORD_SRC, (sb_getmixer(devc, RECORD_SRC) & ~7) | (src & 0x7)); +} + +static int set_recmask(sb_devc * devc, int mask) +{ + int devmask, i; + unsigned char regimageL, regimageR; + + devmask = mask & devc->supported_rec_devices; + + switch (devc->model) + { + case MDL_SBPRO: + case MDL_ESS: + case MDL_JAZZ: + case MDL_SMW: + if (devc->model == MDL_ESS && ess_set_recmask (devc, &devmask)) { + break; + }; + if (devmask != SOUND_MASK_MIC && + devmask != SOUND_MASK_LINE && + devmask != SOUND_MASK_CD) + { + /* + * More than one device selected. Drop the + * previous selection + */ + devmask &= ~devc->recmask; + } + if (devmask != SOUND_MASK_MIC && + devmask != SOUND_MASK_LINE && + devmask != SOUND_MASK_CD) + { + /* + * More than one device selected. Default to + * mic + */ + devmask = SOUND_MASK_MIC; + } + if (devmask ^ devc->recmask) /* + * Input source changed + */ + { + switch (devmask) + { + case SOUND_MASK_MIC: + set_recsrc(devc, SRC__MIC); + break; + + case SOUND_MASK_LINE: + set_recsrc(devc, SRC__LINE); + break; + + case SOUND_MASK_CD: + set_recsrc(devc, SRC__CD); + break; + + default: + set_recsrc(devc, SRC__MIC); + } + } + break; + + case MDL_SB16: + if (!devmask) + devmask = SOUND_MASK_MIC; + + if (devc->submodel == SUBMDL_ALS007) + { + switch (devmask) + { + case SOUND_MASK_LINE: + sb_setmixer(devc, ALS007_RECORD_SRC, ALS007_LINE); + break; + case SOUND_MASK_CD: + sb_setmixer(devc, ALS007_RECORD_SRC, ALS007_CD); + break; + case SOUND_MASK_SYNTH: + sb_setmixer(devc, ALS007_RECORD_SRC, ALS007_SYNTH); + break; + default: /* Also takes care of SOUND_MASK_MIC case */ + sb_setmixer(devc, ALS007_RECORD_SRC, ALS007_MIC); + break; + } + } + else + { + regimageL = regimageR = 0; + for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) + { + if ((1 << i) & devmask) + { + regimageL |= sb16_recmasks_L[i]; + regimageR |= sb16_recmasks_R[i]; + } + sb_setmixer (devc, SB16_IMASK_L, regimageL); + sb_setmixer (devc, SB16_IMASK_R, regimageR); + } + } + break; + } + devc->recmask = devmask; + return devc->recmask; +} + +static int set_outmask(sb_devc * devc, int mask) +{ + int devmask, i; + unsigned char regimage; + + devmask = mask & devc->supported_out_devices; + + switch (devc->model) + { + case MDL_SB16: + if (devc->submodel == SUBMDL_ALS007) + break; + else + { + regimage = 0; + for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) + { + if ((1 << i) & devmask) + { + regimage |= (sb16_recmasks_L[i] | sb16_recmasks_R[i]); + } + sb_setmixer (devc, SB16_OMASK, regimage); + } + } + break; + default: + break; + } + + devc->outmask = devmask; + return devc->outmask; +} + +static int sb_mixer_ioctl(int dev, unsigned int cmd, void __user *arg) +{ + sb_devc *devc = mixer_devs[dev]->devc; + int val, ret; + int __user *p = arg; + + /* + * Use ioctl(fd, SOUND_MIXER_AGC, &mode) to turn AGC off (0) or on (1). + * Use ioctl(fd, SOUND_MIXER_3DSE, &mode) to turn 3DSE off (0) or on (1) + * or mode==2 put 3DSE state to mode. + */ + if (devc->model == MDL_SB16) { + if (cmd == SOUND_MIXER_AGC) + { + if (get_user(val, p)) + return -EFAULT; + sb_setmixer(devc, 0x43, (~val) & 0x01); + return 0; + } + if (cmd == SOUND_MIXER_3DSE) + { + /* I put here 15, but I don't know the exact version. + At least my 4.13 havn't 3DSE, 4.16 has it. */ + if (devc->minor < 15) + return -EINVAL; + if (get_user(val, p)) + return -EFAULT; + if (val == 0 || val == 1) + sb_chgmixer(devc, AWE_3DSE, 0x01, val); + else if (val == 2) + { + ret = sb_getmixer(devc, AWE_3DSE)&0x01; + return put_user(ret, p); + } + else + return -EINVAL; + return 0; + } + } + if (((cmd >> 8) & 0xff) == 'M') + { + if (_SIOC_DIR(cmd) & _SIOC_WRITE) + { + if (get_user(val, p)) + return -EFAULT; + switch (cmd & 0xff) + { + case SOUND_MIXER_RECSRC: + ret = set_recmask(devc, val); + break; + + case SOUND_MIXER_OUTSRC: + ret = set_outmask(devc, val); + break; + + default: + ret = sb_mixer_set(devc, cmd & 0xff, val); + } + } + else switch (cmd & 0xff) + { + case SOUND_MIXER_RECSRC: + ret = devc->recmask; + break; + + case SOUND_MIXER_OUTSRC: + ret = devc->outmask; + break; + + case SOUND_MIXER_DEVMASK: + ret = devc->supported_devices; + break; + + case SOUND_MIXER_STEREODEVS: + ret = devc->supported_devices; + /* The ESS seems to have stereo mic controls */ + if (devc->model == MDL_ESS) + ret &= ~(SOUND_MASK_SPEAKER|SOUND_MASK_IMIX); + else if (devc->model != MDL_JAZZ && devc->model != MDL_SMW) + ret &= ~(SOUND_MASK_MIC | SOUND_MASK_SPEAKER | SOUND_MASK_IMIX); + break; + + case SOUND_MIXER_RECMASK: + ret = devc->supported_rec_devices; + break; + + case SOUND_MIXER_OUTMASK: + ret = devc->supported_out_devices; + break; + + case SOUND_MIXER_CAPS: + ret = devc->mixer_caps; + break; + + default: + ret = sb_mixer_get(devc, cmd & 0xff); + break; + } + return put_user(ret, p); + } else + return -EINVAL; +} + +static struct mixer_operations sb_mixer_operations = +{ + .owner = THIS_MODULE, + .id = "SB", + .name = "Sound Blaster", + .ioctl = sb_mixer_ioctl +}; + +static struct mixer_operations als007_mixer_operations = +{ + .owner = THIS_MODULE, + .id = "ALS007", + .name = "Avance ALS-007", + .ioctl = sb_mixer_ioctl +}; + +static void sb_mixer_reset(sb_devc * devc) +{ + char name[32]; + int i; + + sprintf(name, "SB_%d", devc->sbmixnum); + + if (devc->sbmo.sm_games) + devc->levels = load_mixer_volumes(name, smg_default_levels, 1); + else + devc->levels = load_mixer_volumes(name, sb_default_levels, 1); + + for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) + sb_mixer_set(devc, i, devc->levels[i]); + + if (devc->model != MDL_ESS || !ess_mixer_reset (devc)) { + set_recmask(devc, SOUND_MASK_MIC); + }; +} + +int sb_mixer_init(sb_devc * devc, struct module *owner) +{ + int mixer_type = 0; + int m; + + devc->sbmixnum = sbmixnum++; + devc->levels = NULL; + + sb_setmixer(devc, 0x00, 0); /* Reset mixer */ + + if (!(mixer_type = detect_mixer(devc))) + return 0; /* No mixer. Why? */ + + switch (devc->model) + { + case MDL_ESSPCI: + case MDL_YMPCI: + case MDL_SBPRO: + case MDL_AZTECH: + case MDL_JAZZ: + devc->mixer_caps = SOUND_CAP_EXCL_INPUT; + devc->supported_devices = SBPRO_MIXER_DEVICES; + devc->supported_rec_devices = SBPRO_RECORDING_DEVICES; + devc->iomap = &sbpro_mix; + devc->iomap_sz = ARRAY_SIZE(sbpro_mix); + break; + + case MDL_ESS: + ess_mixer_init (devc); + break; + + case MDL_SMW: + devc->mixer_caps = SOUND_CAP_EXCL_INPUT; + devc->supported_devices = 0; + devc->supported_rec_devices = 0; + devc->iomap = &sbpro_mix; + devc->iomap_sz = ARRAY_SIZE(sbpro_mix); + smw_mixer_init(devc); + break; + + case MDL_SB16: + devc->mixer_caps = 0; + devc->supported_rec_devices = SB16_RECORDING_DEVICES; + devc->supported_out_devices = SB16_OUTFILTER_DEVICES; + if (devc->submodel != SUBMDL_ALS007) + { + devc->supported_devices = SB16_MIXER_DEVICES; + devc->iomap = &sb16_mix; + devc->iomap_sz = ARRAY_SIZE(sb16_mix); + } + else + { + devc->supported_devices = ALS007_MIXER_DEVICES; + devc->iomap = &als007_mix; + devc->iomap_sz = ARRAY_SIZE(als007_mix); + } + break; + + default: + printk(KERN_WARNING "sb_mixer: Unsupported mixer type %d\n", devc->model); + return 0; + } + + m = sound_alloc_mixerdev(); + if (m == -1) + return 0; + + mixer_devs[m] = kmalloc(sizeof(struct mixer_operations), GFP_KERNEL); + if (mixer_devs[m] == NULL) + { + printk(KERN_ERR "sb_mixer: Can't allocate memory\n"); + sound_unload_mixerdev(m); + return 0; + } + + if (devc->submodel != SUBMDL_ALS007) + memcpy ((char *) mixer_devs[m], (char *) &sb_mixer_operations, sizeof (struct mixer_operations)); + else + memcpy ((char *) mixer_devs[m], (char *) &als007_mixer_operations, sizeof (struct mixer_operations)); + + mixer_devs[m]->devc = devc; + + if (owner) + mixer_devs[m]->owner = owner; + + devc->my_mixerdev = m; + sb_mixer_reset(devc); + return 1; +} + +void sb_mixer_unload(sb_devc *devc) +{ + if (devc->my_mixerdev == -1) + return; + + kfree(mixer_devs[devc->my_mixerdev]); + sound_unload_mixerdev(devc->my_mixerdev); + sbmixnum--; +} diff --git a/sound/oss/sb_mixer.h b/sound/oss/sb_mixer.h new file mode 100644 index 00000000..4b9425f0 --- /dev/null +++ b/sound/oss/sb_mixer.h @@ -0,0 +1,105 @@ +/* + * sound/oss/sb_mixer.h + * + * Definitions for the SB Pro and SB16 mixers + */ +/* + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + */ + +/* + * Modified: + * Hunyue Yau Jan 6 1994 + * Added defines for the Sound Galaxy NX Pro mixer. + * + * Rolf Fokkens Dec 20 1998 + * Added defines for some ES188x chips. + * + * Rolf Fokkens Dec 27 1998 + * Moved static stuff to sb_mixer.c + * + */ +/* + * Mixer registers + * + * NOTE! RECORD_SRC == IN_FILTER + */ + +/* + * Mixer registers of SB Pro + */ +#define VOC_VOL 0x04 +#define MIC_VOL 0x0A +#define MIC_MIX 0x0A +#define RECORD_SRC 0x0C +#define IN_FILTER 0x0C +#define OUT_FILTER 0x0E +#define MASTER_VOL 0x22 +#define FM_VOL 0x26 +#define CD_VOL 0x28 +#define LINE_VOL 0x2E +#define IRQ_NR 0x80 +#define DMA_NR 0x81 +#define IRQ_STAT 0x82 +#define OPSW 0x3c + +/* + * Additional registers on the SG NX Pro + */ +#define COVOX_VOL 0x42 +#define TREBLE_LVL 0x44 +#define BASS_LVL 0x46 + +#define FREQ_HI (1 << 3)/* Use High-frequency ANFI filters */ +#define FREQ_LOW 0 /* Use Low-frequency ANFI filters */ +#define FILT_ON 0 /* Yes, 0 to turn it on, 1 for off */ +#define FILT_OFF (1 << 5) + +#define MONO_DAC 0x00 +#define STEREO_DAC 0x02 + +/* + * Mixer registers of SB16 + */ +#define SB16_OMASK 0x3c +#define SB16_IMASK_L 0x3d +#define SB16_IMASK_R 0x3e + +#define LEFT_CHN 0 +#define RIGHT_CHN 1 + +/* + * 3DSE register of AWE32/64 + */ +#define AWE_3DSE 0x90 + +/* + * Mixer registers of ALS007 + */ +#define ALS007_RECORD_SRC 0x6c +#define ALS007_OUTPUT_CTRL1 0x3c +#define ALS007_OUTPUT_CTRL2 0x4c + +#define MIX_ENT(name, reg_l, bit_l, len_l, reg_r, bit_r, len_r) \ + {{reg_l, bit_l, len_l}, {reg_r, bit_r, len_r}} + +/* + * Recording sources (SB Pro) + */ + +#define SRC__MIC 1 /* Select Microphone recording source */ +#define SRC__CD 3 /* Select CD recording source */ +#define SRC__LINE 7 /* Use Line-in for recording source */ + +/* + * Recording sources for ALS-007 + */ + +#define ALS007_MIC 4 +#define ALS007_LINE 6 +#define ALS007_CD 2 +#define ALS007_SYNTH 7 diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c new file mode 100644 index 00000000..30bcfe47 --- /dev/null +++ b/sound/oss/sequencer.c @@ -0,0 +1,1671 @@ +/* + * sound/oss/sequencer.c + * + * The sequencer personality manager. + */ +/* + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + */ +/* + * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed) + * Alan Cox : reformatted and fixed a pair of null pointer bugs + */ +#include <linux/kmod.h> +#include <linux/spinlock.h> +#include "sound_config.h" + +#include "midi_ctrl.h" + +static int sequencer_ok; +static struct sound_timer_operations *tmr; +static int tmr_no = -1; /* Currently selected timer */ +static int pending_timer = -1; /* For timer change operation */ +extern unsigned long seq_time; + +static int obsolete_api_used; +static DEFINE_SPINLOCK(lock); + +/* + * Local counts for number of synth and MIDI devices. These are initialized + * by the sequencer_open. + */ +static int max_mididev; +static int max_synthdev; + +/* + * The seq_mode gives the operating mode of the sequencer: + * 1 = level1 (the default) + * 2 = level2 (extended capabilities) + */ + +#define SEQ_1 1 +#define SEQ_2 2 +static int seq_mode = SEQ_1; + +static DECLARE_WAIT_QUEUE_HEAD(seq_sleeper); +static DECLARE_WAIT_QUEUE_HEAD(midi_sleeper); + +static int midi_opened[MAX_MIDI_DEV]; + +static int midi_written[MAX_MIDI_DEV]; + +static unsigned long prev_input_time; +static int prev_event_time; + +#include "tuning.h" + +#define EV_SZ 8 +#define IEV_SZ 8 + +static unsigned char *queue; +static unsigned char *iqueue; + +static volatile int qhead, qtail, qlen; +static volatile int iqhead, iqtail, iqlen; +static volatile int seq_playing; +static volatile int sequencer_busy; +static int output_threshold; +static long pre_event_timeout; +static unsigned synth_open_mask; + +static int seq_queue(unsigned char *note, char nonblock); +static void seq_startplay(void); +static int seq_sync(void); +static void seq_reset(void); + +#if MAX_SYNTH_DEV > 15 +#error Too many synthesizer devices enabled. +#endif + +int sequencer_read(int dev, struct file *file, char __user *buf, int count) +{ + int c = count, p = 0; + int ev_len; + unsigned long flags; + + dev = dev >> 4; + + ev_len = seq_mode == SEQ_1 ? 4 : 8; + + spin_lock_irqsave(&lock,flags); + + if (!iqlen) + { + spin_unlock_irqrestore(&lock,flags); + if (file->f_flags & O_NONBLOCK) { + return -EAGAIN; + } + + interruptible_sleep_on_timeout(&midi_sleeper, + pre_event_timeout); + spin_lock_irqsave(&lock,flags); + if (!iqlen) + { + spin_unlock_irqrestore(&lock,flags); + return 0; + } + } + while (iqlen && c >= ev_len) + { + char *fixit = (char *) &iqueue[iqhead * IEV_SZ]; + spin_unlock_irqrestore(&lock,flags); + if (copy_to_user(&(buf)[p], fixit, ev_len)) + return count - c; + p += ev_len; + c -= ev_len; + + spin_lock_irqsave(&lock,flags); + iqhead = (iqhead + 1) % SEQ_MAX_QUEUE; + iqlen--; + } + spin_unlock_irqrestore(&lock,flags); + return count - c; +} + +static void sequencer_midi_output(int dev) +{ + /* + * Currently NOP + */ +} + +void seq_copy_to_input(unsigned char *event_rec, int len) +{ + unsigned long flags; + + /* + * Verify that the len is valid for the current mode. + */ + + if (len != 4 && len != 8) + return; + if ((seq_mode == SEQ_1) != (len == 4)) + return; + + if (iqlen >= (SEQ_MAX_QUEUE - 1)) + return; /* Overflow */ + + spin_lock_irqsave(&lock,flags); + memcpy(&iqueue[iqtail * IEV_SZ], event_rec, len); + iqlen++; + iqtail = (iqtail + 1) % SEQ_MAX_QUEUE; + wake_up(&midi_sleeper); + spin_unlock_irqrestore(&lock,flags); +} +EXPORT_SYMBOL(seq_copy_to_input); + +static void sequencer_midi_input(int dev, unsigned char data) +{ + unsigned int tstamp; + unsigned char event_rec[4]; + + if (data == 0xfe) /* Ignore active sensing */ + return; + + tstamp = jiffies - seq_time; + + if (tstamp != prev_input_time) + { + tstamp = (tstamp << 8) | SEQ_WAIT; + seq_copy_to_input((unsigned char *) &tstamp, 4); + prev_input_time = tstamp; + } + event_rec[0] = SEQ_MIDIPUTC; + event_rec[1] = data; + event_rec[2] = dev; + event_rec[3] = 0; + + seq_copy_to_input(event_rec, 4); +} + +void seq_input_event(unsigned char *event_rec, int len) +{ + unsigned long this_time; + + if (seq_mode == SEQ_2) + this_time = tmr->get_time(tmr_no); + else + this_time = jiffies - seq_time; + + if (this_time != prev_input_time) + { + unsigned char tmp_event[8]; + + tmp_event[0] = EV_TIMING; + tmp_event[1] = TMR_WAIT_ABS; + tmp_event[2] = 0; + tmp_event[3] = 0; + *(unsigned int *) &tmp_event[4] = this_time; + + seq_copy_to_input(tmp_event, 8); + prev_input_time = this_time; + } + seq_copy_to_input(event_rec, len); +} +EXPORT_SYMBOL(seq_input_event); + +int sequencer_write(int dev, struct file *file, const char __user *buf, int count) +{ + unsigned char event_rec[EV_SZ], ev_code; + int p = 0, c, ev_size; + int mode = translate_mode(file); + + dev = dev >> 4; + + DEB(printk("sequencer_write(dev=%d, count=%d)\n", dev, count)); + + if (mode == OPEN_READ) + return -EIO; + + c = count; + + while (c >= 4) + { + if (copy_from_user((char *) event_rec, &(buf)[p], 4)) + goto out; + ev_code = event_rec[0]; + + if (ev_code == SEQ_FULLSIZE) + { + int err, fmt; + + dev = *(unsigned short *) &event_rec[2]; + if (dev < 0 || dev >= max_synthdev || synth_devs[dev] == NULL) + return -ENXIO; + + if (!(synth_open_mask & (1 << dev))) + return -ENXIO; + + fmt = (*(short *) &event_rec[0]) & 0xffff; + err = synth_devs[dev]->load_patch(dev, fmt, buf + p, c, 0); + if (err < 0) + return err; + + return err; + } + if (ev_code >= 128) + { + if (seq_mode == SEQ_2 && ev_code == SEQ_EXTENDED) + { + printk(KERN_WARNING "Sequencer: Invalid level 2 event %x\n", ev_code); + return -EINVAL; + } + ev_size = 8; + + if (c < ev_size) + { + if (!seq_playing) + seq_startplay(); + return count - c; + } + if (copy_from_user((char *)&event_rec[4], + &(buf)[p + 4], 4)) + goto out; + + } + else + { + if (seq_mode == SEQ_2) + { + printk(KERN_WARNING "Sequencer: 4 byte event in level 2 mode\n"); + return -EINVAL; + } + ev_size = 4; + + if (event_rec[0] != SEQ_MIDIPUTC) + obsolete_api_used = 1; + } + + if (event_rec[0] == SEQ_MIDIPUTC) + { + if (!midi_opened[event_rec[2]]) + { + int err, mode; + int dev = event_rec[2]; + + if (dev >= max_mididev || midi_devs[dev]==NULL) + { + /*printk("Sequencer Error: Nonexistent MIDI device %d\n", dev);*/ + return -ENXIO; + } + mode = translate_mode(file); + + if ((err = midi_devs[dev]->open(dev, mode, + sequencer_midi_input, sequencer_midi_output)) < 0) + { + seq_reset(); + printk(KERN_WARNING "Sequencer Error: Unable to open Midi #%d\n", dev); + return err; + } + midi_opened[dev] = 1; + } + } + if (!seq_queue(event_rec, (file->f_flags & (O_NONBLOCK) ? 1 : 0))) + { + int processed = count - c; + + if (!seq_playing) + seq_startplay(); + + if (!processed && (file->f_flags & O_NONBLOCK)) + return -EAGAIN; + else + return processed; + } + p += ev_size; + c -= ev_size; + } + + if (!seq_playing) + seq_startplay(); +out: + return count; +} + +static int seq_queue(unsigned char *note, char nonblock) +{ + + /* + * Test if there is space in the queue + */ + + if (qlen >= SEQ_MAX_QUEUE) + if (!seq_playing) + seq_startplay(); /* + * Give chance to drain the queue + */ + + if (!nonblock && qlen >= SEQ_MAX_QUEUE && !waitqueue_active(&seq_sleeper)) { + /* + * Sleep until there is enough space on the queue + */ + interruptible_sleep_on(&seq_sleeper); + } + if (qlen >= SEQ_MAX_QUEUE) + { + return 0; /* + * To be sure + */ + } + memcpy(&queue[qtail * EV_SZ], note, EV_SZ); + + qtail = (qtail + 1) % SEQ_MAX_QUEUE; + qlen++; + + return 1; +} + +static int extended_event(unsigned char *q) +{ + int dev = q[2]; + + if (dev < 0 || dev >= max_synthdev) + return -ENXIO; + + if (!(synth_open_mask & (1 << dev))) + return -ENXIO; + + switch (q[1]) + { + case SEQ_NOTEOFF: + synth_devs[dev]->kill_note(dev, q[3], q[4], q[5]); + break; + + case SEQ_NOTEON: + if (q[4] > 127 && q[4] != 255) + return 0; + + if (q[5] == 0) + { + synth_devs[dev]->kill_note(dev, q[3], q[4], q[5]); + break; + } + synth_devs[dev]->start_note(dev, q[3], q[4], q[5]); + break; + + case SEQ_PGMCHANGE: + synth_devs[dev]->set_instr(dev, q[3], q[4]); + break; + + case SEQ_AFTERTOUCH: + synth_devs[dev]->aftertouch(dev, q[3], q[4]); + break; + + case SEQ_BALANCE: + synth_devs[dev]->panning(dev, q[3], (char) q[4]); + break; + + case SEQ_CONTROLLER: + synth_devs[dev]->controller(dev, q[3], q[4], (short) (q[5] | (q[6] << 8))); + break; + + case SEQ_VOLMODE: + if (synth_devs[dev]->volume_method != NULL) + synth_devs[dev]->volume_method(dev, q[3]); + break; + + default: + return -EINVAL; + } + return 0; +} + +static int find_voice(int dev, int chn, int note) +{ + unsigned short key; + int i; + + key = (chn << 8) | (note + 1); + for (i = 0; i < synth_devs[dev]->alloc.max_voice; i++) + if (synth_devs[dev]->alloc.map[i] == key) + return i; + return -1; +} + +static int alloc_voice(int dev, int chn, int note) +{ + unsigned short key; + int voice; + + key = (chn << 8) | (note + 1); + + voice = synth_devs[dev]->alloc_voice(dev, chn, note, + &synth_devs[dev]->alloc); + synth_devs[dev]->alloc.map[voice] = key; + synth_devs[dev]->alloc.alloc_times[voice] = + synth_devs[dev]->alloc.timestamp++; + return voice; +} + +static void seq_chn_voice_event(unsigned char *event_rec) +{ +#define dev event_rec[1] +#define cmd event_rec[2] +#define chn event_rec[3] +#define note event_rec[4] +#define parm event_rec[5] + + int voice = -1; + + if ((int) dev > max_synthdev || synth_devs[dev] == NULL) + return; + if (!(synth_open_mask & (1 << dev))) + return; + if (!synth_devs[dev]) + return; + + if (seq_mode == SEQ_2) + { + if (synth_devs[dev]->alloc_voice) + voice = find_voice(dev, chn, note); + + if (cmd == MIDI_NOTEON && parm == 0) + { + cmd = MIDI_NOTEOFF; + parm = 64; + } + } + + switch (cmd) + { + case MIDI_NOTEON: + if (note > 127 && note != 255) /* Not a seq2 feature */ + return; + + if (voice == -1 && seq_mode == SEQ_2 && synth_devs[dev]->alloc_voice) + { + /* Internal synthesizer (FM, GUS, etc) */ + voice = alloc_voice(dev, chn, note); + } + if (voice == -1) + voice = chn; + + if (seq_mode == SEQ_2 && (int) dev < num_synths) + { + /* + * The MIDI channel 10 is a percussive channel. Use the note + * number to select the proper patch (128 to 255) to play. + */ + + if (chn == 9) + { + synth_devs[dev]->set_instr(dev, voice, 128 + note); + synth_devs[dev]->chn_info[chn].pgm_num = 128 + note; + } + synth_devs[dev]->setup_voice(dev, voice, chn); + } + synth_devs[dev]->start_note(dev, voice, note, parm); + break; + + case MIDI_NOTEOFF: + if (voice == -1) + voice = chn; + synth_devs[dev]->kill_note(dev, voice, note, parm); + break; + + case MIDI_KEY_PRESSURE: + if (voice == -1) + voice = chn; + synth_devs[dev]->aftertouch(dev, voice, parm); + break; + + default:; + } +#undef dev +#undef cmd +#undef chn +#undef note +#undef parm +} + + +static void seq_chn_common_event(unsigned char *event_rec) +{ + unsigned char dev = event_rec[1]; + unsigned char cmd = event_rec[2]; + unsigned char chn = event_rec[3]; + unsigned char p1 = event_rec[4]; + + /* unsigned char p2 = event_rec[5]; */ + unsigned short w14 = *(short *) &event_rec[6]; + + if ((int) dev > max_synthdev || synth_devs[dev] == NULL) + return; + if (!(synth_open_mask & (1 << dev))) + return; + if (!synth_devs[dev]) + return; + + switch (cmd) + { + case MIDI_PGM_CHANGE: + if (seq_mode == SEQ_2) + { + synth_devs[dev]->chn_info[chn].pgm_num = p1; + if ((int) dev >= num_synths) + synth_devs[dev]->set_instr(dev, chn, p1); + } + else + synth_devs[dev]->set_instr(dev, chn, p1); + + break; + + case MIDI_CTL_CHANGE: + if (seq_mode == SEQ_2) + { + if (chn > 15 || p1 > 127) + break; + + synth_devs[dev]->chn_info[chn].controllers[p1] = w14 & 0x7f; + + if (p1 < 32) /* Setting MSB should clear LSB to 0 */ + synth_devs[dev]->chn_info[chn].controllers[p1 + 32] = 0; + + if ((int) dev < num_synths) + { + int val = w14 & 0x7f; + int i, key; + + if (p1 < 64) /* Combine MSB and LSB */ + { + val = ((synth_devs[dev]-> + chn_info[chn].controllers[p1 & ~32] & 0x7f) << 7) + | (synth_devs[dev]-> + chn_info[chn].controllers[p1 | 32] & 0x7f); + p1 &= ~32; + } + /* Handle all playing notes on this channel */ + + key = ((int) chn << 8); + + for (i = 0; i < synth_devs[dev]->alloc.max_voice; i++) + if ((synth_devs[dev]->alloc.map[i] & 0xff00) == key) + synth_devs[dev]->controller(dev, i, p1, val); + } + else + synth_devs[dev]->controller(dev, chn, p1, w14); + } + else /* Mode 1 */ + synth_devs[dev]->controller(dev, chn, p1, w14); + break; + + case MIDI_PITCH_BEND: + if (seq_mode == SEQ_2) + { + synth_devs[dev]->chn_info[chn].bender_value = w14; + + if ((int) dev < num_synths) + { + /* Handle all playing notes on this channel */ + int i, key; + + key = (chn << 8); + + for (i = 0; i < synth_devs[dev]->alloc.max_voice; i++) + if ((synth_devs[dev]->alloc.map[i] & 0xff00) == key) + synth_devs[dev]->bender(dev, i, w14); + } + else + synth_devs[dev]->bender(dev, chn, w14); + } + else /* MODE 1 */ + synth_devs[dev]->bender(dev, chn, w14); + break; + + default:; + } +} + +static int seq_timing_event(unsigned char *event_rec) +{ + unsigned char cmd = event_rec[1]; + unsigned int parm = *(int *) &event_rec[4]; + + if (seq_mode == SEQ_2) + { + int ret; + + if ((ret = tmr->event(tmr_no, event_rec)) == TIMER_ARMED) + if ((SEQ_MAX_QUEUE - qlen) >= output_threshold) + wake_up(&seq_sleeper); + return ret; + } + switch (cmd) + { + case TMR_WAIT_REL: + parm += prev_event_time; + + /* + * NOTE! No break here. Execution of TMR_WAIT_REL continues in the + * next case (TMR_WAIT_ABS) + */ + + case TMR_WAIT_ABS: + if (parm > 0) + { + long time; + + time = parm; + prev_event_time = time; + + seq_playing = 1; + request_sound_timer(time); + + if ((SEQ_MAX_QUEUE - qlen) >= output_threshold) + wake_up(&seq_sleeper); + return TIMER_ARMED; + } + break; + + case TMR_START: + seq_time = jiffies; + prev_input_time = 0; + prev_event_time = 0; + break; + + case TMR_STOP: + break; + + case TMR_CONTINUE: + break; + + case TMR_TEMPO: + break; + + case TMR_ECHO: + if (seq_mode == SEQ_2) + seq_copy_to_input(event_rec, 8); + else + { + parm = (parm << 8 | SEQ_ECHO); + seq_copy_to_input((unsigned char *) &parm, 4); + } + break; + + default:; + } + + return TIMER_NOT_ARMED; +} + +static void seq_local_event(unsigned char *event_rec) +{ + unsigned char cmd = event_rec[1]; + unsigned int parm = *((unsigned int *) &event_rec[4]); + + switch (cmd) + { + case LOCL_STARTAUDIO: + DMAbuf_start_devices(parm); + break; + + default:; + } +} + +static void seq_sysex_message(unsigned char *event_rec) +{ + unsigned int dev = event_rec[1]; + int i, l = 0; + unsigned char *buf = &event_rec[2]; + + if (dev > max_synthdev) + return; + if (!(synth_open_mask & (1 << dev))) + return; + if (!synth_devs[dev]) + return; + + l = 0; + for (i = 0; i < 6 && buf[i] != 0xff; i++) + l = i + 1; + + if (!synth_devs[dev]->send_sysex) + return; + if (l > 0) + synth_devs[dev]->send_sysex(dev, buf, l); +} + +static int play_event(unsigned char *q) +{ + /* + * NOTE! This routine returns + * 0 = normal event played. + * 1 = Timer armed. Suspend playback until timer callback. + * 2 = MIDI output buffer full. Restore queue and suspend until timer + */ + unsigned int *delay; + + switch (q[0]) + { + case SEQ_NOTEOFF: + if (synth_open_mask & (1 << 0)) + if (synth_devs[0]) + synth_devs[0]->kill_note(0, q[1], 255, q[3]); + break; + + case SEQ_NOTEON: + if (q[4] < 128 || q[4] == 255) + if (synth_open_mask & (1 << 0)) + if (synth_devs[0]) + synth_devs[0]->start_note(0, q[1], q[2], q[3]); + break; + + case SEQ_WAIT: + delay = (unsigned int *) q; /* + * Bytes 1 to 3 are containing the * + * delay in 'ticks' + */ + *delay = (*delay >> 8) & 0xffffff; + + if (*delay > 0) + { + long time; + + seq_playing = 1; + time = *delay; + prev_event_time = time; + + request_sound_timer(time); + + if ((SEQ_MAX_QUEUE - qlen) >= output_threshold) + wake_up(&seq_sleeper); + /* + * The timer is now active and will reinvoke this function + * after the timer expires. Return to the caller now. + */ + return 1; + } + break; + + case SEQ_PGMCHANGE: + if (synth_open_mask & (1 << 0)) + if (synth_devs[0]) + synth_devs[0]->set_instr(0, q[1], q[2]); + break; + + case SEQ_SYNCTIMER: /* + * Reset timer + */ + seq_time = jiffies; + prev_input_time = 0; + prev_event_time = 0; + break; + + case SEQ_MIDIPUTC: /* + * Put a midi character + */ + if (midi_opened[q[2]]) + { + int dev; + + dev = q[2]; + + if (dev < 0 || dev >= num_midis || midi_devs[dev] == NULL) + break; + + if (!midi_devs[dev]->outputc(dev, q[1])) + { + /* + * Output FIFO is full. Wait one timer cycle and try again. + */ + + seq_playing = 1; + request_sound_timer(-1); + return 2; + } + else + midi_written[dev] = 1; + } + break; + + case SEQ_ECHO: + seq_copy_to_input(q, 4); /* + * Echo back to the process + */ + break; + + case SEQ_PRIVATE: + if ((int) q[1] < max_synthdev) + synth_devs[q[1]]->hw_control(q[1], q); + break; + + case SEQ_EXTENDED: + extended_event(q); + break; + + case EV_CHN_VOICE: + seq_chn_voice_event(q); + break; + + case EV_CHN_COMMON: + seq_chn_common_event(q); + break; + + case EV_TIMING: + if (seq_timing_event(q) == TIMER_ARMED) + { + return 1; + } + break; + + case EV_SEQ_LOCAL: + seq_local_event(q); + break; + + case EV_SYSEX: + seq_sysex_message(q); + break; + + default:; + } + return 0; +} + +/* called also as timer in irq context */ +static void seq_startplay(void) +{ + int this_one, action; + unsigned long flags; + + while (qlen > 0) + { + + spin_lock_irqsave(&lock,flags); + qhead = ((this_one = qhead) + 1) % SEQ_MAX_QUEUE; + qlen--; + spin_unlock_irqrestore(&lock,flags); + + seq_playing = 1; + + if ((action = play_event(&queue[this_one * EV_SZ]))) + { /* Suspend playback. Next timer routine invokes this routine again */ + if (action == 2) + { + qlen++; + qhead = this_one; + } + return; + } + } + + seq_playing = 0; + + if ((SEQ_MAX_QUEUE - qlen) >= output_threshold) + wake_up(&seq_sleeper); +} + +static void reset_controllers(int dev, unsigned char *controller, int update_dev) +{ + int i; + for (i = 0; i < 128; i++) + controller[i] = ctrl_def_values[i]; +} + +static void setup_mode2(void) +{ + int dev; + + max_synthdev = num_synths; + + for (dev = 0; dev < num_midis; dev++) + { + if (midi_devs[dev] && midi_devs[dev]->converter != NULL) + { + synth_devs[max_synthdev++] = midi_devs[dev]->converter; + } + } + + for (dev = 0; dev < max_synthdev; dev++) + { + int chn; + + synth_devs[dev]->sysex_ptr = 0; + synth_devs[dev]->emulation = 0; + + for (chn = 0; chn < 16; chn++) + { + synth_devs[dev]->chn_info[chn].pgm_num = 0; + reset_controllers(dev, + synth_devs[dev]->chn_info[chn].controllers,0); + synth_devs[dev]->chn_info[chn].bender_value = (1 << 7); /* Neutral */ + synth_devs[dev]->chn_info[chn].bender_range = 200; + } + } + max_mididev = 0; + seq_mode = SEQ_2; +} + +int sequencer_open(int dev, struct file *file) +{ + int retval, mode, i; + int level, tmp; + + if (!sequencer_ok) + sequencer_init(); + + level = ((dev & 0x0f) == SND_DEV_SEQ2) ? 2 : 1; + + dev = dev >> 4; + mode = translate_mode(file); + + DEB(printk("sequencer_open(dev=%d)\n", dev)); + + if (!sequencer_ok) + { +/* printk("Sound card: sequencer not initialized\n");*/ + return -ENXIO; + } + if (dev) /* Patch manager device (obsolete) */ + return -ENXIO; + + if(synth_devs[dev] == NULL) + request_module("synth0"); + + if (mode == OPEN_READ) + { + if (!num_midis) + { + /*printk("Sequencer: No MIDI devices. Input not possible\n");*/ + sequencer_busy = 0; + return -ENXIO; + } + } + if (sequencer_busy) + { + return -EBUSY; + } + sequencer_busy = 1; + obsolete_api_used = 0; + + max_mididev = num_midis; + max_synthdev = num_synths; + pre_event_timeout = MAX_SCHEDULE_TIMEOUT; + seq_mode = SEQ_1; + + if (pending_timer != -1) + { + tmr_no = pending_timer; + pending_timer = -1; + } + if (tmr_no == -1) /* Not selected yet */ + { + int i, best; + + best = -1; + for (i = 0; i < num_sound_timers; i++) + if (sound_timer_devs[i] && sound_timer_devs[i]->priority > best) + { + tmr_no = i; + best = sound_timer_devs[i]->priority; + } + if (tmr_no == -1) /* Should not be */ + tmr_no = 0; + } + tmr = sound_timer_devs[tmr_no]; + + if (level == 2) + { + if (tmr == NULL) + { + /*printk("sequencer: No timer for level 2\n");*/ + sequencer_busy = 0; + return -ENXIO; + } + setup_mode2(); + } + if (!max_synthdev && !max_mididev) + { + sequencer_busy=0; + return -ENXIO; + } + + synth_open_mask = 0; + + for (i = 0; i < max_mididev; i++) + { + midi_opened[i] = 0; + midi_written[i] = 0; + } + + for (i = 0; i < max_synthdev; i++) + { + if (synth_devs[i]==NULL) + continue; + + if (!try_module_get(synth_devs[i]->owner)) + continue; + + if ((tmp = synth_devs[i]->open(i, mode)) < 0) + { + printk(KERN_WARNING "Sequencer: Warning! Cannot open synth device #%d (%d)\n", i, tmp); + if (synth_devs[i]->midi_dev) + printk(KERN_WARNING "(Maps to MIDI dev #%d)\n", synth_devs[i]->midi_dev); + } + else + { + synth_open_mask |= (1 << i); + if (synth_devs[i]->midi_dev) + midi_opened[synth_devs[i]->midi_dev] = 1; + } + } + + seq_time = jiffies; + + prev_input_time = 0; + prev_event_time = 0; + + if (seq_mode == SEQ_1 && (mode == OPEN_READ || mode == OPEN_READWRITE)) + { + /* + * Initialize midi input devices + */ + + for (i = 0; i < max_mididev; i++) + if (!midi_opened[i] && midi_devs[i]) + { + if (!try_module_get(midi_devs[i]->owner)) + continue; + + if ((retval = midi_devs[i]->open(i, mode, + sequencer_midi_input, sequencer_midi_output)) >= 0) + { + midi_opened[i] = 1; + } + } + } + + if (seq_mode == SEQ_2) { + if (try_module_get(tmr->owner)) + tmr->open(tmr_no, seq_mode); + } + + init_waitqueue_head(&seq_sleeper); + init_waitqueue_head(&midi_sleeper); + output_threshold = SEQ_MAX_QUEUE / 2; + + return 0; +} + +static void seq_drain_midi_queues(void) +{ + int i, n; + + /* + * Give the Midi drivers time to drain their output queues + */ + + n = 1; + + while (!signal_pending(current) && n) + { + n = 0; + + for (i = 0; i < max_mididev; i++) + if (midi_opened[i] && midi_written[i]) + if (midi_devs[i]->buffer_status != NULL) + if (midi_devs[i]->buffer_status(i)) + n++; + + /* + * Let's have a delay + */ + + if (n) + interruptible_sleep_on_timeout(&seq_sleeper, + HZ/10); + } +} + +void sequencer_release(int dev, struct file *file) +{ + int i; + int mode = translate_mode(file); + + dev = dev >> 4; + + DEB(printk("sequencer_release(dev=%d)\n", dev)); + + /* + * Wait until the queue is empty (if we don't have nonblock) + */ + + if (mode != OPEN_READ && !(file->f_flags & O_NONBLOCK)) + { + while (!signal_pending(current) && qlen > 0) + { + seq_sync(); + interruptible_sleep_on_timeout(&seq_sleeper, + 3*HZ); + /* Extra delay */ + } + } + + if (mode != OPEN_READ) + seq_drain_midi_queues(); /* + * Ensure the output queues are empty + */ + seq_reset(); + if (mode != OPEN_READ) + seq_drain_midi_queues(); /* + * Flush the all notes off messages + */ + + for (i = 0; i < max_synthdev; i++) + { + if (synth_open_mask & (1 << i)) /* + * Actually opened + */ + if (synth_devs[i]) + { + synth_devs[i]->close(i); + + module_put(synth_devs[i]->owner); + + if (synth_devs[i]->midi_dev) + midi_opened[synth_devs[i]->midi_dev] = 0; + } + } + + for (i = 0; i < max_mididev; i++) + { + if (midi_opened[i]) { + midi_devs[i]->close(i); + module_put(midi_devs[i]->owner); + } + } + + if (seq_mode == SEQ_2) { + tmr->close(tmr_no); + module_put(tmr->owner); + } + + if (obsolete_api_used) + printk(KERN_WARNING "/dev/music: Obsolete (4 byte) API was used by %s\n", current->comm); + sequencer_busy = 0; +} + +static int seq_sync(void) +{ + if (qlen && !seq_playing && !signal_pending(current)) + seq_startplay(); + + if (qlen > 0) + interruptible_sleep_on_timeout(&seq_sleeper, HZ); + return qlen; +} + +static void midi_outc(int dev, unsigned char data) +{ + /* + * NOTE! Calls sleep(). Don't call this from interrupt. + */ + + int n; + unsigned long flags; + + /* + * This routine sends one byte to the Midi channel. + * If the output FIFO is full, it waits until there + * is space in the queue + */ + + n = 3 * HZ; /* Timeout */ + + spin_lock_irqsave(&lock,flags); + while (n && !midi_devs[dev]->outputc(dev, data)) { + interruptible_sleep_on_timeout(&seq_sleeper, HZ/25); + n--; + } + spin_unlock_irqrestore(&lock,flags); +} + +static void seq_reset(void) +{ + /* + * NOTE! Calls sleep(). Don't call this from interrupt. + */ + + int i; + int chn; + unsigned long flags; + + sound_stop_timer(); + + seq_time = jiffies; + prev_input_time = 0; + prev_event_time = 0; + + qlen = qhead = qtail = 0; + iqlen = iqhead = iqtail = 0; + + for (i = 0; i < max_synthdev; i++) + if (synth_open_mask & (1 << i)) + if (synth_devs[i]) + synth_devs[i]->reset(i); + + if (seq_mode == SEQ_2) + { + for (chn = 0; chn < 16; chn++) + for (i = 0; i < max_synthdev; i++) + if (synth_open_mask & (1 << i)) + if (synth_devs[i]) + { + synth_devs[i]->controller(i, chn, 123, 0); /* All notes off */ + synth_devs[i]->controller(i, chn, 121, 0); /* Reset all ctl */ + synth_devs[i]->bender(i, chn, 1 << 13); /* Bender off */ + } + } + else /* seq_mode == SEQ_1 */ + { + for (i = 0; i < max_mididev; i++) + if (midi_written[i]) /* + * Midi used. Some notes may still be playing + */ + { + /* + * Sending just a ACTIVE SENSING message should be enough to stop all + * playing notes. Since there are devices not recognizing the + * active sensing, we have to send some all notes off messages also. + */ + midi_outc(i, 0xfe); + + for (chn = 0; chn < 16; chn++) + { + midi_outc(i, (unsigned char) (0xb0 + (chn & 0x0f))); /* control change */ + midi_outc(i, 0x7b); /* All notes off */ + midi_outc(i, 0); /* Dummy parameter */ + } + + midi_devs[i]->close(i); + + midi_written[i] = 0; + midi_opened[i] = 0; + } + } + + seq_playing = 0; + + spin_lock_irqsave(&lock,flags); + + if (waitqueue_active(&seq_sleeper)) { + /* printk( "Sequencer Warning: Unexpected sleeping process - Waking up\n"); */ + wake_up(&seq_sleeper); + } + spin_unlock_irqrestore(&lock,flags); +} + +static void seq_panic(void) +{ + /* + * This routine is called by the application in case the user + * wants to reset the system to the default state. + */ + + seq_reset(); + + /* + * Since some of the devices don't recognize the active sensing and + * all notes off messages, we have to shut all notes manually. + * + * TO BE IMPLEMENTED LATER + */ + + /* + * Also return the controllers to their default states + */ +} + +int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *arg) +{ + int midi_dev, orig_dev, val, err; + int mode = translate_mode(file); + struct synth_info inf; + struct seq_event_rec event_rec; + unsigned long flags; + int __user *p = arg; + + orig_dev = dev = dev >> 4; + + switch (cmd) + { + case SNDCTL_TMR_TIMEBASE: + case SNDCTL_TMR_TEMPO: + case SNDCTL_TMR_START: + case SNDCTL_TMR_STOP: + case SNDCTL_TMR_CONTINUE: + case SNDCTL_TMR_METRONOME: + case SNDCTL_TMR_SOURCE: + if (seq_mode != SEQ_2) + return -EINVAL; + return tmr->ioctl(tmr_no, cmd, arg); + + case SNDCTL_TMR_SELECT: + if (seq_mode != SEQ_2) + return -EINVAL; + if (get_user(pending_timer, p)) + return -EFAULT; + if (pending_timer < 0 || pending_timer >= num_sound_timers || sound_timer_devs[pending_timer] == NULL) + { + pending_timer = -1; + return -EINVAL; + } + val = pending_timer; + break; + + case SNDCTL_SEQ_PANIC: + seq_panic(); + return -EINVAL; + + case SNDCTL_SEQ_SYNC: + if (mode == OPEN_READ) + return 0; + while (qlen > 0 && !signal_pending(current)) + seq_sync(); + return qlen ? -EINTR : 0; + + case SNDCTL_SEQ_RESET: + seq_reset(); + return 0; + + case SNDCTL_SEQ_TESTMIDI: + if (__get_user(midi_dev, p)) + return -EFAULT; + if (midi_dev < 0 || midi_dev >= max_mididev || !midi_devs[midi_dev]) + return -ENXIO; + + if (!midi_opened[midi_dev] && + (err = midi_devs[midi_dev]->open(midi_dev, mode, sequencer_midi_input, + sequencer_midi_output)) < 0) + return err; + midi_opened[midi_dev] = 1; + return 0; + + case SNDCTL_SEQ_GETINCOUNT: + if (mode == OPEN_WRITE) + return 0; + val = iqlen; + break; + + case SNDCTL_SEQ_GETOUTCOUNT: + if (mode == OPEN_READ) + return 0; + val = SEQ_MAX_QUEUE - qlen; + break; + + case SNDCTL_SEQ_GETTIME: + if (seq_mode == SEQ_2) + return tmr->ioctl(tmr_no, cmd, arg); + val = jiffies - seq_time; + break; + + case SNDCTL_SEQ_CTRLRATE: + /* + * If *arg == 0, just return the current rate + */ + if (seq_mode == SEQ_2) + return tmr->ioctl(tmr_no, cmd, arg); + + if (get_user(val, p)) + return -EFAULT; + if (val != 0) + return -EINVAL; + val = HZ; + break; + + case SNDCTL_SEQ_RESETSAMPLES: + case SNDCTL_SYNTH_REMOVESAMPLE: + case SNDCTL_SYNTH_CONTROL: + if (get_user(dev, p)) + return -EFAULT; + if (dev < 0 || dev >= num_synths || synth_devs[dev] == NULL) + return -ENXIO; + if (!(synth_open_mask & (1 << dev)) && !orig_dev) + return -EBUSY; + return synth_devs[dev]->ioctl(dev, cmd, arg); + + case SNDCTL_SEQ_NRSYNTHS: + val = max_synthdev; + break; + + case SNDCTL_SEQ_NRMIDIS: + val = max_mididev; + break; + + case SNDCTL_SYNTH_MEMAVL: + if (get_user(dev, p)) + return -EFAULT; + if (dev < 0 || dev >= num_synths || synth_devs[dev] == NULL) + return -ENXIO; + if (!(synth_open_mask & (1 << dev)) && !orig_dev) + return -EBUSY; + val = synth_devs[dev]->ioctl(dev, cmd, arg); + break; + + case SNDCTL_FM_4OP_ENABLE: + if (get_user(dev, p)) + return -EFAULT; + if (dev < 0 || dev >= num_synths || synth_devs[dev] == NULL) + return -ENXIO; + if (!(synth_open_mask & (1 << dev))) + return -ENXIO; + synth_devs[dev]->ioctl(dev, cmd, arg); + return 0; + + case SNDCTL_SYNTH_INFO: + if (get_user(dev, &((struct synth_info __user *)arg)->device)) + return -EFAULT; + if (dev < 0 || dev >= max_synthdev) + return -ENXIO; + if (!(synth_open_mask & (1 << dev)) && !orig_dev) + return -EBUSY; + return synth_devs[dev]->ioctl(dev, cmd, arg); + + /* Like SYNTH_INFO but returns ID in the name field */ + case SNDCTL_SYNTH_ID: + if (get_user(dev, &((struct synth_info __user *)arg)->device)) + return -EFAULT; + if (dev < 0 || dev >= max_synthdev) + return -ENXIO; + if (!(synth_open_mask & (1 << dev)) && !orig_dev) + return -EBUSY; + memcpy(&inf, synth_devs[dev]->info, sizeof(inf)); + strlcpy(inf.name, synth_devs[dev]->id, sizeof(inf.name)); + inf.device = dev; + return copy_to_user(arg, &inf, sizeof(inf))?-EFAULT:0; + + case SNDCTL_SEQ_OUTOFBAND: + if (copy_from_user(&event_rec, arg, sizeof(event_rec))) + return -EFAULT; + spin_lock_irqsave(&lock,flags); + play_event(event_rec.arr); + spin_unlock_irqrestore(&lock,flags); + return 0; + + case SNDCTL_MIDI_INFO: + if (get_user(dev, &((struct midi_info __user *)arg)->device)) + return -EFAULT; + if (dev < 0 || dev >= max_mididev || !midi_devs[dev]) + return -ENXIO; + midi_devs[dev]->info.device = dev; + return copy_to_user(arg, &midi_devs[dev]->info, sizeof(struct midi_info))?-EFAULT:0; + + case SNDCTL_SEQ_THRESHOLD: + if (get_user(val, p)) + return -EFAULT; + if (val < 1) + val = 1; + if (val >= SEQ_MAX_QUEUE) + val = SEQ_MAX_QUEUE - 1; + output_threshold = val; + return 0; + + case SNDCTL_MIDI_PRETIME: + if (get_user(val, p)) + return -EFAULT; + if (val < 0) + val = 0; + val = (HZ * val) / 10; + pre_event_timeout = val; + break; + + default: + if (mode == OPEN_READ) + return -EIO; + if (!synth_devs[0]) + return -ENXIO; + if (!(synth_open_mask & (1 << 0))) + return -ENXIO; + if (!synth_devs[0]->ioctl) + return -EINVAL; + return synth_devs[0]->ioctl(0, cmd, arg); + } + return put_user(val, p); +} + +/* No kernel lock - we're using the global irq lock here */ +unsigned int sequencer_poll(int dev, struct file *file, poll_table * wait) +{ + unsigned long flags; + unsigned int mask = 0; + + dev = dev >> 4; + + spin_lock_irqsave(&lock,flags); + /* input */ + poll_wait(file, &midi_sleeper, wait); + if (iqlen) + mask |= POLLIN | POLLRDNORM; + + /* output */ + poll_wait(file, &seq_sleeper, wait); + if ((SEQ_MAX_QUEUE - qlen) >= output_threshold) + mask |= POLLOUT | POLLWRNORM; + spin_unlock_irqrestore(&lock,flags); + return mask; +} + + +void sequencer_timer(unsigned long dummy) +{ + seq_startplay(); +} +EXPORT_SYMBOL(sequencer_timer); + +int note_to_freq(int note_num) +{ + + /* + * This routine converts a midi note to a frequency (multiplied by 1000) + */ + + int note, octave, note_freq; + static int notes[] = + { + 261632, 277189, 293671, 311132, 329632, 349232, + 369998, 391998, 415306, 440000, 466162, 493880 + }; + +#define BASE_OCTAVE 5 + + octave = note_num / 12; + note = note_num % 12; + + note_freq = notes[note]; + + if (octave < BASE_OCTAVE) + note_freq >>= (BASE_OCTAVE - octave); + else if (octave > BASE_OCTAVE) + note_freq <<= (octave - BASE_OCTAVE); + + /* + * note_freq >>= 1; + */ + + return note_freq; +} +EXPORT_SYMBOL(note_to_freq); + +unsigned long compute_finetune(unsigned long base_freq, int bend, int range, + int vibrato_cents) +{ + unsigned long amount; + int negative, semitones, cents, multiplier = 1; + + if (!bend) + return base_freq; + if (!range) + return base_freq; + + if (!base_freq) + return base_freq; + + if (range >= 8192) + range = 8192; + + bend = bend * range / 8192; /* Convert to cents */ + bend += vibrato_cents; + + if (!bend) + return base_freq; + + negative = bend < 0 ? 1 : 0; + + if (bend < 0) + bend *= -1; + if (bend > range) + bend = range; + + /* + if (bend > 2399) + bend = 2399; + */ + while (bend > 2399) + { + multiplier *= 4; + bend -= 2400; + } + + semitones = bend / 100; + cents = bend % 100; + + amount = (int) (semitone_tuning[semitones] * multiplier * cent_tuning[cents]) / 10000; + + if (negative) + return (base_freq * 10000) / amount; /* Bend down */ + else + return (base_freq * amount) / 10000; /* Bend up */ +} +EXPORT_SYMBOL(compute_finetune); + +void sequencer_init(void) +{ + if (sequencer_ok) + return; + queue = vmalloc(SEQ_MAX_QUEUE * EV_SZ); + if (queue == NULL) + { + printk(KERN_ERR "sequencer: Can't allocate memory for sequencer output queue\n"); + return; + } + iqueue = vmalloc(SEQ_MAX_QUEUE * IEV_SZ); + if (iqueue == NULL) + { + printk(KERN_ERR "sequencer: Can't allocate memory for sequencer input queue\n"); + vfree(queue); + return; + } + sequencer_ok = 1; +} +EXPORT_SYMBOL(sequencer_init); + +void sequencer_unload(void) +{ + vfree(queue); + vfree(iqueue); + queue = iqueue = NULL; +} diff --git a/sound/oss/sound_calls.h b/sound/oss/sound_calls.h new file mode 100644 index 00000000..87d8ad4a --- /dev/null +++ b/sound/oss/sound_calls.h @@ -0,0 +1,87 @@ +/* + * DMA buffer calls + */ + +int DMAbuf_open(int dev, int mode); +int DMAbuf_release(int dev, int mode); +int DMAbuf_getwrbuffer(int dev, char **buf, int *size, int dontblock); +int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock); +int DMAbuf_rmchars(int dev, int buff_no, int c); +int DMAbuf_start_output(int dev, int buff_no, int l); +int DMAbuf_move_wrpointer(int dev, int l); +/* int DMAbuf_ioctl(int dev, unsigned int cmd, void __user *arg, int local); */ +void DMAbuf_init(int dev, int dma1, int dma2); +void DMAbuf_deinit(int dev); +int DMAbuf_start_dma (int dev, unsigned long physaddr, int count, int dma_mode); +void DMAbuf_inputintr(int dev); +void DMAbuf_outputintr(int dev, int underflow_flag); +struct dma_buffparms; +int DMAbuf_space_in_queue (int dev); +int DMAbuf_activate_recording (int dev, struct dma_buffparms *dmap); +int DMAbuf_get_buffer_pointer (int dev, struct dma_buffparms *dmap, int direction); +void DMAbuf_launch_output(int dev, struct dma_buffparms *dmap); +unsigned int DMAbuf_poll(struct file *file, int dev, poll_table *wait); +void DMAbuf_start_devices(unsigned int devmask); +void DMAbuf_reset (int dev); +int DMAbuf_sync (int dev); + +/* + * System calls for /dev/dsp and /dev/audio (audio.c) + */ + +int audio_read (int dev, struct file *file, char __user *buf, int count); +int audio_write (int dev, struct file *file, const char __user *buf, int count); +int audio_open (int dev, struct file *file); +void audio_release (int dev, struct file *file); +int audio_ioctl (int dev, struct file *file, + unsigned int cmd, void __user *arg); +void audio_init_devices (void); +void reorganize_buffers (int dev, struct dma_buffparms *dmap, int recording); + +/* + * System calls for the /dev/sequencer + */ + +int sequencer_read (int dev, struct file *file, char __user *buf, int count); +int sequencer_write (int dev, struct file *file, const char __user *buf, int count); +int sequencer_open (int dev, struct file *file); +void sequencer_release (int dev, struct file *file); +int sequencer_ioctl (int dev, struct file *file, unsigned int cmd, void __user *arg); +unsigned int sequencer_poll(int dev, struct file *file, poll_table * wait); + +void sequencer_init (void); +void sequencer_unload (void); +void sequencer_timer(unsigned long dummy); +int note_to_freq(int note_num); +unsigned long compute_finetune(unsigned long base_freq, int bend, int range, + int vibrato_bend); +void seq_input_event(unsigned char *event, int len); +void seq_copy_to_input (unsigned char *event, int len); + +/* + * System calls for the /dev/midi + */ + +int MIDIbuf_read (int dev, struct file *file, char __user *buf, int count); +int MIDIbuf_write (int dev, struct file *file, const char __user *buf, int count); +int MIDIbuf_open (int dev, struct file *file); +void MIDIbuf_release (int dev, struct file *file); +int MIDIbuf_ioctl (int dev, struct file *file, unsigned int cmd, void __user *arg); +unsigned int MIDIbuf_poll(int dev, struct file *file, poll_table * wait); +int MIDIbuf_avail(int dev); + +void MIDIbuf_bytes_received(int dev, unsigned char *buf, int count); + + +/* From soundcard.c */ +void request_sound_timer (int count); +void sound_stop_timer(void); +void conf_printf(char *name, struct address_info *hw_config); +void conf_printf2(char *name, int base, int irq, int dma, int dma2); + +/* From sound_timer.c */ +void sound_timer_interrupt(void); +void sound_timer_syncinterval(unsigned int new_usecs); + +/* From midi_synth.c */ +void do_midi_msg (int synthno, unsigned char *msg, int mlen); diff --git a/sound/oss/sound_config.h b/sound/oss/sound_config.h new file mode 100644 index 00000000..9d35c4c6 --- /dev/null +++ b/sound/oss/sound_config.h @@ -0,0 +1,147 @@ +/* sound_config.h + * + * A driver for sound cards, misc. configuration parameters. + */ +/* + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + */ + + +#ifndef _SOUND_CONFIG_H_ +#define _SOUND_CONFIG_H_ + +#include <linux/fs.h> +#include <linux/sound.h> + +#include "os.h" +#include "soundvers.h" + + +#ifndef SND_DEFAULT_ENABLE +#define SND_DEFAULT_ENABLE 1 +#endif + +#ifndef MAX_REALTIME_FACTOR +#define MAX_REALTIME_FACTOR 4 +#endif + +/* + * Use always 64k buffer size. There is no reason to use shorter. + */ +#undef DSP_BUFFSIZE +#define DSP_BUFFSIZE (64*1024) + +#ifndef DSP_BUFFCOUNT +#define DSP_BUFFCOUNT 1 /* 1 is recommended. */ +#endif + +#define FM_MONO 0x388 /* This is the I/O address used by AdLib */ + +#ifndef CONFIG_PAS_BASE +#define CONFIG_PAS_BASE 0x388 +#endif + +/* SEQ_MAX_QUEUE is the maximum number of sequencer events buffered by the + driver. (There is no need to alter this) */ +#define SEQ_MAX_QUEUE 1024 + +#define SBFM_MAXINSTR (256) /* Size of the FM Instrument bank */ +/* 128 instruments for general MIDI setup and 16 unassigned */ + +#define SND_NDEVS 256 /* Number of supported devices */ + +#define DSP_DEFAULT_SPEED 8000 + +#define MAX_AUDIO_DEV 5 +#define MAX_MIXER_DEV 5 +#define MAX_SYNTH_DEV 5 +#define MAX_MIDI_DEV 6 +#define MAX_TIMER_DEV 4 + +struct address_info { + int io_base; + int irq; + int dma; + int dma2; + int always_detect; /* 1=Trust me, it's there */ + char *name; + int driver_use_1; /* Driver defined field 1 */ + int driver_use_2; /* Driver defined field 2 */ + int *osp; /* OS specific info */ + int card_subtype; /* Driver specific. Usually 0 */ + void *memptr; /* Module memory chainer */ + int slots[6]; /* To remember driver slot ids */ +}; + +#define SYNTH_MAX_VOICES 32 + +struct voice_alloc_info { + int max_voice; + int used_voices; + int ptr; /* For device specific use */ + unsigned short map[SYNTH_MAX_VOICES]; /* (ch << 8) | (note+1) */ + int timestamp; + int alloc_times[SYNTH_MAX_VOICES]; + }; + +struct channel_info { + int pgm_num; + int bender_value; + int bender_range; + unsigned char controllers[128]; + }; + +/* + * Process wakeup reasons + */ +#define WK_NONE 0x00 +#define WK_WAKEUP 0x01 +#define WK_TIMEOUT 0x02 +#define WK_SIGNAL 0x04 +#define WK_SLEEP 0x08 +#define WK_SELECT 0x10 +#define WK_ABORT 0x20 + +#define OPEN_READ PCM_ENABLE_INPUT +#define OPEN_WRITE PCM_ENABLE_OUTPUT +#define OPEN_READWRITE (OPEN_READ|OPEN_WRITE) + +static inline int translate_mode(struct file *file) +{ + if (OPEN_READ == (__force int)FMODE_READ && + OPEN_WRITE == (__force int)FMODE_WRITE) + return (__force int)(file->f_mode & (FMODE_READ | FMODE_WRITE)); + else + return ((file->f_mode & FMODE_READ) ? OPEN_READ : 0) | + ((file->f_mode & FMODE_WRITE) ? OPEN_WRITE : 0); +} + +#include "sound_calls.h" +#include "dev_table.h" + +#ifndef DEB +#define DEB(x) +#endif + +#ifndef DDB +#define DDB(x) do {} while (0) +#endif + +#ifndef MDB +#ifdef MODULE +#define MDB(x) x +#else +#define MDB(x) +#endif +#endif + +#define TIMER_ARMED 121234 +#define TIMER_NOT_ARMED 1 + +#define MAX_MEM_BLOCKS 1024 + +#endif diff --git a/sound/oss/sound_firmware.h b/sound/oss/sound_firmware.h new file mode 100644 index 00000000..0a0cbfdf --- /dev/null +++ b/sound/oss/sound_firmware.h @@ -0,0 +1,2 @@ +extern int mod_firmware_load(const char *fn, char **fp); + diff --git a/sound/oss/sound_timer.c b/sound/oss/sound_timer.c new file mode 100644 index 00000000..48cda6c4 --- /dev/null +++ b/sound/oss/sound_timer.c @@ -0,0 +1,327 @@ +/* + * sound/oss/sound_timer.c + */ +/* + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + */ +/* + * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed) + */ +#include <linux/string.h> +#include <linux/spinlock.h> + +#include "sound_config.h" + +static volatile int initialized, opened, tmr_running; +static volatile time_t tmr_offs, tmr_ctr; +static volatile unsigned long ticks_offs; +static volatile int curr_tempo, curr_timebase; +static volatile unsigned long curr_ticks; +static volatile unsigned long next_event_time; +static unsigned long prev_event_time; +static volatile unsigned long usecs_per_tmr; /* Length of the current interval */ + +static struct sound_lowlev_timer *tmr; +static DEFINE_SPINLOCK(lock); + +static unsigned long tmr2ticks(int tmr_value) +{ + /* + * Convert timer ticks to MIDI ticks + */ + + unsigned long tmp; + unsigned long scale; + + tmp = tmr_value * usecs_per_tmr; /* Convert to usecs */ + scale = (60 * 1000000) / (curr_tempo * curr_timebase); /* usecs per MIDI tick */ + return (tmp + (scale / 2)) / scale; +} + +void reprogram_timer(void) +{ + unsigned long usecs_per_tick; + + /* + * The user is changing the timer rate before setting a timer + * slap, bad bad not allowed. + */ + + if(!tmr) + return; + + usecs_per_tick = (60 * 1000000) / (curr_tempo * curr_timebase); + + /* + * Don't kill the system by setting too high timer rate + */ + if (usecs_per_tick < 2000) + usecs_per_tick = 2000; + + usecs_per_tmr = tmr->tmr_start(tmr->dev, usecs_per_tick); +} + +void sound_timer_syncinterval(unsigned int new_usecs) +{ + /* + * This routine is called by the hardware level if + * the clock frequency has changed for some reason. + */ + tmr_offs = tmr_ctr; + ticks_offs += tmr2ticks(tmr_ctr); + tmr_ctr = 0; + usecs_per_tmr = new_usecs; +} +EXPORT_SYMBOL(sound_timer_syncinterval); + +static void tmr_reset(void) +{ + unsigned long flags; + + spin_lock_irqsave(&lock,flags); + tmr_offs = 0; + ticks_offs = 0; + tmr_ctr = 0; + next_event_time = (unsigned long) -1; + prev_event_time = 0; + curr_ticks = 0; + spin_unlock_irqrestore(&lock,flags); +} + +static int timer_open(int dev, int mode) +{ + if (opened) + return -EBUSY; + tmr_reset(); + curr_tempo = 60; + curr_timebase = 100; + opened = 1; + reprogram_timer(); + return 0; +} + +static void timer_close(int dev) +{ + opened = tmr_running = 0; + tmr->tmr_disable(tmr->dev); +} + +static int timer_event(int dev, unsigned char *event) +{ + unsigned char cmd = event[1]; + unsigned long parm = *(int *) &event[4]; + + switch (cmd) + { + case TMR_WAIT_REL: + parm += prev_event_time; + case TMR_WAIT_ABS: + if (parm > 0) + { + long time; + + if (parm <= curr_ticks) /* It's the time */ + return TIMER_NOT_ARMED; + time = parm; + next_event_time = prev_event_time = time; + return TIMER_ARMED; + } + break; + + case TMR_START: + tmr_reset(); + tmr_running = 1; + reprogram_timer(); + break; + + case TMR_STOP: + tmr_running = 0; + break; + + case TMR_CONTINUE: + tmr_running = 1; + reprogram_timer(); + break; + + case TMR_TEMPO: + if (parm) + { + if (parm < 8) + parm = 8; + if (parm > 250) + parm = 250; + tmr_offs = tmr_ctr; + ticks_offs += tmr2ticks(tmr_ctr); + tmr_ctr = 0; + curr_tempo = parm; + reprogram_timer(); + } + break; + + case TMR_ECHO: + seq_copy_to_input(event, 8); + break; + + default:; + } + return TIMER_NOT_ARMED; +} + +static unsigned long timer_get_time(int dev) +{ + if (!opened) + return 0; + return curr_ticks; +} + +static int timer_ioctl(int dev, unsigned int cmd, void __user *arg) +{ + int __user *p = arg; + int val; + + switch (cmd) + { + case SNDCTL_TMR_SOURCE: + val = TMR_INTERNAL; + break; + + case SNDCTL_TMR_START: + tmr_reset(); + tmr_running = 1; + return 0; + + case SNDCTL_TMR_STOP: + tmr_running = 0; + return 0; + + case SNDCTL_TMR_CONTINUE: + tmr_running = 1; + return 0; + + case SNDCTL_TMR_TIMEBASE: + if (get_user(val, p)) + return -EFAULT; + if (val) + { + if (val < 1) + val = 1; + if (val > 1000) + val = 1000; + curr_timebase = val; + } + val = curr_timebase; + break; + + case SNDCTL_TMR_TEMPO: + if (get_user(val, p)) + return -EFAULT; + if (val) + { + if (val < 8) + val = 8; + if (val > 250) + val = 250; + tmr_offs = tmr_ctr; + ticks_offs += tmr2ticks(tmr_ctr); + tmr_ctr = 0; + curr_tempo = val; + reprogram_timer(); + } + val = curr_tempo; + break; + + case SNDCTL_SEQ_CTRLRATE: + if (get_user(val, p)) + return -EFAULT; + if (val != 0) /* Can't change */ + return -EINVAL; + val = ((curr_tempo * curr_timebase) + 30) / 60; + break; + + case SNDCTL_SEQ_GETTIME: + val = curr_ticks; + break; + + case SNDCTL_TMR_METRONOME: + default: + return -EINVAL; + } + return put_user(val, p); +} + +static void timer_arm(int dev, long time) +{ + if (time < 0) + time = curr_ticks + 1; + else if (time <= curr_ticks) /* It's the time */ + return; + + next_event_time = prev_event_time = time; + return; +} + +static struct sound_timer_operations sound_timer = +{ + .owner = THIS_MODULE, + .info = {"Sound Timer", 0}, + .priority = 1, /* Priority */ + .devlink = 0, /* Local device link */ + .open = timer_open, + .close = timer_close, + .event = timer_event, + .get_time = timer_get_time, + .ioctl = timer_ioctl, + .arm_timer = timer_arm +}; + +void sound_timer_interrupt(void) +{ + unsigned long flags; + + if (!opened) + return; + + tmr->tmr_restart(tmr->dev); + + if (!tmr_running) + return; + + spin_lock_irqsave(&lock,flags); + tmr_ctr++; + curr_ticks = ticks_offs + tmr2ticks(tmr_ctr); + + if (curr_ticks >= next_event_time) + { + next_event_time = (unsigned long) -1; + sequencer_timer(0); + } + spin_unlock_irqrestore(&lock,flags); +} +EXPORT_SYMBOL(sound_timer_interrupt); + +void sound_timer_init(struct sound_lowlev_timer *t, char *name) +{ + int n; + + if (initialized) + { + if (t->priority <= tmr->priority) + return; /* There is already a similar or better timer */ + tmr = t; + return; + } + initialized = 1; + tmr = t; + + n = sound_alloc_timerdev(); + if (n == -1) + n = 0; /* Overwrite the system timer */ + strcpy(sound_timer.info.name, name); + sound_timer_devs[n] = &sound_timer; +} +EXPORT_SYMBOL(sound_timer_init); + diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c new file mode 100644 index 00000000..7c7793a0 --- /dev/null +++ b/sound/oss/soundcard.c @@ -0,0 +1,739 @@ +/* + * linux/sound/oss/soundcard.c + * + * Sound card driver for Linux + * + * + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * + * + * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed) + * integrated sound_switch.c + * Stefan Reinauer : integrated /proc/sound (equals to /dev/sndstat, + * which should disappear in the near future) + * Eric Dumas : devfs support (22-Jan-98) <dumas@linux.eu.org> with + * fixups by C. Scott Ananian <cananian@alumni.princeton.edu> + * Richard Gooch : moved common (non OSS-specific) devices to sound_core.c + * Rob Riggs : Added persistent DMA buffers support (1998/10/17) + * Christoph Hellwig : Some cleanup work (2000/03/01) + */ + + +#include "sound_config.h" +#include <linux/init.h> +#include <linux/types.h> +#include <linux/errno.h> +#include <linux/signal.h> +#include <linux/fcntl.h> +#include <linux/ctype.h> +#include <linux/stddef.h> +#include <linux/kmod.h> +#include <linux/kernel.h> +#include <asm/dma.h> +#include <asm/io.h> +#include <linux/wait.h> +#include <linux/ioport.h> +#include <linux/major.h> +#include <linux/delay.h> +#include <linux/proc_fs.h> +#include <linux/mutex.h> +#include <linux/module.h> +#include <linux/mm.h> +#include <linux/device.h> + +/* + * This ought to be moved into include/asm/dma.h + */ +#ifndef valid_dma +#define valid_dma(n) ((n) >= 0 && (n) < MAX_DMA_CHANNELS && (n) != 4) +#endif + +/* + * Table for permanently allocated memory (used when unloading the module) + */ +void * sound_mem_blocks[MAX_MEM_BLOCKS]; +static DEFINE_MUTEX(soundcard_mutex); +int sound_nblocks = 0; + +/* Persistent DMA buffers */ +#ifdef CONFIG_SOUND_DMAP +int sound_dmap_flag = 1; +#else +int sound_dmap_flag = 0; +#endif + +static char dma_alloc_map[MAX_DMA_CHANNELS]; + +#define DMA_MAP_UNAVAIL 0 +#define DMA_MAP_FREE 1 +#define DMA_MAP_BUSY 2 + + +unsigned long seq_time = 0; /* Time for /dev/sequencer */ +extern struct class *sound_class; + +/* + * Table for configurable mixer volume handling + */ +static mixer_vol_table mixer_vols[MAX_MIXER_DEV]; +static int num_mixer_volumes; + +int *load_mixer_volumes(char *name, int *levels, int present) +{ + int i, n; + + for (i = 0; i < num_mixer_volumes; i++) { + if (strncmp(name, mixer_vols[i].name, 32) == 0) { + if (present) + mixer_vols[i].num = i; + return mixer_vols[i].levels; + } + } + if (num_mixer_volumes >= MAX_MIXER_DEV) { + printk(KERN_ERR "Sound: Too many mixers (%s)\n", name); + return levels; + } + n = num_mixer_volumes++; + + strncpy(mixer_vols[n].name, name, 32); + + if (present) + mixer_vols[n].num = n; + else + mixer_vols[n].num = -1; + + for (i = 0; i < 32; i++) + mixer_vols[n].levels[i] = levels[i]; + return mixer_vols[n].levels; +} +EXPORT_SYMBOL(load_mixer_volumes); + +static int set_mixer_levels(void __user * arg) +{ + /* mixer_vol_table is 174 bytes, so IMHO no reason to not allocate it on the stack */ + mixer_vol_table buf; + + if (__copy_from_user(&buf, arg, sizeof(buf))) + return -EFAULT; + load_mixer_volumes(buf.name, buf.levels, 0); + if (__copy_to_user(arg, &buf, sizeof(buf))) + return -EFAULT; + return 0; +} + +static int get_mixer_levels(void __user * arg) +{ + int n; + + if (__get_user(n, (int __user *)(&(((mixer_vol_table __user *)arg)->num)))) + return -EFAULT; + if (n < 0 || n >= num_mixer_volumes) + return -EINVAL; + if (__copy_to_user(arg, &mixer_vols[n], sizeof(mixer_vol_table))) + return -EFAULT; + return 0; +} + +/* 4K page size but our output routines use some slack for overruns */ +#define PROC_BLOCK_SIZE (3*1024) + +static ssize_t sound_read(struct file *file, char __user *buf, size_t count, loff_t *ppos) +{ + int dev = iminor(file->f_path.dentry->d_inode); + int ret = -EINVAL; + + /* + * The OSS drivers aren't remotely happy without this locking, + * and unless someone fixes them when they are about to bite the + * big one anyway, we might as well bandage here.. + */ + + mutex_lock(&soundcard_mutex); + + DEB(printk("sound_read(dev=%d, count=%d)\n", dev, count)); + switch (dev & 0x0f) { + case SND_DEV_DSP: + case SND_DEV_DSP16: + case SND_DEV_AUDIO: + ret = audio_read(dev, file, buf, count); + break; + + case SND_DEV_SEQ: + case SND_DEV_SEQ2: + ret = sequencer_read(dev, file, buf, count); + break; + + case SND_DEV_MIDIN: + ret = MIDIbuf_read(dev, file, buf, count); + } + mutex_unlock(&soundcard_mutex); + return ret; +} + +static ssize_t sound_write(struct file *file, const char __user *buf, size_t count, loff_t *ppos) +{ + int dev = iminor(file->f_path.dentry->d_inode); + int ret = -EINVAL; + + mutex_lock(&soundcard_mutex); + DEB(printk("sound_write(dev=%d, count=%d)\n", dev, count)); + switch (dev & 0x0f) { + case SND_DEV_SEQ: + case SND_DEV_SEQ2: + ret = sequencer_write(dev, file, buf, count); + break; + + case SND_DEV_DSP: + case SND_DEV_DSP16: + case SND_DEV_AUDIO: + ret = audio_write(dev, file, buf, count); + break; + + case SND_DEV_MIDIN: + ret = MIDIbuf_write(dev, file, buf, count); + break; + } + mutex_unlock(&soundcard_mutex); + return ret; +} + +static int sound_open(struct inode *inode, struct file *file) +{ + int dev = iminor(inode); + int retval; + + DEB(printk("sound_open(dev=%d)\n", dev)); + if ((dev >= SND_NDEVS) || (dev < 0)) { + printk(KERN_ERR "Invalid minor device %d\n", dev); + return -ENXIO; + } + mutex_lock(&soundcard_mutex); + switch (dev & 0x0f) { + case SND_DEV_CTL: + dev >>= 4; + if (dev >= 0 && dev < MAX_MIXER_DEV && mixer_devs[dev] == NULL) { + request_module("mixer%d", dev); + } + retval = -ENXIO; + if (dev && (dev >= num_mixers || mixer_devs[dev] == NULL)) + break; + + if (!try_module_get(mixer_devs[dev]->owner)) + break; + + retval = 0; + break; + + case SND_DEV_SEQ: + case SND_DEV_SEQ2: + retval = sequencer_open(dev, file); + break; + + case SND_DEV_MIDIN: + retval = MIDIbuf_open(dev, file); + break; + + case SND_DEV_DSP: + case SND_DEV_DSP16: + case SND_DEV_AUDIO: + retval = audio_open(dev, file); + break; + + default: + printk(KERN_ERR "Invalid minor device %d\n", dev); + retval = -ENXIO; + } + + mutex_unlock(&soundcard_mutex); + return retval; +} + +static int sound_release(struct inode *inode, struct file *file) +{ + int dev = iminor(inode); + + mutex_lock(&soundcard_mutex); + DEB(printk("sound_release(dev=%d)\n", dev)); + switch (dev & 0x0f) { + case SND_DEV_CTL: + module_put(mixer_devs[dev >> 4]->owner); + break; + + case SND_DEV_SEQ: + case SND_DEV_SEQ2: + sequencer_release(dev, file); + break; + + case SND_DEV_MIDIN: + MIDIbuf_release(dev, file); + break; + + case SND_DEV_DSP: + case SND_DEV_DSP16: + case SND_DEV_AUDIO: + audio_release(dev, file); + break; + + default: + printk(KERN_ERR "Sound error: Releasing unknown device 0x%02x\n", dev); + } + mutex_unlock(&soundcard_mutex); + + return 0; +} + +static int get_mixer_info(int dev, void __user *arg) +{ + mixer_info info; + memset(&info, 0, sizeof(info)); + strlcpy(info.id, mixer_devs[dev]->id, sizeof(info.id)); + strlcpy(info.name, mixer_devs[dev]->name, sizeof(info.name)); + info.modify_counter = mixer_devs[dev]->modify_counter; + if (__copy_to_user(arg, &info, sizeof(info))) + return -EFAULT; + return 0; +} + +static int get_old_mixer_info(int dev, void __user *arg) +{ + _old_mixer_info info; + memset(&info, 0, sizeof(info)); + strlcpy(info.id, mixer_devs[dev]->id, sizeof(info.id)); + strlcpy(info.name, mixer_devs[dev]->name, sizeof(info.name)); + if (copy_to_user(arg, &info, sizeof(info))) + return -EFAULT; + return 0; +} + +static int sound_mixer_ioctl(int mixdev, unsigned int cmd, void __user *arg) +{ + if (mixdev < 0 || mixdev >= MAX_MIXER_DEV) + return -ENXIO; + /* Try to load the mixer... */ + if (mixer_devs[mixdev] == NULL) { + request_module("mixer%d", mixdev); + } + if (mixdev >= num_mixers || !mixer_devs[mixdev]) + return -ENXIO; + if (cmd == SOUND_MIXER_INFO) + return get_mixer_info(mixdev, arg); + if (cmd == SOUND_OLD_MIXER_INFO) + return get_old_mixer_info(mixdev, arg); + if (_SIOC_DIR(cmd) & _SIOC_WRITE) + mixer_devs[mixdev]->modify_counter++; + if (!mixer_devs[mixdev]->ioctl) + return -EINVAL; + return mixer_devs[mixdev]->ioctl(mixdev, cmd, arg); +} + +static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg) +{ + int len = 0, dtype; + int dev = iminor(file->f_dentry->d_inode); + long ret = -EINVAL; + void __user *p = (void __user *)arg; + + if (_SIOC_DIR(cmd) != _SIOC_NONE && _SIOC_DIR(cmd) != 0) { + /* + * Have to validate the address given by the process. + */ + len = _SIOC_SIZE(cmd); + if (len < 1 || len > 65536 || !p) + return -EFAULT; + if (_SIOC_DIR(cmd) & _SIOC_WRITE) + if (!access_ok(VERIFY_READ, p, len)) + return -EFAULT; + if (_SIOC_DIR(cmd) & _SIOC_READ) + if (!access_ok(VERIFY_WRITE, p, len)) + return -EFAULT; + } + DEB(printk("sound_ioctl(dev=%d, cmd=0x%x, arg=0x%x)\n", dev, cmd, arg)); + if (cmd == OSS_GETVERSION) + return __put_user(SOUND_VERSION, (int __user *)p); + + mutex_lock(&soundcard_mutex); + if (_IOC_TYPE(cmd) == 'M' && num_mixers > 0 && /* Mixer ioctl */ + (dev & 0x0f) != SND_DEV_CTL) { + dtype = dev & 0x0f; + switch (dtype) { + case SND_DEV_DSP: + case SND_DEV_DSP16: + case SND_DEV_AUDIO: + ret = sound_mixer_ioctl(audio_devs[dev >> 4]->mixer_dev, + cmd, p); + break; + default: + ret = sound_mixer_ioctl(dev >> 4, cmd, p); + break; + } + mutex_unlock(&soundcard_mutex); + return ret; + } + + switch (dev & 0x0f) { + case SND_DEV_CTL: + if (cmd == SOUND_MIXER_GETLEVELS) + ret = get_mixer_levels(p); + else if (cmd == SOUND_MIXER_SETLEVELS) + ret = set_mixer_levels(p); + else + ret = sound_mixer_ioctl(dev >> 4, cmd, p); + break; + + case SND_DEV_SEQ: + case SND_DEV_SEQ2: + ret = sequencer_ioctl(dev, file, cmd, p); + break; + + case SND_DEV_DSP: + case SND_DEV_DSP16: + case SND_DEV_AUDIO: + ret = audio_ioctl(dev, file, cmd, p); + break; + + case SND_DEV_MIDIN: + ret = MIDIbuf_ioctl(dev, file, cmd, p); + break; + + } + mutex_unlock(&soundcard_mutex); + return ret; +} + +static unsigned int sound_poll(struct file *file, poll_table * wait) +{ + struct inode *inode = file->f_path.dentry->d_inode; + int dev = iminor(inode); + + DEB(printk("sound_poll(dev=%d)\n", dev)); + switch (dev & 0x0f) { + case SND_DEV_SEQ: + case SND_DEV_SEQ2: + return sequencer_poll(dev, file, wait); + + case SND_DEV_MIDIN: + return MIDIbuf_poll(dev, file, wait); + + case SND_DEV_DSP: + case SND_DEV_DSP16: + case SND_DEV_AUDIO: + return DMAbuf_poll(file, dev >> 4, wait); + } + return 0; +} + +static int sound_mmap(struct file *file, struct vm_area_struct *vma) +{ + int dev_class; + unsigned long size; + struct dma_buffparms *dmap = NULL; + int dev = iminor(file->f_path.dentry->d_inode); + + dev_class = dev & 0x0f; + dev >>= 4; + + if (dev_class != SND_DEV_DSP && dev_class != SND_DEV_DSP16 && dev_class != SND_DEV_AUDIO) { + printk(KERN_ERR "Sound: mmap() not supported for other than audio devices\n"); + return -EINVAL; + } + mutex_lock(&soundcard_mutex); + if (vma->vm_flags & VM_WRITE) /* Map write and read/write to the output buf */ + dmap = audio_devs[dev]->dmap_out; + else if (vma->vm_flags & VM_READ) + dmap = audio_devs[dev]->dmap_in; + else { + printk(KERN_ERR "Sound: Undefined mmap() access\n"); + mutex_unlock(&soundcard_mutex); + return -EINVAL; + } + + if (dmap == NULL) { + printk(KERN_ERR "Sound: mmap() error. dmap == NULL\n"); + mutex_unlock(&soundcard_mutex); + return -EIO; + } + if (dmap->raw_buf == NULL) { + printk(KERN_ERR "Sound: mmap() called when raw_buf == NULL\n"); + mutex_unlock(&soundcard_mutex); + return -EIO; + } + if (dmap->mapping_flags) { + printk(KERN_ERR "Sound: mmap() called twice for the same DMA buffer\n"); + mutex_unlock(&soundcard_mutex); + return -EIO; + } + if (vma->vm_pgoff != 0) { + printk(KERN_ERR "Sound: mmap() offset must be 0.\n"); + mutex_unlock(&soundcard_mutex); + return -EINVAL; + } + size = vma->vm_end - vma->vm_start; + + if (size != dmap->bytes_in_use) { + printk(KERN_WARNING "Sound: mmap() size = %ld. Should be %d\n", size, dmap->bytes_in_use); + } + if (remap_pfn_range(vma, vma->vm_start, + virt_to_phys(dmap->raw_buf) >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot)) { + mutex_unlock(&soundcard_mutex); + return -EAGAIN; + } + + dmap->mapping_flags |= DMA_MAP_MAPPED; + + if( audio_devs[dev]->d->mmap) + audio_devs[dev]->d->mmap(dev); + + memset(dmap->raw_buf, + dmap->neutral_byte, + dmap->bytes_in_use); + mutex_unlock(&soundcard_mutex); + return 0; +} + +const struct file_operations oss_sound_fops = { + .owner = THIS_MODULE, + .llseek = no_llseek, + .read = sound_read, + .write = sound_write, + .poll = sound_poll, + .unlocked_ioctl = sound_ioctl, + .mmap = sound_mmap, + .open = sound_open, + .release = sound_release, +}; + +/* + * Create the required special subdevices + */ + +static int create_special_devices(void) +{ + int seq1,seq2; + seq1=register_sound_special(&oss_sound_fops, 1); + if(seq1==-1) + goto bad; + seq2=register_sound_special(&oss_sound_fops, 8); + if(seq2!=-1) + return 0; + unregister_sound_special(1); +bad: + return -1; +} + + +static int dmabuf; +static int dmabug; + +module_param(dmabuf, int, 0444); +module_param(dmabug, int, 0444); + +/* additional minors for compatibility */ +struct oss_minor_dev { + unsigned short minor; + unsigned int enabled; +} dev_list[] = { + { SND_DEV_DSP16 }, + { SND_DEV_AUDIO }, +}; + +static int __init oss_init(void) +{ + int err; + int i, j; + +#ifdef CONFIG_PCI + if(dmabug) + isa_dma_bridge_buggy = dmabug; +#endif + + err = create_special_devices(); + if (err) { + printk(KERN_ERR "sound: driver already loaded/included in kernel\n"); + return err; + } + + /* Protecting the innocent */ + sound_dmap_flag = (dmabuf > 0 ? 1 : 0); + + for (i = 0; i < ARRAY_SIZE(dev_list); i++) { + j = 0; + do { + unsigned short minor = dev_list[i].minor + j * 0x10; + if (!register_sound_special(&oss_sound_fops, minor)) + dev_list[i].enabled = (1 << j); + } while (++j < num_audiodevs); + } + + if (sound_nblocks >= MAX_MEM_BLOCKS - 1) + printk(KERN_ERR "Sound warning: Deallocation table was too small.\n"); + + return 0; +} + +static void __exit oss_cleanup(void) +{ + int i, j; + + for (i = 0; i < ARRAY_SIZE(dev_list); i++) { + j = 0; + do { + if (dev_list[i].enabled & (1 << j)) + unregister_sound_special(dev_list[i].minor); + } while (++j < num_audiodevs); + } + + unregister_sound_special(1); + unregister_sound_special(8); + + sound_stop_timer(); + + sequencer_unload(); + + for (i = 0; i < MAX_DMA_CHANNELS; i++) + if (dma_alloc_map[i] != DMA_MAP_UNAVAIL) { + printk(KERN_ERR "Sound: Hmm, DMA%d was left allocated - fixed\n", i); + sound_free_dma(i); + } + + for (i = 0; i < sound_nblocks; i++) + vfree(sound_mem_blocks[i]); + +} + +module_init(oss_init); +module_exit(oss_cleanup); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("OSS Sound subsystem"); +MODULE_AUTHOR("Hannu Savolainen, et al."); + + +int sound_alloc_dma(int chn, char *deviceID) +{ + int err; + + if ((err = request_dma(chn, deviceID)) != 0) + return err; + + dma_alloc_map[chn] = DMA_MAP_FREE; + + return 0; +} +EXPORT_SYMBOL(sound_alloc_dma); + +int sound_open_dma(int chn, char *deviceID) +{ + if (!valid_dma(chn)) { + printk(KERN_ERR "sound_open_dma: Invalid DMA channel %d\n", chn); + return 1; + } + + if (dma_alloc_map[chn] != DMA_MAP_FREE) { + printk("sound_open_dma: DMA channel %d busy or not allocated (%d)\n", chn, dma_alloc_map[chn]); + return 1; + } + dma_alloc_map[chn] = DMA_MAP_BUSY; + return 0; +} +EXPORT_SYMBOL(sound_open_dma); + +void sound_free_dma(int chn) +{ + if (dma_alloc_map[chn] == DMA_MAP_UNAVAIL) { + /* printk( "sound_free_dma: Bad access to DMA channel %d\n", chn); */ + return; + } + free_dma(chn); + dma_alloc_map[chn] = DMA_MAP_UNAVAIL; +} +EXPORT_SYMBOL(sound_free_dma); + +void sound_close_dma(int chn) +{ + if (dma_alloc_map[chn] != DMA_MAP_BUSY) { + printk(KERN_ERR "sound_close_dma: Bad access to DMA channel %d\n", chn); + return; + } + dma_alloc_map[chn] = DMA_MAP_FREE; +} +EXPORT_SYMBOL(sound_close_dma); + +static void do_sequencer_timer(unsigned long dummy) +{ + sequencer_timer(0); +} + + +static DEFINE_TIMER(seq_timer, do_sequencer_timer, 0, 0); + +void request_sound_timer(int count) +{ + extern unsigned long seq_time; + + if (count < 0) { + seq_timer.expires = (-count) + jiffies; + add_timer(&seq_timer); + return; + } + count += seq_time; + + count -= jiffies; + + if (count < 1) + count = 1; + + seq_timer.expires = (count) + jiffies; + add_timer(&seq_timer); +} + +void sound_stop_timer(void) +{ + del_timer(&seq_timer); +} + +void conf_printf(char *name, struct address_info *hw_config) +{ +#ifndef CONFIG_SOUND_TRACEINIT + return; +#else + printk("<%s> at 0x%03x", name, hw_config->io_base); + + if (hw_config->irq) + printk(" irq %d", (hw_config->irq > 0) ? hw_config->irq : -hw_config->irq); + + if (hw_config->dma != -1 || hw_config->dma2 != -1) + { + printk(" dma %d", hw_config->dma); + if (hw_config->dma2 != -1) + printk(",%d", hw_config->dma2); + } + printk("\n"); +#endif +} +EXPORT_SYMBOL(conf_printf); + +void conf_printf2(char *name, int base, int irq, int dma, int dma2) +{ +#ifndef CONFIG_SOUND_TRACEINIT + return; +#else + printk("<%s> at 0x%03x", name, base); + + if (irq) + printk(" irq %d", (irq > 0) ? irq : -irq); + + if (dma != -1 || dma2 != -1) + { + printk(" dma %d", dma); + if (dma2 != -1) + printk(",%d", dma2); + } + printk("\n"); +#endif +} +EXPORT_SYMBOL(conf_printf2); + diff --git a/sound/oss/soundvers.h b/sound/oss/soundvers.h new file mode 100644 index 00000000..e9084d2f --- /dev/null +++ b/sound/oss/soundvers.h @@ -0,0 +1,2 @@ +#define SOUND_VERSION_STRING "3.8s2++-971130" +#define SOUND_INTERNAL_VERSION 0x030804 diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c new file mode 100644 index 00000000..09d46484 --- /dev/null +++ b/sound/oss/swarm_cs4297a.c @@ -0,0 +1,2768 @@ +/******************************************************************************* +* +* "swarm_cs4297a.c" -- Cirrus Logic-Crystal CS4297a linux audio driver. +* +* Copyright (C) 2001 Broadcom Corporation. +* Copyright (C) 2000,2001 Cirrus Logic Corp. +* -- adapted from drivers by Thomas Sailer, +* -- but don't bug him; Problems should go to: +* -- tom woller (twoller@crystal.cirrus.com) or +* (audio@crystal.cirrus.com). +* -- adapted from cs4281 PCI driver for cs4297a on +* BCM1250 Synchronous Serial interface +* (Kip Walker, Broadcom Corp.) +* Copyright (C) 2004 Maciej W. Rozycki +* Copyright (C) 2005 Ralf Baechle (ralf@linux-mips.org) +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of the GNU General Public License as published by +* the Free Software Foundation; either version 2 of the License, or +* (at your option) any later version. +* +* This program is distributed in the hope that it will be useful, +* but WITHOUT ANY WARRANTY; without even the implied warranty of +* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +* GNU General Public License for more details. +* +* You should have received a copy of the GNU General Public License +* along with this program; if not, write to the Free Software +* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. +* +* Module command line parameters: +* none +* +* Supported devices: +* /dev/dsp standard /dev/dsp device, (mostly) OSS compatible +* /dev/mixer standard /dev/mixer device, (mostly) OSS compatible +* /dev/midi simple MIDI UART interface, no ioctl +* +* Modification History +* 08/20/00 trw - silence and no stopping DAC until release +* 08/23/00 trw - added CS_DBG statements, fix interrupt hang issue on DAC stop. +* 09/18/00 trw - added 16bit only record with conversion +* 09/24/00 trw - added Enhanced Full duplex (separate simultaneous +* capture/playback rates) +* 10/03/00 trw - fixed mmap (fixed GRECORD and the XMMS mmap test plugin +* libOSSm.so) +* 10/11/00 trw - modified for 2.4.0-test9 kernel enhancements (NR_MAP removal) +* 11/03/00 trw - fixed interrupt loss/stutter, added debug. +* 11/10/00 bkz - added __devinit to cs4297a_hw_init() +* 11/10/00 trw - fixed SMP and capture spinlock hang. +* 12/04/00 trw - cleaned up CSDEBUG flags and added "defaultorder" moduleparm. +* 12/05/00 trw - fixed polling (myth2), and added underrun swptr fix. +* 12/08/00 trw - added PM support. +* 12/14/00 trw - added wrapper code, builds under 2.4.0, 2.2.17-20, 2.2.17-8 +* (RH/Dell base), 2.2.18, 2.2.12. cleaned up code mods by ident. +* 12/19/00 trw - added PM support for 2.2 base (apm_callback). other PM cleanup. +* 12/21/00 trw - added fractional "defaultorder" inputs. if >100 then use +* defaultorder-100 as power of 2 for the buffer size. example: +* 106 = 2^(106-100) = 2^6 = 64 bytes for the buffer size. +* +*******************************************************************************/ + +#include <linux/list.h> +#include <linux/module.h> +#include <linux/string.h> +#include <linux/ioport.h> +#include <linux/sched.h> +#include <linux/delay.h> +#include <linux/sound.h> +#include <linux/slab.h> +#include <linux/soundcard.h> +#include <linux/ac97_codec.h> +#include <linux/pci.h> +#include <linux/bitops.h> +#include <linux/interrupt.h> +#include <linux/init.h> +#include <linux/poll.h> +#include <linux/mutex.h> +#include <linux/kernel.h> + +#include <asm/byteorder.h> +#include <asm/dma.h> +#include <asm/io.h> +#include <asm/uaccess.h> + +#include <asm/sibyte/sb1250_regs.h> +#include <asm/sibyte/sb1250_int.h> +#include <asm/sibyte/sb1250_dma.h> +#include <asm/sibyte/sb1250_scd.h> +#include <asm/sibyte/sb1250_syncser.h> +#include <asm/sibyte/sb1250_mac.h> +#include <asm/sibyte/sb1250.h> + +struct cs4297a_state; + +static DEFINE_MUTEX(swarm_cs4297a_mutex); +static void stop_dac(struct cs4297a_state *s); +static void stop_adc(struct cs4297a_state *s); +static void start_dac(struct cs4297a_state *s); +static void start_adc(struct cs4297a_state *s); +#undef OSS_DOCUMENTED_MIXER_SEMANTICS + +// --------------------------------------------------------------------- + +#define CS4297a_MAGIC 0xf00beef1 + +// buffer order determines the size of the dma buffer for the driver. +// under Linux, a smaller buffer allows more responsiveness from many of the +// applications (e.g. games). A larger buffer allows some of the apps (esound) +// to not underrun the dma buffer as easily. As default, use 32k (order=3) +// rather than 64k as some of the games work more responsively. +// log base 2( buff sz = 32k). + +// +// Turn on/off debugging compilation by commenting out "#define CSDEBUG" +// +#define CSDEBUG 0 +#if CSDEBUG +#define CSDEBUG_INTERFACE 1 +#else +#undef CSDEBUG_INTERFACE +#endif +// +// cs_debugmask areas +// +#define CS_INIT 0x00000001 // initialization and probe functions +#define CS_ERROR 0x00000002 // tmp debugging bit placeholder +#define CS_INTERRUPT 0x00000004 // interrupt handler (separate from all other) +#define CS_FUNCTION 0x00000008 // enter/leave functions +#define CS_WAVE_WRITE 0x00000010 // write information for wave +#define CS_WAVE_READ 0x00000020 // read information for wave +#define CS_AC97 0x00000040 // AC97 register access +#define CS_DESCR 0x00000080 // descriptor management +#define CS_OPEN 0x00000400 // all open functions in the driver +#define CS_RELEASE 0x00000800 // all release functions in the driver +#define CS_PARMS 0x00001000 // functional and operational parameters +#define CS_IOCTL 0x00002000 // ioctl (non-mixer) +#define CS_TMP 0x10000000 // tmp debug mask bit + +// +// CSDEBUG is usual mode is set to 1, then use the +// cs_debuglevel and cs_debugmask to turn on or off debugging. +// Debug level of 1 has been defined to be kernel errors and info +// that should be printed on any released driver. +// +#if CSDEBUG +#define CS_DBGOUT(mask,level,x) if((cs_debuglevel >= (level)) && ((mask) & cs_debugmask) ) {x;} +#else +#define CS_DBGOUT(mask,level,x) +#endif + +#if CSDEBUG +static unsigned long cs_debuglevel = 4; // levels range from 1-9 +static unsigned long cs_debugmask = CS_INIT /*| CS_IOCTL*/; +module_param(cs_debuglevel, int, 0); +module_param(cs_debugmask, int, 0); +#endif +#define CS_TRUE 1 +#define CS_FALSE 0 + +#define CS_TYPE_ADC 0 +#define CS_TYPE_DAC 1 + +#define SER_BASE (A_SER_BASE_1 + KSEG1) +#define SS_CSR(t) (SER_BASE+t) +#define SS_TXTBL(t) (SER_BASE+R_SER_TX_TABLE_BASE+(t*8)) +#define SS_RXTBL(t) (SER_BASE+R_SER_RX_TABLE_BASE+(t*8)) + +#define FRAME_BYTES 32 +#define FRAME_SAMPLE_BYTES 4 + +/* Should this be variable? */ +#define SAMPLE_BUF_SIZE (16*1024) +#define SAMPLE_FRAME_COUNT (SAMPLE_BUF_SIZE / FRAME_SAMPLE_BYTES) +/* The driver can explode/shrink the frames to/from a smaller sample + buffer */ +#define DMA_BLOAT_FACTOR 1 +#define DMA_DESCR (SAMPLE_FRAME_COUNT / DMA_BLOAT_FACTOR) +#define DMA_BUF_SIZE (DMA_DESCR * FRAME_BYTES) + +/* Use the maxmium count (255 == 5.1 ms between interrupts) */ +#define DMA_INT_CNT ((1 << S_DMA_INT_PKTCNT) - 1) + +/* Figure this out: how many TX DMAs ahead to schedule a reg access */ +#define REG_LATENCY 150 + +#define FRAME_TX_US 20 + +#define SERDMA_NEXTBUF(d,f) (((d)->f+1) % (d)->ringsz) + +static const char invalid_magic[] = + KERN_CRIT "cs4297a: invalid magic value\n"; + +#define VALIDATE_STATE(s) \ +({ \ + if (!(s) || (s)->magic != CS4297a_MAGIC) { \ + printk(invalid_magic); \ + return -ENXIO; \ + } \ +}) + +struct list_head cs4297a_devs = { &cs4297a_devs, &cs4297a_devs }; + +typedef struct serdma_descr_s { + u64 descr_a; + u64 descr_b; +} serdma_descr_t; + +typedef unsigned long paddr_t; + +typedef struct serdma_s { + unsigned ringsz; + serdma_descr_t *descrtab; + serdma_descr_t *descrtab_end; + paddr_t descrtab_phys; + + serdma_descr_t *descr_add; + serdma_descr_t *descr_rem; + + u64 *dma_buf; // buffer for DMA contents (frames) + paddr_t dma_buf_phys; + u16 *sample_buf; // tmp buffer for sample conversions + u16 *sb_swptr; + u16 *sb_hwptr; + u16 *sb_end; + + dma_addr_t dmaaddr; +// unsigned buforder; // Log base 2 of 'dma_buf' size in bytes.. + unsigned numfrag; // # of 'fragments' in the buffer. + unsigned fragshift; // Log base 2 of fragment size. + unsigned hwptr, swptr; + unsigned total_bytes; // # bytes process since open. + unsigned blocks; // last returned blocks value GETOPTR + unsigned wakeup; // interrupt occurred on block + int count; + unsigned underrun; // underrun flag + unsigned error; // over/underrun + wait_queue_head_t wait; + wait_queue_head_t reg_wait; + // redundant, but makes calculations easier + unsigned fragsize; // 2**fragshift.. + unsigned sbufsz; // 2**buforder. + unsigned fragsamples; + // OSS stuff + unsigned mapped:1; // Buffer mapped in cs4297a_mmap()? + unsigned ready:1; // prog_dmabuf_dac()/adc() successful? + unsigned endcleared:1; + unsigned type:1; // adc or dac buffer (CS_TYPE_XXX) + unsigned ossfragshift; + int ossmaxfrags; + unsigned subdivision; +} serdma_t; + +struct cs4297a_state { + // magic + unsigned int magic; + + struct list_head list; + + // soundcore stuff + int dev_audio; + int dev_mixer; + + // hardware resources + unsigned int irq; + + struct { + unsigned int rx_ovrrn; /* FIFO */ + unsigned int rx_overflow; /* staging buffer */ + unsigned int tx_underrun; + unsigned int rx_bad; + unsigned int rx_good; + } stats; + + // mixer registers + struct { + unsigned short vol[10]; + unsigned int recsrc; + unsigned int modcnt; + unsigned short micpreamp; + } mix; + + // wave stuff + struct properties { + unsigned fmt; + unsigned fmt_original; // original requested format + unsigned channels; + unsigned rate; + } prop_dac, prop_adc; + unsigned conversion:1; // conversion from 16 to 8 bit in progress + unsigned ena; + spinlock_t lock; + struct mutex open_mutex; + struct mutex open_sem_adc; + struct mutex open_sem_dac; + fmode_t open_mode; + wait_queue_head_t open_wait; + wait_queue_head_t open_wait_adc; + wait_queue_head_t open_wait_dac; + + dma_addr_t dmaaddr_sample_buf; + unsigned buforder_sample_buf; // Log base 2 of 'dma_buf' size in bytes.. + + serdma_t dma_dac, dma_adc; + + volatile u16 read_value; + volatile u16 read_reg; + volatile u64 reg_request; +}; + +#if 1 +#define prog_codec(a,b) +#define dealloc_dmabuf(a,b); +#endif + +static int prog_dmabuf_adc(struct cs4297a_state *s) +{ + s->dma_adc.ready = 1; + return 0; +} + + +static int prog_dmabuf_dac(struct cs4297a_state *s) +{ + s->dma_dac.ready = 1; + return 0; +} + +static void clear_advance(void *buf, unsigned bsize, unsigned bptr, + unsigned len, unsigned char c) +{ + if (bptr + len > bsize) { + unsigned x = bsize - bptr; + memset(((char *) buf) + bptr, c, x); + bptr = 0; + len -= x; + } + CS_DBGOUT(CS_WAVE_WRITE, 4, printk(KERN_INFO + "cs4297a: clear_advance(): memset %d at 0x%.8x for %d size \n", + (unsigned)c, (unsigned)((char *) buf) + bptr, len)); + memset(((char *) buf) + bptr, c, len); +} + +#if CSDEBUG + +// DEBUG ROUTINES + +#define SOUND_MIXER_CS_GETDBGLEVEL _SIOWR('M',120, int) +#define SOUND_MIXER_CS_SETDBGLEVEL _SIOWR('M',121, int) +#define SOUND_MIXER_CS_GETDBGMASK _SIOWR('M',122, int) +#define SOUND_MIXER_CS_SETDBGMASK _SIOWR('M',123, int) + +static void cs_printioctl(unsigned int x) +{ + unsigned int i; + unsigned char vidx; + // Index of mixtable1[] member is Device ID + // and must be <= SOUND_MIXER_NRDEVICES. + // Value of array member is index into s->mix.vol[] + static const unsigned char mixtable1[SOUND_MIXER_NRDEVICES] = { + [SOUND_MIXER_PCM] = 1, // voice + [SOUND_MIXER_LINE1] = 2, // AUX + [SOUND_MIXER_CD] = 3, // CD + [SOUND_MIXER_LINE] = 4, // Line + [SOUND_MIXER_SYNTH] = 5, // FM + [SOUND_MIXER_MIC] = 6, // Mic + [SOUND_MIXER_SPEAKER] = 7, // Speaker + [SOUND_MIXER_RECLEV] = 8, // Recording level + [SOUND_MIXER_VOLUME] = 9 // Master Volume + }; + + switch (x) { + case SOUND_MIXER_CS_GETDBGMASK: + CS_DBGOUT(CS_IOCTL, 4, + printk("SOUND_MIXER_CS_GETDBGMASK:\n")); + break; + case SOUND_MIXER_CS_GETDBGLEVEL: + CS_DBGOUT(CS_IOCTL, 4, + printk("SOUND_MIXER_CS_GETDBGLEVEL:\n")); + break; + case SOUND_MIXER_CS_SETDBGMASK: + CS_DBGOUT(CS_IOCTL, 4, + printk("SOUND_MIXER_CS_SETDBGMASK:\n")); + break; + case SOUND_MIXER_CS_SETDBGLEVEL: + CS_DBGOUT(CS_IOCTL, 4, + printk("SOUND_MIXER_CS_SETDBGLEVEL:\n")); + break; + case OSS_GETVERSION: + CS_DBGOUT(CS_IOCTL, 4, printk("OSS_GETVERSION:\n")); + break; + case SNDCTL_DSP_SYNC: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SYNC:\n")); + break; + case SNDCTL_DSP_SETDUPLEX: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SETDUPLEX:\n")); + break; + case SNDCTL_DSP_GETCAPS: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETCAPS:\n")); + break; + case SNDCTL_DSP_RESET: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_RESET:\n")); + break; + case SNDCTL_DSP_SPEED: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SPEED:\n")); + break; + case SNDCTL_DSP_STEREO: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_STEREO:\n")); + break; + case SNDCTL_DSP_CHANNELS: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_CHANNELS:\n")); + break; + case SNDCTL_DSP_GETFMTS: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETFMTS:\n")); + break; + case SNDCTL_DSP_SETFMT: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SETFMT:\n")); + break; + case SNDCTL_DSP_POST: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_POST:\n")); + break; + case SNDCTL_DSP_GETTRIGGER: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETTRIGGER:\n")); + break; + case SNDCTL_DSP_SETTRIGGER: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SETTRIGGER:\n")); + break; + case SNDCTL_DSP_GETOSPACE: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETOSPACE:\n")); + break; + case SNDCTL_DSP_GETISPACE: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETISPACE:\n")); + break; + case SNDCTL_DSP_NONBLOCK: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_NONBLOCK:\n")); + break; + case SNDCTL_DSP_GETODELAY: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETODELAY:\n")); + break; + case SNDCTL_DSP_GETIPTR: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETIPTR:\n")); + break; + case SNDCTL_DSP_GETOPTR: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETOPTR:\n")); + break; + case SNDCTL_DSP_GETBLKSIZE: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETBLKSIZE:\n")); + break; + case SNDCTL_DSP_SETFRAGMENT: + CS_DBGOUT(CS_IOCTL, 4, + printk("SNDCTL_DSP_SETFRAGMENT:\n")); + break; + case SNDCTL_DSP_SUBDIVIDE: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SUBDIVIDE:\n")); + break; + case SOUND_PCM_READ_RATE: + CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_PCM_READ_RATE:\n")); + break; + case SOUND_PCM_READ_CHANNELS: + CS_DBGOUT(CS_IOCTL, 4, + printk("SOUND_PCM_READ_CHANNELS:\n")); + break; + case SOUND_PCM_READ_BITS: + CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_PCM_READ_BITS:\n")); + break; + case SOUND_PCM_WRITE_FILTER: + CS_DBGOUT(CS_IOCTL, 4, + printk("SOUND_PCM_WRITE_FILTER:\n")); + break; + case SNDCTL_DSP_SETSYNCRO: + CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SETSYNCRO:\n")); + break; + case SOUND_PCM_READ_FILTER: + CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_PCM_READ_FILTER:\n")); + break; + case SOUND_MIXER_PRIVATE1: + CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE1:\n")); + break; + case SOUND_MIXER_PRIVATE2: + CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE2:\n")); + break; + case SOUND_MIXER_PRIVATE3: + CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE3:\n")); + break; + case SOUND_MIXER_PRIVATE4: + CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE4:\n")); + break; + case SOUND_MIXER_PRIVATE5: + CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE5:\n")); + break; + case SOUND_MIXER_INFO: + CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_INFO:\n")); + break; + case SOUND_OLD_MIXER_INFO: + CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_OLD_MIXER_INFO:\n")); + break; + + default: + switch (_IOC_NR(x)) { + case SOUND_MIXER_VOLUME: + CS_DBGOUT(CS_IOCTL, 4, + printk("SOUND_MIXER_VOLUME:\n")); + break; + case SOUND_MIXER_SPEAKER: + CS_DBGOUT(CS_IOCTL, 4, + printk("SOUND_MIXER_SPEAKER:\n")); + break; + case SOUND_MIXER_RECLEV: + CS_DBGOUT(CS_IOCTL, 4, + printk("SOUND_MIXER_RECLEV:\n")); + break; + case SOUND_MIXER_MIC: + CS_DBGOUT(CS_IOCTL, 4, + printk("SOUND_MIXER_MIC:\n")); + break; + case SOUND_MIXER_SYNTH: + CS_DBGOUT(CS_IOCTL, 4, + printk("SOUND_MIXER_SYNTH:\n")); + break; + case SOUND_MIXER_RECSRC: + CS_DBGOUT(CS_IOCTL, 4, + printk("SOUND_MIXER_RECSRC:\n")); + break; + case SOUND_MIXER_DEVMASK: + CS_DBGOUT(CS_IOCTL, 4, + printk("SOUND_MIXER_DEVMASK:\n")); + break; + case SOUND_MIXER_RECMASK: + CS_DBGOUT(CS_IOCTL, 4, + printk("SOUND_MIXER_RECMASK:\n")); + break; + case SOUND_MIXER_STEREODEVS: + CS_DBGOUT(CS_IOCTL, 4, + printk("SOUND_MIXER_STEREODEVS:\n")); + break; + case SOUND_MIXER_CAPS: + CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_CAPS:\n")); + break; + default: + i = _IOC_NR(x); + if (i >= SOUND_MIXER_NRDEVICES + || !(vidx = mixtable1[i])) { + CS_DBGOUT(CS_IOCTL, 4, printk + ("UNKNOWN IOCTL: 0x%.8x NR=%d\n", + x, i)); + } else { + CS_DBGOUT(CS_IOCTL, 4, printk + ("SOUND_MIXER_IOCTL AC9x: 0x%.8x NR=%d\n", + x, i)); + } + break; + } + } +} +#endif + + +static int ser_init(struct cs4297a_state *s) +{ + int i; + + CS_DBGOUT(CS_INIT, 2, + printk(KERN_INFO "cs4297a: Setting up serial parameters\n")); + + __raw_writeq(M_SYNCSER_CMD_RX_RESET | M_SYNCSER_CMD_TX_RESET, SS_CSR(R_SER_CMD)); + + __raw_writeq(M_SYNCSER_MSB_FIRST, SS_CSR(R_SER_MODE)); + __raw_writeq(32, SS_CSR(R_SER_MINFRM_SZ)); + __raw_writeq(32, SS_CSR(R_SER_MAXFRM_SZ)); + + __raw_writeq(1, SS_CSR(R_SER_TX_RD_THRSH)); + __raw_writeq(4, SS_CSR(R_SER_TX_WR_THRSH)); + __raw_writeq(8, SS_CSR(R_SER_RX_RD_THRSH)); + + /* This looks good from experimentation */ + __raw_writeq((M_SYNCSER_TXSYNC_INT | V_SYNCSER_TXSYNC_DLY(0) | M_SYNCSER_TXCLK_EXT | + M_SYNCSER_RXSYNC_INT | V_SYNCSER_RXSYNC_DLY(1) | M_SYNCSER_RXCLK_EXT | M_SYNCSER_RXSYNC_EDGE), + SS_CSR(R_SER_LINE_MODE)); + + /* This looks good from experimentation */ + __raw_writeq(V_SYNCSER_SEQ_COUNT(14) | M_SYNCSER_SEQ_ENABLE | M_SYNCSER_SEQ_STROBE, + SS_TXTBL(0)); + __raw_writeq(V_SYNCSER_SEQ_COUNT(15) | M_SYNCSER_SEQ_ENABLE | M_SYNCSER_SEQ_BYTE, + SS_TXTBL(1)); + __raw_writeq(V_SYNCSER_SEQ_COUNT(13) | M_SYNCSER_SEQ_ENABLE | M_SYNCSER_SEQ_BYTE, + SS_TXTBL(2)); + __raw_writeq(V_SYNCSER_SEQ_COUNT( 0) | M_SYNCSER_SEQ_ENABLE | + M_SYNCSER_SEQ_STROBE | M_SYNCSER_SEQ_LAST, SS_TXTBL(3)); + + __raw_writeq(V_SYNCSER_SEQ_COUNT(14) | M_SYNCSER_SEQ_ENABLE | M_SYNCSER_SEQ_STROBE, + SS_RXTBL(0)); + __raw_writeq(V_SYNCSER_SEQ_COUNT(15) | M_SYNCSER_SEQ_ENABLE | M_SYNCSER_SEQ_BYTE, + SS_RXTBL(1)); + __raw_writeq(V_SYNCSER_SEQ_COUNT(13) | M_SYNCSER_SEQ_ENABLE | M_SYNCSER_SEQ_BYTE, + SS_RXTBL(2)); + __raw_writeq(V_SYNCSER_SEQ_COUNT( 0) | M_SYNCSER_SEQ_ENABLE | M_SYNCSER_SEQ_STROBE | + M_SYNCSER_SEQ_LAST, SS_RXTBL(3)); + + for (i=4; i<16; i++) { + /* Just in case... */ + __raw_writeq(M_SYNCSER_SEQ_LAST, SS_TXTBL(i)); + __raw_writeq(M_SYNCSER_SEQ_LAST, SS_RXTBL(i)); + } + + return 0; +} + +static int init_serdma(serdma_t *dma) +{ + CS_DBGOUT(CS_INIT, 2, + printk(KERN_ERR "cs4297a: desc - %d sbufsize - %d dbufsize - %d\n", + DMA_DESCR, SAMPLE_BUF_SIZE, DMA_BUF_SIZE)); + + /* Descriptors */ + dma->ringsz = DMA_DESCR; + dma->descrtab = kzalloc(dma->ringsz * sizeof(serdma_descr_t), GFP_KERNEL); + if (!dma->descrtab) { + printk(KERN_ERR "cs4297a: kzalloc descrtab failed\n"); + return -1; + } + dma->descrtab_end = dma->descrtab + dma->ringsz; + /* XXX bloddy mess, use proper DMA API here ... */ + dma->descrtab_phys = CPHYSADDR((long)dma->descrtab); + dma->descr_add = dma->descr_rem = dma->descrtab; + + /* Frame buffer area */ + dma->dma_buf = kzalloc(DMA_BUF_SIZE, GFP_KERNEL); + if (!dma->dma_buf) { + printk(KERN_ERR "cs4297a: kzalloc dma_buf failed\n"); + kfree(dma->descrtab); + return -1; + } + dma->dma_buf_phys = CPHYSADDR((long)dma->dma_buf); + + /* Samples buffer area */ + dma->sbufsz = SAMPLE_BUF_SIZE; + dma->sample_buf = kmalloc(dma->sbufsz, GFP_KERNEL); + if (!dma->sample_buf) { + printk(KERN_ERR "cs4297a: kmalloc sample_buf failed\n"); + kfree(dma->descrtab); + kfree(dma->dma_buf); + return -1; + } + dma->sb_swptr = dma->sb_hwptr = dma->sample_buf; + dma->sb_end = (u16 *)((void *)dma->sample_buf + dma->sbufsz); + dma->fragsize = dma->sbufsz >> 1; + + CS_DBGOUT(CS_INIT, 4, + printk(KERN_ERR "cs4297a: descrtab - %08x dma_buf - %x sample_buf - %x\n", + (int)dma->descrtab, (int)dma->dma_buf, + (int)dma->sample_buf)); + + return 0; +} + +static int dma_init(struct cs4297a_state *s) +{ + int i; + + CS_DBGOUT(CS_INIT, 2, + printk(KERN_INFO "cs4297a: Setting up DMA\n")); + + if (init_serdma(&s->dma_adc) || + init_serdma(&s->dma_dac)) + return -1; + + if (__raw_readq(SS_CSR(R_SER_DMA_DSCR_COUNT_RX))|| + __raw_readq(SS_CSR(R_SER_DMA_DSCR_COUNT_TX))) { + panic("DMA state corrupted?!"); + } + + /* Initialize now - the descr/buffer pairings will never + change... */ + for (i=0; i<DMA_DESCR; i++) { + s->dma_dac.descrtab[i].descr_a = M_DMA_SERRX_SOP | V_DMA_DSCRA_A_SIZE(1) | + (s->dma_dac.dma_buf_phys + i*FRAME_BYTES); + s->dma_dac.descrtab[i].descr_b = V_DMA_DSCRB_PKT_SIZE(FRAME_BYTES); + s->dma_adc.descrtab[i].descr_a = V_DMA_DSCRA_A_SIZE(1) | + (s->dma_adc.dma_buf_phys + i*FRAME_BYTES); + s->dma_adc.descrtab[i].descr_b = 0; + } + + __raw_writeq((M_DMA_EOP_INT_EN | V_DMA_INT_PKTCNT(DMA_INT_CNT) | + V_DMA_RINGSZ(DMA_DESCR) | M_DMA_TDX_EN), + SS_CSR(R_SER_DMA_CONFIG0_RX)); + __raw_writeq(M_DMA_L2CA, SS_CSR(R_SER_DMA_CONFIG1_RX)); + __raw_writeq(s->dma_adc.descrtab_phys, SS_CSR(R_SER_DMA_DSCR_BASE_RX)); + + __raw_writeq(V_DMA_RINGSZ(DMA_DESCR), SS_CSR(R_SER_DMA_CONFIG0_TX)); + __raw_writeq(M_DMA_L2CA | M_DMA_NO_DSCR_UPDT, SS_CSR(R_SER_DMA_CONFIG1_TX)); + __raw_writeq(s->dma_dac.descrtab_phys, SS_CSR(R_SER_DMA_DSCR_BASE_TX)); + + /* Prep the receive DMA descriptor ring */ + __raw_writeq(DMA_DESCR, SS_CSR(R_SER_DMA_DSCR_COUNT_RX)); + + __raw_writeq(M_SYNCSER_DMA_RX_EN | M_SYNCSER_DMA_TX_EN, SS_CSR(R_SER_DMA_ENABLE)); + + __raw_writeq((M_SYNCSER_RX_SYNC_ERR | M_SYNCSER_RX_OVERRUN | M_SYNCSER_RX_EOP_COUNT), + SS_CSR(R_SER_INT_MASK)); + + /* Enable the rx/tx; let the codec warm up to the sync and + start sending good frames before the receive FIFO is + enabled */ + __raw_writeq(M_SYNCSER_CMD_TX_EN, SS_CSR(R_SER_CMD)); + udelay(1000); + __raw_writeq(M_SYNCSER_CMD_RX_EN | M_SYNCSER_CMD_TX_EN, SS_CSR(R_SER_CMD)); + + /* XXXKW is this magic? (the "1" part) */ + while ((__raw_readq(SS_CSR(R_SER_STATUS)) & 0xf1) != 1) + ; + + CS_DBGOUT(CS_INIT, 4, + printk(KERN_INFO "cs4297a: status: %08x\n", + (unsigned int)(__raw_readq(SS_CSR(R_SER_STATUS)) & 0xffffffff))); + + return 0; +} + +static int serdma_reg_access(struct cs4297a_state *s, u64 data) +{ + serdma_t *d = &s->dma_dac; + u64 *data_p; + unsigned swptr; + unsigned long flags; + serdma_descr_t *descr; + + if (s->reg_request) { + printk(KERN_ERR "cs4297a: attempt to issue multiple reg_access\n"); + return -1; + } + + if (s->ena & FMODE_WRITE) { + /* Since a writer has the DSP open, we have to mux the + request in */ + s->reg_request = data; + interruptible_sleep_on(&s->dma_dac.reg_wait); + /* XXXKW how can I deal with the starvation case where + the opener isn't writing? */ + } else { + /* Be safe when changing ring pointers */ + spin_lock_irqsave(&s->lock, flags); + if (d->hwptr != d->swptr) { + printk(KERN_ERR "cs4297a: reg access found bookkeeping error (hw/sw = %d/%d\n", + d->hwptr, d->swptr); + spin_unlock_irqrestore(&s->lock, flags); + return -1; + } + swptr = d->swptr; + d->hwptr = d->swptr = (d->swptr + 1) % d->ringsz; + spin_unlock_irqrestore(&s->lock, flags); + + descr = &d->descrtab[swptr]; + data_p = &d->dma_buf[swptr * 4]; + *data_p = cpu_to_be64(data); + __raw_writeq(1, SS_CSR(R_SER_DMA_DSCR_COUNT_TX)); + CS_DBGOUT(CS_DESCR, 4, + printk(KERN_INFO "cs4297a: add_tx %p (%x -> %x)\n", + data_p, swptr, d->hwptr)); + } + + CS_DBGOUT(CS_FUNCTION, 6, + printk(KERN_INFO "cs4297a: serdma_reg_access()-\n")); + + return 0; +} + +//**************************************************************************** +// "cs4297a_read_ac97" -- Reads an AC97 register +//**************************************************************************** +static int cs4297a_read_ac97(struct cs4297a_state *s, u32 offset, + u32 * value) +{ + CS_DBGOUT(CS_AC97, 1, + printk(KERN_INFO "cs4297a: read reg %2x\n", offset)); + if (serdma_reg_access(s, (0xCLL << 60) | (1LL << 47) | ((u64)(offset & 0x7F) << 40))) + return -1; + + interruptible_sleep_on(&s->dma_adc.reg_wait); + *value = s->read_value; + CS_DBGOUT(CS_AC97, 2, + printk(KERN_INFO "cs4297a: rdr reg %x -> %x\n", s->read_reg, s->read_value)); + + return 0; +} + + +//**************************************************************************** +// "cs4297a_write_ac97()"-- writes an AC97 register +//**************************************************************************** +static int cs4297a_write_ac97(struct cs4297a_state *s, u32 offset, + u32 value) +{ + CS_DBGOUT(CS_AC97, 1, + printk(KERN_INFO "cs4297a: write reg %2x -> %04x\n", offset, value)); + return (serdma_reg_access(s, (0xELL << 60) | ((u64)(offset & 0x7F) << 40) | ((value & 0xffff) << 12))); +} + +static void stop_dac(struct cs4297a_state *s) +{ + unsigned long flags; + + CS_DBGOUT(CS_WAVE_WRITE, 3, printk(KERN_INFO "cs4297a: stop_dac():\n")); + spin_lock_irqsave(&s->lock, flags); + s->ena &= ~FMODE_WRITE; +#if 0 + /* XXXKW what do I really want here? My theory for now is + that I just flip the "ena" bit, and the interrupt handler + will stop processing the xmit channel */ + __raw_writeq((s->ena & FMODE_READ) ? M_SYNCSER_DMA_RX_EN : 0, + SS_CSR(R_SER_DMA_ENABLE)); +#endif + + spin_unlock_irqrestore(&s->lock, flags); +} + + +static void start_dac(struct cs4297a_state *s) +{ + unsigned long flags; + + CS_DBGOUT(CS_FUNCTION, 3, printk(KERN_INFO "cs4297a: start_dac()+\n")); + spin_lock_irqsave(&s->lock, flags); + if (!(s->ena & FMODE_WRITE) && (s->dma_dac.mapped || + (s->dma_dac.count > 0 + && s->dma_dac.ready))) { + s->ena |= FMODE_WRITE; + /* XXXKW what do I really want here? My theory for + now is that I just flip the "ena" bit, and the + interrupt handler will start processing the xmit + channel */ + + CS_DBGOUT(CS_WAVE_WRITE | CS_PARMS, 8, printk(KERN_INFO + "cs4297a: start_dac(): start dma\n")); + + } + spin_unlock_irqrestore(&s->lock, flags); + CS_DBGOUT(CS_FUNCTION, 3, + printk(KERN_INFO "cs4297a: start_dac()-\n")); +} + + +static void stop_adc(struct cs4297a_state *s) +{ + unsigned long flags; + + CS_DBGOUT(CS_FUNCTION, 3, + printk(KERN_INFO "cs4297a: stop_adc()+\n")); + + spin_lock_irqsave(&s->lock, flags); + s->ena &= ~FMODE_READ; + + if (s->conversion == 1) { + s->conversion = 0; + s->prop_adc.fmt = s->prop_adc.fmt_original; + } + /* Nothing to do really, I need to keep the DMA going + XXXKW when do I get here, and is there more I should do? */ + spin_unlock_irqrestore(&s->lock, flags); + CS_DBGOUT(CS_FUNCTION, 3, + printk(KERN_INFO "cs4297a: stop_adc()-\n")); +} + + +static void start_adc(struct cs4297a_state *s) +{ + unsigned long flags; + + CS_DBGOUT(CS_FUNCTION, 2, + printk(KERN_INFO "cs4297a: start_adc()+\n")); + + if (!(s->ena & FMODE_READ) && + (s->dma_adc.mapped || s->dma_adc.count <= + (signed) (s->dma_adc.sbufsz - 2 * s->dma_adc.fragsize)) + && s->dma_adc.ready) { + if (s->prop_adc.fmt & AFMT_S8 || s->prop_adc.fmt & AFMT_U8) { + // + // now only use 16 bit capture, due to truncation issue + // in the chip, noticeable distortion occurs. + // allocate buffer and then convert from 16 bit to + // 8 bit for the user buffer. + // + s->prop_adc.fmt_original = s->prop_adc.fmt; + if (s->prop_adc.fmt & AFMT_S8) { + s->prop_adc.fmt &= ~AFMT_S8; + s->prop_adc.fmt |= AFMT_S16_LE; + } + if (s->prop_adc.fmt & AFMT_U8) { + s->prop_adc.fmt &= ~AFMT_U8; + s->prop_adc.fmt |= AFMT_U16_LE; + } + // + // prog_dmabuf_adc performs a stop_adc() but that is + // ok since we really haven't started the DMA yet. + // + prog_codec(s, CS_TYPE_ADC); + + prog_dmabuf_adc(s); + s->conversion = 1; + } + spin_lock_irqsave(&s->lock, flags); + s->ena |= FMODE_READ; + /* Nothing to do really, I am probably already + DMAing... XXXKW when do I get here, and is there + more I should do? */ + spin_unlock_irqrestore(&s->lock, flags); + + CS_DBGOUT(CS_PARMS, 6, printk(KERN_INFO + "cs4297a: start_adc(): start adc\n")); + } + CS_DBGOUT(CS_FUNCTION, 2, + printk(KERN_INFO "cs4297a: start_adc()-\n")); + +} + + +// call with spinlock held! +static void cs4297a_update_ptr(struct cs4297a_state *s, int intflag) +{ + int good_diff, diff, diff2; + u64 *data_p, data; + u32 *s_ptr; + unsigned hwptr; + u32 status; + serdma_t *d; + serdma_descr_t *descr; + + // update ADC pointer + status = intflag ? __raw_readq(SS_CSR(R_SER_STATUS)) : 0; + + if ((s->ena & FMODE_READ) || (status & (M_SYNCSER_RX_EOP_COUNT))) { + d = &s->dma_adc; + hwptr = (unsigned) (((__raw_readq(SS_CSR(R_SER_DMA_CUR_DSCR_ADDR_RX)) & M_DMA_CURDSCR_ADDR) - + d->descrtab_phys) / sizeof(serdma_descr_t)); + + if (s->ena & FMODE_READ) { + CS_DBGOUT(CS_FUNCTION, 2, + printk(KERN_INFO "cs4297a: upd_rcv sw->hw->hw %x/%x/%x (int-%d)n", + d->swptr, d->hwptr, hwptr, intflag)); + /* Number of DMA buffers available for software: */ + diff2 = diff = (d->ringsz + hwptr - d->hwptr) % d->ringsz; + d->hwptr = hwptr; + good_diff = 0; + s_ptr = (u32 *)&(d->dma_buf[d->swptr*4]); + descr = &d->descrtab[d->swptr]; + while (diff2--) { + u64 data = be64_to_cpu(*(u64 *)s_ptr); + u64 descr_a; + u16 left, right; + descr_a = descr->descr_a; + descr->descr_a &= ~M_DMA_SERRX_SOP; + if ((descr_a & M_DMA_DSCRA_A_ADDR) != CPHYSADDR((long)s_ptr)) { + printk(KERN_ERR "cs4297a: RX Bad address (read)\n"); + } + if (((data & 0x9800000000000000) != 0x9800000000000000) || + (!(descr_a & M_DMA_SERRX_SOP)) || + (G_DMA_DSCRB_PKT_SIZE(descr->descr_b) != FRAME_BYTES)) { + s->stats.rx_bad++; + printk(KERN_DEBUG "cs4297a: RX Bad attributes (read)\n"); + continue; + } + s->stats.rx_good++; + if ((data >> 61) == 7) { + s->read_value = (data >> 12) & 0xffff; + s->read_reg = (data >> 40) & 0x7f; + wake_up(&d->reg_wait); + } + if (d->count && (d->sb_hwptr == d->sb_swptr)) { + s->stats.rx_overflow++; + printk(KERN_DEBUG "cs4297a: RX overflow\n"); + continue; + } + good_diff++; + left = ((be32_to_cpu(s_ptr[1]) & 0xff) << 8) | + ((be32_to_cpu(s_ptr[2]) >> 24) & 0xff); + right = (be32_to_cpu(s_ptr[2]) >> 4) & 0xffff; + *d->sb_hwptr++ = cpu_to_be16(left); + *d->sb_hwptr++ = cpu_to_be16(right); + if (d->sb_hwptr == d->sb_end) + d->sb_hwptr = d->sample_buf; + descr++; + if (descr == d->descrtab_end) { + descr = d->descrtab; + s_ptr = (u32 *)s->dma_adc.dma_buf; + } else { + s_ptr += 8; + } + } + d->total_bytes += good_diff * FRAME_SAMPLE_BYTES; + d->count += good_diff * FRAME_SAMPLE_BYTES; + if (d->count > d->sbufsz) { + printk(KERN_ERR "cs4297a: bogus receive overflow!!\n"); + } + d->swptr = (d->swptr + diff) % d->ringsz; + __raw_writeq(diff, SS_CSR(R_SER_DMA_DSCR_COUNT_RX)); + if (d->mapped) { + if (d->count >= (signed) d->fragsize) + wake_up(&d->wait); + } else { + if (d->count > 0) { + CS_DBGOUT(CS_WAVE_READ, 4, + printk(KERN_INFO + "cs4297a: update count -> %d\n", d->count)); + wake_up(&d->wait); + } + } + } else { + /* Receive is going even if no one is + listening (for register accesses and to + avoid FIFO overrun) */ + diff2 = diff = (hwptr + d->ringsz - d->hwptr) % d->ringsz; + if (!diff) { + printk(KERN_ERR "cs4297a: RX full or empty?\n"); + } + + descr = &d->descrtab[d->swptr]; + data_p = &d->dma_buf[d->swptr*4]; + + /* Force this to happen at least once; I got + here because of an interrupt, so there must + be a buffer to process. */ + do { + data = be64_to_cpu(*data_p); + if ((descr->descr_a & M_DMA_DSCRA_A_ADDR) != CPHYSADDR((long)data_p)) { + printk(KERN_ERR "cs4297a: RX Bad address %d (%llx %lx)\n", d->swptr, + (long long)(descr->descr_a & M_DMA_DSCRA_A_ADDR), + (long)CPHYSADDR((long)data_p)); + } + if (!(data & (1LL << 63)) || + !(descr->descr_a & M_DMA_SERRX_SOP) || + (G_DMA_DSCRB_PKT_SIZE(descr->descr_b) != FRAME_BYTES)) { + s->stats.rx_bad++; + printk(KERN_DEBUG "cs4297a: RX Bad attributes\n"); + } else { + s->stats.rx_good++; + if ((data >> 61) == 7) { + s->read_value = (data >> 12) & 0xffff; + s->read_reg = (data >> 40) & 0x7f; + wake_up(&d->reg_wait); + } + } + descr->descr_a &= ~M_DMA_SERRX_SOP; + descr++; + d->swptr++; + data_p += 4; + if (descr == d->descrtab_end) { + descr = d->descrtab; + d->swptr = 0; + data_p = d->dma_buf; + } + __raw_writeq(1, SS_CSR(R_SER_DMA_DSCR_COUNT_RX)); + } while (--diff); + d->hwptr = hwptr; + + CS_DBGOUT(CS_DESCR, 6, + printk(KERN_INFO "cs4297a: hw/sw %x/%x\n", d->hwptr, d->swptr)); + } + + CS_DBGOUT(CS_PARMS, 8, printk(KERN_INFO + "cs4297a: cs4297a_update_ptr(): s=0x%.8x hwptr=%d total_bytes=%d count=%d \n", + (unsigned)s, d->hwptr, + d->total_bytes, d->count)); + } + + /* XXXKW worry about s->reg_request -- there is a starvation + case if s->ena has FMODE_WRITE on, but the client isn't + doing writes */ + + // update DAC pointer + // + // check for end of buffer, means that we are going to wait for another interrupt + // to allow silence to fill the fifos on the part, to keep pops down to a minimum. + // + if (s->ena & FMODE_WRITE) { + serdma_t *d = &s->dma_dac; + hwptr = (unsigned) (((__raw_readq(SS_CSR(R_SER_DMA_CUR_DSCR_ADDR_TX)) & M_DMA_CURDSCR_ADDR) - + d->descrtab_phys) / sizeof(serdma_descr_t)); + diff = (d->ringsz + hwptr - d->hwptr) % d->ringsz; + CS_DBGOUT(CS_WAVE_WRITE, 4, printk(KERN_INFO + "cs4297a: cs4297a_update_ptr(): hw/hw/sw %x/%x/%x diff %d count %d\n", + d->hwptr, hwptr, d->swptr, diff, d->count)); + d->hwptr = hwptr; + /* XXXKW stereo? conversion? Just assume 2 16-bit samples for now */ + d->total_bytes += diff * FRAME_SAMPLE_BYTES; + if (d->mapped) { + d->count += diff * FRAME_SAMPLE_BYTES; + if (d->count >= d->fragsize) { + d->wakeup = 1; + wake_up(&d->wait); + if (d->count > d->sbufsz) + d->count &= d->sbufsz - 1; + } + } else { + d->count -= diff * FRAME_SAMPLE_BYTES; + if (d->count <= 0) { + // + // fill with silence, and do not shut down the DAC. + // Continue to play silence until the _release. + // + CS_DBGOUT(CS_WAVE_WRITE, 6, printk(KERN_INFO + "cs4297a: cs4297a_update_ptr(): memset %d at 0x%.8x for %d size \n", + (unsigned)(s->prop_dac.fmt & + (AFMT_U8 | AFMT_U16_LE)) ? 0x80 : 0, + (unsigned)d->dma_buf, + d->ringsz)); + memset(d->dma_buf, 0, d->ringsz * FRAME_BYTES); + if (d->count < 0) { + d->underrun = 1; + s->stats.tx_underrun++; + d->count = 0; + CS_DBGOUT(CS_ERROR, 9, printk(KERN_INFO + "cs4297a: cs4297a_update_ptr(): underrun\n")); + } + } else if (d->count <= + (signed) d->fragsize + && !d->endcleared) { + /* XXXKW what is this for? */ + clear_advance(d->dma_buf, + d->sbufsz, + d->swptr, + d->fragsize, + 0); + d->endcleared = 1; + } + if ( (d->count <= (signed) d->sbufsz/2) || intflag) + { + CS_DBGOUT(CS_WAVE_WRITE, 4, + printk(KERN_INFO + "cs4297a: update count -> %d\n", d->count)); + wake_up(&d->wait); + } + } + CS_DBGOUT(CS_PARMS, 8, printk(KERN_INFO + "cs4297a: cs4297a_update_ptr(): s=0x%.8x hwptr=%d total_bytes=%d count=%d \n", + (unsigned) s, d->hwptr, + d->total_bytes, d->count)); + } +} + +static int mixer_ioctl(struct cs4297a_state *s, unsigned int cmd, + unsigned long arg) +{ + // Index to mixer_src[] is value of AC97 Input Mux Select Reg. + // Value of array member is recording source Device ID Mask. + static const unsigned int mixer_src[8] = { + SOUND_MASK_MIC, SOUND_MASK_CD, 0, SOUND_MASK_LINE1, + SOUND_MASK_LINE, SOUND_MASK_VOLUME, 0, 0 + }; + + // Index of mixtable1[] member is Device ID + // and must be <= SOUND_MIXER_NRDEVICES. + // Value of array member is index into s->mix.vol[] + static const unsigned char mixtable1[SOUND_MIXER_NRDEVICES] = { + [SOUND_MIXER_PCM] = 1, // voice + [SOUND_MIXER_LINE1] = 2, // AUX + [SOUND_MIXER_CD] = 3, // CD + [SOUND_MIXER_LINE] = 4, // Line + [SOUND_MIXER_SYNTH] = 5, // FM + [SOUND_MIXER_MIC] = 6, // Mic + [SOUND_MIXER_SPEAKER] = 7, // Speaker + [SOUND_MIXER_RECLEV] = 8, // Recording level + [SOUND_MIXER_VOLUME] = 9 // Master Volume + }; + + static const unsigned mixreg[] = { + AC97_PCMOUT_VOL, + AC97_AUX_VOL, + AC97_CD_VOL, + AC97_LINEIN_VOL + }; + unsigned char l, r, rl, rr, vidx; + unsigned char attentbl[11] = + { 63, 42, 26, 17, 14, 11, 8, 6, 4, 2, 0 }; + unsigned temp1; + int i, val; + + VALIDATE_STATE(s); + CS_DBGOUT(CS_FUNCTION, 4, printk(KERN_INFO + "cs4297a: mixer_ioctl(): s=0x%.8x cmd=0x%.8x\n", + (unsigned) s, cmd)); +#if CSDEBUG + cs_printioctl(cmd); +#endif +#if CSDEBUG_INTERFACE + + if ((cmd == SOUND_MIXER_CS_GETDBGMASK) || + (cmd == SOUND_MIXER_CS_SETDBGMASK) || + (cmd == SOUND_MIXER_CS_GETDBGLEVEL) || + (cmd == SOUND_MIXER_CS_SETDBGLEVEL)) + { + switch (cmd) { + + case SOUND_MIXER_CS_GETDBGMASK: + return put_user(cs_debugmask, + (unsigned long *) arg); + + case SOUND_MIXER_CS_GETDBGLEVEL: + return put_user(cs_debuglevel, + (unsigned long *) arg); + + case SOUND_MIXER_CS_SETDBGMASK: + if (get_user(val, (unsigned long *) arg)) + return -EFAULT; + cs_debugmask = val; + return 0; + + case SOUND_MIXER_CS_SETDBGLEVEL: + if (get_user(val, (unsigned long *) arg)) + return -EFAULT; + cs_debuglevel = val; + return 0; + default: + CS_DBGOUT(CS_ERROR, 1, printk(KERN_INFO + "cs4297a: mixer_ioctl(): ERROR unknown debug cmd\n")); + return 0; + } + } +#endif + + if (cmd == SOUND_MIXER_PRIVATE1) { + return -EINVAL; + } + if (cmd == SOUND_MIXER_PRIVATE2) { + // enable/disable/query spatializer + if (get_user(val, (int *) arg)) + return -EFAULT; + if (val != -1) { + temp1 = (val & 0x3f) >> 2; + cs4297a_write_ac97(s, AC97_3D_CONTROL, temp1); + cs4297a_read_ac97(s, AC97_GENERAL_PURPOSE, + &temp1); + cs4297a_write_ac97(s, AC97_GENERAL_PURPOSE, + temp1 | 0x2000); + } + cs4297a_read_ac97(s, AC97_3D_CONTROL, &temp1); + return put_user((temp1 << 2) | 3, (int *) arg); + } + if (cmd == SOUND_MIXER_INFO) { + mixer_info info; + memset(&info, 0, sizeof(info)); + strlcpy(info.id, "CS4297a", sizeof(info.id)); + strlcpy(info.name, "Crystal CS4297a", sizeof(info.name)); + info.modify_counter = s->mix.modcnt; + if (copy_to_user((void *) arg, &info, sizeof(info))) + return -EFAULT; + return 0; + } + if (cmd == SOUND_OLD_MIXER_INFO) { + _old_mixer_info info; + memset(&info, 0, sizeof(info)); + strlcpy(info.id, "CS4297a", sizeof(info.id)); + strlcpy(info.name, "Crystal CS4297a", sizeof(info.name)); + if (copy_to_user((void *) arg, &info, sizeof(info))) + return -EFAULT; + return 0; + } + if (cmd == OSS_GETVERSION) + return put_user(SOUND_VERSION, (int *) arg); + + if (_IOC_TYPE(cmd) != 'M' || _SIOC_SIZE(cmd) != sizeof(int)) + return -EINVAL; + + // If ioctl has only the SIOC_READ bit(bit 31) + // on, process the only-read commands. + if (_SIOC_DIR(cmd) == _SIOC_READ) { + switch (_IOC_NR(cmd)) { + case SOUND_MIXER_RECSRC: // Arg contains a bit for each recording source + cs4297a_read_ac97(s, AC97_RECORD_SELECT, + &temp1); + return put_user(mixer_src[temp1 & 7], (int *) arg); + + case SOUND_MIXER_DEVMASK: // Arg contains a bit for each supported device + return put_user(SOUND_MASK_PCM | SOUND_MASK_LINE | + SOUND_MASK_VOLUME | SOUND_MASK_RECLEV, + (int *) arg); + + case SOUND_MIXER_RECMASK: // Arg contains a bit for each supported recording source + return put_user(SOUND_MASK_LINE | SOUND_MASK_VOLUME, + (int *) arg); + + case SOUND_MIXER_STEREODEVS: // Mixer channels supporting stereo + return put_user(SOUND_MASK_PCM | SOUND_MASK_LINE | + SOUND_MASK_VOLUME | SOUND_MASK_RECLEV, + (int *) arg); + + case SOUND_MIXER_CAPS: + return put_user(SOUND_CAP_EXCL_INPUT, (int *) arg); + + default: + i = _IOC_NR(cmd); + if (i >= SOUND_MIXER_NRDEVICES + || !(vidx = mixtable1[i])) + return -EINVAL; + return put_user(s->mix.vol[vidx - 1], (int *) arg); + } + } + // If ioctl doesn't have both the SIOC_READ and + // the SIOC_WRITE bit set, return invalid. + if (_SIOC_DIR(cmd) != (_SIOC_READ | _SIOC_WRITE)) + return -EINVAL; + + // Increment the count of volume writes. + s->mix.modcnt++; + + // Isolate the command; it must be a write. + switch (_IOC_NR(cmd)) { + + case SOUND_MIXER_RECSRC: // Arg contains a bit for each recording source + if (get_user(val, (int *) arg)) + return -EFAULT; + i = hweight32(val); // i = # bits on in val. + if (i != 1) // One & only 1 bit must be on. + return 0; + for (i = 0; i < sizeof(mixer_src) / sizeof(int); i++) { + if (val == mixer_src[i]) { + temp1 = (i << 8) | i; + cs4297a_write_ac97(s, + AC97_RECORD_SELECT, + temp1); + return 0; + } + } + return 0; + + case SOUND_MIXER_VOLUME: + if (get_user(val, (int *) arg)) + return -EFAULT; + l = val & 0xff; + if (l > 100) + l = 100; // Max soundcard.h vol is 100. + if (l < 6) { + rl = 63; + l = 0; + } else + rl = attentbl[(10 * l) / 100]; // Convert 0-100 vol to 63-0 atten. + + r = (val >> 8) & 0xff; + if (r > 100) + r = 100; // Max right volume is 100, too + if (r < 6) { + rr = 63; + r = 0; + } else + rr = attentbl[(10 * r) / 100]; // Convert volume to attenuation. + + if ((rl > 60) && (rr > 60)) // If both l & r are 'low', + temp1 = 0x8000; // turn on the mute bit. + else + temp1 = 0; + + temp1 |= (rl << 8) | rr; + + cs4297a_write_ac97(s, AC97_MASTER_VOL_STEREO, temp1); + cs4297a_write_ac97(s, AC97_PHONE_VOL, temp1); + +#ifdef OSS_DOCUMENTED_MIXER_SEMANTICS + s->mix.vol[8] = ((unsigned int) r << 8) | l; +#else + s->mix.vol[8] = val; +#endif + return put_user(s->mix.vol[8], (int *) arg); + + case SOUND_MIXER_SPEAKER: + if (get_user(val, (int *) arg)) + return -EFAULT; + l = val & 0xff; + if (l > 100) + l = 100; + if (l < 3) { + rl = 0; + l = 0; + } else { + rl = (l * 2 - 5) / 13; // Convert 0-100 range to 0-15. + l = (rl * 13 + 5) / 2; + } + + if (rl < 3) { + temp1 = 0x8000; + rl = 0; + } else + temp1 = 0; + rl = 15 - rl; // Convert volume to attenuation. + temp1 |= rl << 1; + cs4297a_write_ac97(s, AC97_PCBEEP_VOL, temp1); + +#ifdef OSS_DOCUMENTED_MIXER_SEMANTICS + s->mix.vol[6] = l << 8; +#else + s->mix.vol[6] = val; +#endif + return put_user(s->mix.vol[6], (int *) arg); + + case SOUND_MIXER_RECLEV: + if (get_user(val, (int *) arg)) + return -EFAULT; + l = val & 0xff; + if (l > 100) + l = 100; + r = (val >> 8) & 0xff; + if (r > 100) + r = 100; + rl = (l * 2 - 5) / 13; // Convert 0-100 scale to 0-15. + rr = (r * 2 - 5) / 13; + if (rl < 3 && rr < 3) + temp1 = 0x8000; + else + temp1 = 0; + + temp1 = temp1 | (rl << 8) | rr; + cs4297a_write_ac97(s, AC97_RECORD_GAIN, temp1); + +#ifdef OSS_DOCUMENTED_MIXER_SEMANTICS + s->mix.vol[7] = ((unsigned int) r << 8) | l; +#else + s->mix.vol[7] = val; +#endif + return put_user(s->mix.vol[7], (int *) arg); + + case SOUND_MIXER_MIC: + if (get_user(val, (int *) arg)) + return -EFAULT; + l = val & 0xff; + if (l > 100) + l = 100; + if (l < 1) { + l = 0; + rl = 0; + } else { + rl = ((unsigned) l * 5 - 4) / 16; // Convert 0-100 range to 0-31. + l = (rl * 16 + 4) / 5; + } + cs4297a_read_ac97(s, AC97_MIC_VOL, &temp1); + temp1 &= 0x40; // Isolate 20db gain bit. + if (rl < 3) { + temp1 |= 0x8000; + rl = 0; + } + rl = 31 - rl; // Convert volume to attenuation. + temp1 |= rl; + cs4297a_write_ac97(s, AC97_MIC_VOL, temp1); + +#ifdef OSS_DOCUMENTED_MIXER_SEMANTICS + s->mix.vol[5] = val << 8; +#else + s->mix.vol[5] = val; +#endif + return put_user(s->mix.vol[5], (int *) arg); + + + case SOUND_MIXER_SYNTH: + if (get_user(val, (int *) arg)) + return -EFAULT; + l = val & 0xff; + if (l > 100) + l = 100; + if (get_user(val, (int *) arg)) + return -EFAULT; + r = (val >> 8) & 0xff; + if (r > 100) + r = 100; + rl = (l * 2 - 11) / 3; // Convert 0-100 range to 0-63. + rr = (r * 2 - 11) / 3; + if (rl < 3) // If l is low, turn on + temp1 = 0x0080; // the mute bit. + else + temp1 = 0; + + rl = 63 - rl; // Convert vol to attenuation. +// writel(temp1 | rl, s->pBA0 + FMLVC); + if (rr < 3) // If rr is low, turn on + temp1 = 0x0080; // the mute bit. + else + temp1 = 0; + rr = 63 - rr; // Convert vol to attenuation. +// writel(temp1 | rr, s->pBA0 + FMRVC); + +#ifdef OSS_DOCUMENTED_MIXER_SEMANTICS + s->mix.vol[4] = (r << 8) | l; +#else + s->mix.vol[4] = val; +#endif + return put_user(s->mix.vol[4], (int *) arg); + + + default: + CS_DBGOUT(CS_IOCTL, 4, printk(KERN_INFO + "cs4297a: mixer_ioctl(): default\n")); + + i = _IOC_NR(cmd); + if (i >= SOUND_MIXER_NRDEVICES || !(vidx = mixtable1[i])) + return -EINVAL; + if (get_user(val, (int *) arg)) + return -EFAULT; + l = val & 0xff; + if (l > 100) + l = 100; + if (l < 1) { + l = 0; + rl = 31; + } else + rl = (attentbl[(l * 10) / 100]) >> 1; + + r = (val >> 8) & 0xff; + if (r > 100) + r = 100; + if (r < 1) { + r = 0; + rr = 31; + } else + rr = (attentbl[(r * 10) / 100]) >> 1; + if ((rl > 30) && (rr > 30)) + temp1 = 0x8000; + else + temp1 = 0; + temp1 = temp1 | (rl << 8) | rr; + cs4297a_write_ac97(s, mixreg[vidx - 1], temp1); + +#ifdef OSS_DOCUMENTED_MIXER_SEMANTICS + s->mix.vol[vidx - 1] = ((unsigned int) r << 8) | l; +#else + s->mix.vol[vidx - 1] = val; +#endif + return put_user(s->mix.vol[vidx - 1], (int *) arg); + } +} + + +// --------------------------------------------------------------------- + +static int cs4297a_open_mixdev(struct inode *inode, struct file *file) +{ + int minor = iminor(inode); + struct cs4297a_state *s=NULL; + struct list_head *entry; + + CS_DBGOUT(CS_FUNCTION | CS_OPEN, 4, + printk(KERN_INFO "cs4297a: cs4297a_open_mixdev()+\n")); + + mutex_lock(&swarm_cs4297a_mutex); + list_for_each(entry, &cs4297a_devs) + { + s = list_entry(entry, struct cs4297a_state, list); + if(s->dev_mixer == minor) + break; + } + if (!s) + { + CS_DBGOUT(CS_FUNCTION | CS_OPEN | CS_ERROR, 2, + printk(KERN_INFO "cs4297a: cs4297a_open_mixdev()- -ENODEV\n")); + + mutex_unlock(&swarm_cs4297a_mutex); + return -ENODEV; + } + VALIDATE_STATE(s); + file->private_data = s; + + CS_DBGOUT(CS_FUNCTION | CS_OPEN, 4, + printk(KERN_INFO "cs4297a: cs4297a_open_mixdev()- 0\n")); + mutex_unlock(&swarm_cs4297a_mutex); + + return nonseekable_open(inode, file); +} + + +static int cs4297a_release_mixdev(struct inode *inode, struct file *file) +{ + struct cs4297a_state *s = + (struct cs4297a_state *) file->private_data; + + VALIDATE_STATE(s); + return 0; +} + + +static int cs4297a_ioctl_mixdev(struct file *file, + unsigned int cmd, unsigned long arg) +{ + int ret; + mutex_lock(&swarm_cs4297a_mutex); + ret = mixer_ioctl((struct cs4297a_state *) file->private_data, cmd, + arg); + mutex_unlock(&swarm_cs4297a_mutex); + return ret; +} + + +// ****************************************************************************************** +// Mixer file operations struct. +// ****************************************************************************************** +static const struct file_operations cs4297a_mixer_fops = { + .owner = THIS_MODULE, + .llseek = no_llseek, + .unlocked_ioctl = cs4297a_ioctl_mixdev, + .open = cs4297a_open_mixdev, + .release = cs4297a_release_mixdev, +}; + +// --------------------------------------------------------------------- + + +static int drain_adc(struct cs4297a_state *s, int nonblock) +{ + /* This routine serves no purpose currently - any samples + sitting in the receive queue will just be processed by the + background consumer. This would be different if DMA + actually stopped when there were no clients. */ + return 0; +} + +static int drain_dac(struct cs4297a_state *s, int nonblock) +{ + DECLARE_WAITQUEUE(wait, current); + unsigned long flags; + unsigned hwptr; + unsigned tmo; + int count; + + if (s->dma_dac.mapped) + return 0; + if (nonblock) + return -EBUSY; + add_wait_queue(&s->dma_dac.wait, &wait); + while ((count = __raw_readq(SS_CSR(R_SER_DMA_DSCR_COUNT_TX))) || + (s->dma_dac.count > 0)) { + if (!signal_pending(current)) { + set_current_state(TASK_INTERRUPTIBLE); + /* XXXKW is this calculation working? */ + tmo = ((count * FRAME_TX_US) * HZ) / 1000000; + schedule_timeout(tmo + 1); + } else { + /* XXXKW do I care if there is a signal pending? */ + } + } + spin_lock_irqsave(&s->lock, flags); + /* Reset the bookkeeping */ + hwptr = (int)(((__raw_readq(SS_CSR(R_SER_DMA_CUR_DSCR_ADDR_TX)) & M_DMA_CURDSCR_ADDR) - + s->dma_dac.descrtab_phys) / sizeof(serdma_descr_t)); + s->dma_dac.hwptr = s->dma_dac.swptr = hwptr; + spin_unlock_irqrestore(&s->lock, flags); + remove_wait_queue(&s->dma_dac.wait, &wait); + current->state = TASK_RUNNING; + return 0; +} + + +// --------------------------------------------------------------------- + +static ssize_t cs4297a_read(struct file *file, char *buffer, size_t count, + loff_t * ppos) +{ + struct cs4297a_state *s = + (struct cs4297a_state *) file->private_data; + ssize_t ret; + unsigned long flags; + int cnt, count_fr, cnt_by; + unsigned copied = 0; + + CS_DBGOUT(CS_FUNCTION | CS_WAVE_READ, 2, + printk(KERN_INFO "cs4297a: cs4297a_read()+ %d \n", count)); + + VALIDATE_STATE(s); + if (s->dma_adc.mapped) + return -ENXIO; + if (!s->dma_adc.ready && (ret = prog_dmabuf_adc(s))) + return ret; + if (!access_ok(VERIFY_WRITE, buffer, count)) + return -EFAULT; + ret = 0; +// +// "count" is the amount of bytes to read (from app), is decremented each loop +// by the amount of bytes that have been returned to the user buffer. +// "cnt" is the running total of each read from the buffer (changes each loop) +// "buffer" points to the app's buffer +// "ret" keeps a running total of the amount of bytes that have been copied +// to the user buffer. +// "copied" is the total bytes copied into the user buffer for each loop. +// + while (count > 0) { + CS_DBGOUT(CS_WAVE_READ, 8, printk(KERN_INFO + "_read() count>0 count=%d .count=%d .swptr=%d .hwptr=%d \n", + count, s->dma_adc.count, + s->dma_adc.swptr, s->dma_adc.hwptr)); + spin_lock_irqsave(&s->lock, flags); + + /* cnt will be the number of available samples (16-bit + stereo); it starts out as the maxmimum consequetive + samples */ + cnt = (s->dma_adc.sb_end - s->dma_adc.sb_swptr) / 2; + count_fr = s->dma_adc.count / FRAME_SAMPLE_BYTES; + + // dma_adc.count is the current total bytes that have not been read. + // if the amount of unread bytes from the current sw pointer to the + // end of the buffer is greater than the current total bytes that + // have not been read, then set the "cnt" (unread bytes) to the + // amount of unread bytes. + + if (count_fr < cnt) + cnt = count_fr; + cnt_by = cnt * FRAME_SAMPLE_BYTES; + spin_unlock_irqrestore(&s->lock, flags); + // + // if we are converting from 8/16 then we need to copy + // twice the number of 16 bit bytes then 8 bit bytes. + // + if (s->conversion) { + if (cnt_by > (count * 2)) { + cnt = (count * 2) / FRAME_SAMPLE_BYTES; + cnt_by = count * 2; + } + } else { + if (cnt_by > count) { + cnt = count / FRAME_SAMPLE_BYTES; + cnt_by = count; + } + } + // + // "cnt" NOW is the smaller of the amount that will be read, + // and the amount that is requested in this read (or partial). + // if there are no bytes in the buffer to read, then start the + // ADC and wait for the interrupt handler to wake us up. + // + if (cnt <= 0) { + + // start up the dma engine and then continue back to the top of + // the loop when wake up occurs. + start_adc(s); + if (file->f_flags & O_NONBLOCK) + return ret ? ret : -EAGAIN; + interruptible_sleep_on(&s->dma_adc.wait); + if (signal_pending(current)) + return ret ? ret : -ERESTARTSYS; + continue; + } + // there are bytes in the buffer to read. + // copy from the hw buffer over to the user buffer. + // user buffer is designated by "buffer" + // virtual address to copy from is dma_buf+swptr + // the "cnt" is the number of bytes to read. + + CS_DBGOUT(CS_WAVE_READ, 2, printk(KERN_INFO + "_read() copy_to cnt=%d count=%d ", cnt_by, count)); + CS_DBGOUT(CS_WAVE_READ, 8, printk(KERN_INFO + " .sbufsz=%d .count=%d buffer=0x%.8x ret=%d\n", + s->dma_adc.sbufsz, s->dma_adc.count, + (unsigned) buffer, ret)); + + if (copy_to_user (buffer, ((void *)s->dma_adc.sb_swptr), cnt_by)) + return ret ? ret : -EFAULT; + copied = cnt_by; + + /* Return the descriptors */ + spin_lock_irqsave(&s->lock, flags); + CS_DBGOUT(CS_FUNCTION, 2, + printk(KERN_INFO "cs4297a: upd_rcv sw->hw %x/%x\n", s->dma_adc.swptr, s->dma_adc.hwptr)); + s->dma_adc.count -= cnt_by; + s->dma_adc.sb_swptr += cnt * 2; + if (s->dma_adc.sb_swptr == s->dma_adc.sb_end) + s->dma_adc.sb_swptr = s->dma_adc.sample_buf; + spin_unlock_irqrestore(&s->lock, flags); + count -= copied; + buffer += copied; + ret += copied; + start_adc(s); + } + CS_DBGOUT(CS_FUNCTION | CS_WAVE_READ, 2, + printk(KERN_INFO "cs4297a: cs4297a_read()- %d\n", ret)); + return ret; +} + + +static ssize_t cs4297a_write(struct file *file, const char *buffer, + size_t count, loff_t * ppos) +{ + struct cs4297a_state *s = + (struct cs4297a_state *) file->private_data; + ssize_t ret; + unsigned long flags; + unsigned swptr, hwptr; + int cnt; + + CS_DBGOUT(CS_FUNCTION | CS_WAVE_WRITE, 2, + printk(KERN_INFO "cs4297a: cs4297a_write()+ count=%d\n", + count)); + VALIDATE_STATE(s); + + if (s->dma_dac.mapped) + return -ENXIO; + if (!s->dma_dac.ready && (ret = prog_dmabuf_dac(s))) + return ret; + if (!access_ok(VERIFY_READ, buffer, count)) + return -EFAULT; + ret = 0; + while (count > 0) { + serdma_t *d = &s->dma_dac; + int copy_cnt; + u32 *s_tmpl; + u32 *t_tmpl; + u32 left, right; + int swap = (s->prop_dac.fmt == AFMT_S16_LE) || (s->prop_dac.fmt == AFMT_U16_LE); + + /* XXXXXX this is broken for BLOAT_FACTOR */ + spin_lock_irqsave(&s->lock, flags); + if (d->count < 0) { + d->count = 0; + d->swptr = d->hwptr; + } + if (d->underrun) { + d->underrun = 0; + hwptr = (unsigned) (((__raw_readq(SS_CSR(R_SER_DMA_CUR_DSCR_ADDR_TX)) & M_DMA_CURDSCR_ADDR) - + d->descrtab_phys) / sizeof(serdma_descr_t)); + d->swptr = d->hwptr = hwptr; + } + swptr = d->swptr; + cnt = d->sbufsz - (swptr * FRAME_SAMPLE_BYTES); + /* Will this write fill up the buffer? */ + if (d->count + cnt > d->sbufsz) + cnt = d->sbufsz - d->count; + spin_unlock_irqrestore(&s->lock, flags); + if (cnt > count) + cnt = count; + if (cnt <= 0) { + start_dac(s); + if (file->f_flags & O_NONBLOCK) + return ret ? ret : -EAGAIN; + interruptible_sleep_on(&d->wait); + if (signal_pending(current)) + return ret ? ret : -ERESTARTSYS; + continue; + } + if (copy_from_user(d->sample_buf, buffer, cnt)) + return ret ? ret : -EFAULT; + + copy_cnt = cnt; + s_tmpl = (u32 *)d->sample_buf; + t_tmpl = (u32 *)(d->dma_buf + (swptr * 4)); + + /* XXXKW assuming 16-bit stereo! */ + do { + u32 tmp; + + t_tmpl[0] = cpu_to_be32(0x98000000); + + tmp = be32_to_cpu(s_tmpl[0]); + left = tmp & 0xffff; + right = tmp >> 16; + if (swap) { + left = swab16(left); + right = swab16(right); + } + t_tmpl[1] = cpu_to_be32(left >> 8); + t_tmpl[2] = cpu_to_be32(((left & 0xff) << 24) | + (right << 4)); + + s_tmpl++; + t_tmpl += 8; + copy_cnt -= 4; + } while (copy_cnt); + + /* Mux in any pending read/write accesses */ + if (s->reg_request) { + *(u64 *)(d->dma_buf + (swptr * 4)) |= + cpu_to_be64(s->reg_request); + s->reg_request = 0; + wake_up(&s->dma_dac.reg_wait); + } + + CS_DBGOUT(CS_WAVE_WRITE, 4, + printk(KERN_INFO + "cs4297a: copy in %d to swptr %x\n", cnt, swptr)); + + swptr = (swptr + (cnt/FRAME_SAMPLE_BYTES)) % d->ringsz; + __raw_writeq(cnt/FRAME_SAMPLE_BYTES, SS_CSR(R_SER_DMA_DSCR_COUNT_TX)); + spin_lock_irqsave(&s->lock, flags); + d->swptr = swptr; + d->count += cnt; + d->endcleared = 0; + spin_unlock_irqrestore(&s->lock, flags); + count -= cnt; + buffer += cnt; + ret += cnt; + start_dac(s); + } + CS_DBGOUT(CS_FUNCTION | CS_WAVE_WRITE, 2, + printk(KERN_INFO "cs4297a: cs4297a_write()- %d\n", ret)); + return ret; +} + + +static unsigned int cs4297a_poll(struct file *file, + struct poll_table_struct *wait) +{ + struct cs4297a_state *s = + (struct cs4297a_state *) file->private_data; + unsigned long flags; + unsigned int mask = 0; + + CS_DBGOUT(CS_FUNCTION | CS_WAVE_WRITE | CS_WAVE_READ, 4, + printk(KERN_INFO "cs4297a: cs4297a_poll()+\n")); + VALIDATE_STATE(s); + if (file->f_mode & FMODE_WRITE) { + CS_DBGOUT(CS_FUNCTION | CS_WAVE_WRITE | CS_WAVE_READ, 4, + printk(KERN_INFO + "cs4297a: cs4297a_poll() wait on FMODE_WRITE\n")); + if(!s->dma_dac.ready && prog_dmabuf_dac(s)) + return 0; + poll_wait(file, &s->dma_dac.wait, wait); + } + if (file->f_mode & FMODE_READ) { + CS_DBGOUT(CS_FUNCTION | CS_WAVE_WRITE | CS_WAVE_READ, 4, + printk(KERN_INFO + "cs4297a: cs4297a_poll() wait on FMODE_READ\n")); + if(!s->dma_dac.ready && prog_dmabuf_adc(s)) + return 0; + poll_wait(file, &s->dma_adc.wait, wait); + } + spin_lock_irqsave(&s->lock, flags); + cs4297a_update_ptr(s,CS_FALSE); + if (file->f_mode & FMODE_WRITE) { + if (s->dma_dac.mapped) { + if (s->dma_dac.count >= + (signed) s->dma_dac.fragsize) { + if (s->dma_dac.wakeup) + mask |= POLLOUT | POLLWRNORM; + else + mask = 0; + s->dma_dac.wakeup = 0; + } + } else { + if ((signed) (s->dma_dac.sbufsz/2) >= s->dma_dac.count) + mask |= POLLOUT | POLLWRNORM; + } + } else if (file->f_mode & FMODE_READ) { + if (s->dma_adc.mapped) { + if (s->dma_adc.count >= (signed) s->dma_adc.fragsize) + mask |= POLLIN | POLLRDNORM; + } else { + if (s->dma_adc.count > 0) + mask |= POLLIN | POLLRDNORM; + } + } + spin_unlock_irqrestore(&s->lock, flags); + CS_DBGOUT(CS_FUNCTION | CS_WAVE_WRITE | CS_WAVE_READ, 4, + printk(KERN_INFO "cs4297a: cs4297a_poll()- 0x%.8x\n", + mask)); + return mask; +} + + +static int cs4297a_mmap(struct file *file, struct vm_area_struct *vma) +{ + /* XXXKW currently no mmap support */ + return -EINVAL; + return 0; +} + + +static int cs4297a_ioctl(struct file *file, + unsigned int cmd, unsigned long arg) +{ + struct cs4297a_state *s = + (struct cs4297a_state *) file->private_data; + unsigned long flags; + audio_buf_info abinfo; + count_info cinfo; + int val, mapped, ret; + + CS_DBGOUT(CS_FUNCTION|CS_IOCTL, 4, printk(KERN_INFO + "cs4297a: cs4297a_ioctl(): file=0x%.8x cmd=0x%.8x\n", + (unsigned) file, cmd)); +#if CSDEBUG + cs_printioctl(cmd); +#endif + VALIDATE_STATE(s); + mapped = ((file->f_mode & FMODE_WRITE) && s->dma_dac.mapped) || + ((file->f_mode & FMODE_READ) && s->dma_adc.mapped); + switch (cmd) { + case OSS_GETVERSION: + CS_DBGOUT(CS_IOCTL | CS_PARMS, 4, printk(KERN_INFO + "cs4297a: cs4297a_ioctl(): SOUND_VERSION=0x%.8x\n", + SOUND_VERSION)); + return put_user(SOUND_VERSION, (int *) arg); + + case SNDCTL_DSP_SYNC: + CS_DBGOUT(CS_IOCTL, 4, printk(KERN_INFO + "cs4297a: cs4297a_ioctl(): DSP_SYNC\n")); + if (file->f_mode & FMODE_WRITE) + return drain_dac(s, + 0 /*file->f_flags & O_NONBLOCK */ + ); + return 0; + + case SNDCTL_DSP_SETDUPLEX: + return 0; + + case SNDCTL_DSP_GETCAPS: + return put_user(DSP_CAP_DUPLEX | DSP_CAP_REALTIME | + DSP_CAP_TRIGGER | DSP_CAP_MMAP, + (int *) arg); + + case SNDCTL_DSP_RESET: + CS_DBGOUT(CS_IOCTL, 4, printk(KERN_INFO + "cs4297a: cs4297a_ioctl(): DSP_RESET\n")); + if (file->f_mode & FMODE_WRITE) { + stop_dac(s); + synchronize_irq(s->irq); + s->dma_dac.count = s->dma_dac.total_bytes = + s->dma_dac.blocks = s->dma_dac.wakeup = 0; + s->dma_dac.swptr = s->dma_dac.hwptr = + (int)(((__raw_readq(SS_CSR(R_SER_DMA_CUR_DSCR_ADDR_TX)) & M_DMA_CURDSCR_ADDR) - + s->dma_dac.descrtab_phys) / sizeof(serdma_descr_t)); + } + if (file->f_mode & FMODE_READ) { + stop_adc(s); + synchronize_irq(s->irq); + s->dma_adc.count = s->dma_adc.total_bytes = + s->dma_adc.blocks = s->dma_dac.wakeup = 0; + s->dma_adc.swptr = s->dma_adc.hwptr = + (int)(((__raw_readq(SS_CSR(R_SER_DMA_CUR_DSCR_ADDR_RX)) & M_DMA_CURDSCR_ADDR) - + s->dma_adc.descrtab_phys) / sizeof(serdma_descr_t)); + } + return 0; + + case SNDCTL_DSP_SPEED: + if (get_user(val, (int *) arg)) + return -EFAULT; + CS_DBGOUT(CS_IOCTL | CS_PARMS, 4, printk(KERN_INFO + "cs4297a: cs4297a_ioctl(): DSP_SPEED val=%d -> 48000\n", val)); + val = 48000; + return put_user(val, (int *) arg); + + case SNDCTL_DSP_STEREO: + if (get_user(val, (int *) arg)) + return -EFAULT; + CS_DBGOUT(CS_IOCTL | CS_PARMS, 4, printk(KERN_INFO + "cs4297a: cs4297a_ioctl(): DSP_STEREO val=%d\n", val)); + if (file->f_mode & FMODE_READ) { + stop_adc(s); + s->dma_adc.ready = 0; + s->prop_adc.channels = val ? 2 : 1; + } + if (file->f_mode & FMODE_WRITE) { + stop_dac(s); + s->dma_dac.ready = 0; + s->prop_dac.channels = val ? 2 : 1; + } + return 0; + + case SNDCTL_DSP_CHANNELS: + if (get_user(val, (int *) arg)) + return -EFAULT; + CS_DBGOUT(CS_IOCTL | CS_PARMS, 4, printk(KERN_INFO + "cs4297a: cs4297a_ioctl(): DSP_CHANNELS val=%d\n", + val)); + if (val != 0) { + if (file->f_mode & FMODE_READ) { + stop_adc(s); + s->dma_adc.ready = 0; + if (val >= 2) + s->prop_adc.channels = 2; + else + s->prop_adc.channels = 1; + } + if (file->f_mode & FMODE_WRITE) { + stop_dac(s); + s->dma_dac.ready = 0; + if (val >= 2) + s->prop_dac.channels = 2; + else + s->prop_dac.channels = 1; + } + } + + if (file->f_mode & FMODE_WRITE) + val = s->prop_dac.channels; + else if (file->f_mode & FMODE_READ) + val = s->prop_adc.channels; + + return put_user(val, (int *) arg); + + case SNDCTL_DSP_GETFMTS: // Returns a mask + CS_DBGOUT(CS_IOCTL | CS_PARMS, 4, printk(KERN_INFO + "cs4297a: cs4297a_ioctl(): DSP_GETFMT val=0x%.8x\n", + AFMT_S16_LE | AFMT_U16_LE | AFMT_S8 | + AFMT_U8)); + return put_user(AFMT_S16_LE | AFMT_U16_LE | AFMT_S8 | + AFMT_U8, (int *) arg); + + case SNDCTL_DSP_SETFMT: + if (get_user(val, (int *) arg)) + return -EFAULT; + CS_DBGOUT(CS_IOCTL | CS_PARMS, 4, printk(KERN_INFO + "cs4297a: cs4297a_ioctl(): DSP_SETFMT val=0x%.8x\n", + val)); + if (val != AFMT_QUERY) { + if (file->f_mode & FMODE_READ) { + stop_adc(s); + s->dma_adc.ready = 0; + if (val != AFMT_S16_LE + && val != AFMT_U16_LE && val != AFMT_S8 + && val != AFMT_U8) + val = AFMT_U8; + s->prop_adc.fmt = val; + s->prop_adc.fmt_original = s->prop_adc.fmt; + } + if (file->f_mode & FMODE_WRITE) { + stop_dac(s); + s->dma_dac.ready = 0; + if (val != AFMT_S16_LE + && val != AFMT_U16_LE && val != AFMT_S8 + && val != AFMT_U8) + val = AFMT_U8; + s->prop_dac.fmt = val; + s->prop_dac.fmt_original = s->prop_dac.fmt; + } + } else { + if (file->f_mode & FMODE_WRITE) + val = s->prop_dac.fmt_original; + else if (file->f_mode & FMODE_READ) + val = s->prop_adc.fmt_original; + } + CS_DBGOUT(CS_IOCTL | CS_PARMS, 4, printk(KERN_INFO + "cs4297a: cs4297a_ioctl(): DSP_SETFMT return val=0x%.8x\n", + val)); + return put_user(val, (int *) arg); + + case SNDCTL_DSP_POST: + CS_DBGOUT(CS_IOCTL, 4, printk(KERN_INFO + "cs4297a: cs4297a_ioctl(): DSP_POST\n")); + return 0; + + case SNDCTL_DSP_GETTRIGGER: + val = 0; + if (file->f_mode & s->ena & FMODE_READ) + val |= PCM_ENABLE_INPUT; + if (file->f_mode & s->ena & FMODE_WRITE) + val |= PCM_ENABLE_OUTPUT; + return put_user(val, (int *) arg); + + case SNDCTL_DSP_SETTRIGGER: + if (get_user(val, (int *) arg)) + return -EFAULT; + if (file->f_mode & FMODE_READ) { + if (val & PCM_ENABLE_INPUT) { + if (!s->dma_adc.ready + && (ret = prog_dmabuf_adc(s))) + return ret; + start_adc(s); + } else + stop_adc(s); + } + if (file->f_mode & FMODE_WRITE) { + if (val & PCM_ENABLE_OUTPUT) { + if (!s->dma_dac.ready + && (ret = prog_dmabuf_dac(s))) + return ret; + start_dac(s); + } else + stop_dac(s); + } + return 0; + + case SNDCTL_DSP_GETOSPACE: + if (!(file->f_mode & FMODE_WRITE)) + return -EINVAL; + if (!s->dma_dac.ready && (val = prog_dmabuf_dac(s))) + return val; + spin_lock_irqsave(&s->lock, flags); + cs4297a_update_ptr(s,CS_FALSE); + abinfo.fragsize = s->dma_dac.fragsize; + if (s->dma_dac.mapped) + abinfo.bytes = s->dma_dac.sbufsz; + else + abinfo.bytes = + s->dma_dac.sbufsz - s->dma_dac.count; + abinfo.fragstotal = s->dma_dac.numfrag; + abinfo.fragments = abinfo.bytes >> s->dma_dac.fragshift; + CS_DBGOUT(CS_FUNCTION | CS_PARMS, 4, printk(KERN_INFO + "cs4297a: cs4297a_ioctl(): GETOSPACE .fragsize=%d .bytes=%d .fragstotal=%d .fragments=%d\n", + abinfo.fragsize,abinfo.bytes,abinfo.fragstotal, + abinfo.fragments)); + spin_unlock_irqrestore(&s->lock, flags); + return copy_to_user((void *) arg, &abinfo, + sizeof(abinfo)) ? -EFAULT : 0; + + case SNDCTL_DSP_GETISPACE: + if (!(file->f_mode & FMODE_READ)) + return -EINVAL; + if (!s->dma_adc.ready && (val = prog_dmabuf_adc(s))) + return val; + spin_lock_irqsave(&s->lock, flags); + cs4297a_update_ptr(s,CS_FALSE); + if (s->conversion) { + abinfo.fragsize = s->dma_adc.fragsize / 2; + abinfo.bytes = s->dma_adc.count / 2; + abinfo.fragstotal = s->dma_adc.numfrag; + abinfo.fragments = + abinfo.bytes >> (s->dma_adc.fragshift - 1); + } else { + abinfo.fragsize = s->dma_adc.fragsize; + abinfo.bytes = s->dma_adc.count; + abinfo.fragstotal = s->dma_adc.numfrag; + abinfo.fragments = + abinfo.bytes >> s->dma_adc.fragshift; + } + spin_unlock_irqrestore(&s->lock, flags); + return copy_to_user((void *) arg, &abinfo, + sizeof(abinfo)) ? -EFAULT : 0; + + case SNDCTL_DSP_NONBLOCK: + spin_lock(&file->f_lock); + file->f_flags |= O_NONBLOCK; + spin_unlock(&file->f_lock); + return 0; + + case SNDCTL_DSP_GETODELAY: + if (!(file->f_mode & FMODE_WRITE)) + return -EINVAL; + if(!s->dma_dac.ready && prog_dmabuf_dac(s)) + return 0; + spin_lock_irqsave(&s->lock, flags); + cs4297a_update_ptr(s,CS_FALSE); + val = s->dma_dac.count; + spin_unlock_irqrestore(&s->lock, flags); + return put_user(val, (int *) arg); + + case SNDCTL_DSP_GETIPTR: + if (!(file->f_mode & FMODE_READ)) + return -EINVAL; + if(!s->dma_adc.ready && prog_dmabuf_adc(s)) + return 0; + spin_lock_irqsave(&s->lock, flags); + cs4297a_update_ptr(s,CS_FALSE); + cinfo.bytes = s->dma_adc.total_bytes; + if (s->dma_adc.mapped) { + cinfo.blocks = + (cinfo.bytes >> s->dma_adc.fragshift) - + s->dma_adc.blocks; + s->dma_adc.blocks = + cinfo.bytes >> s->dma_adc.fragshift; + } else { + if (s->conversion) { + cinfo.blocks = + s->dma_adc.count / + 2 >> (s->dma_adc.fragshift - 1); + } else + cinfo.blocks = + s->dma_adc.count >> s->dma_adc. + fragshift; + } + if (s->conversion) + cinfo.ptr = s->dma_adc.hwptr / 2; + else + cinfo.ptr = s->dma_adc.hwptr; + if (s->dma_adc.mapped) + s->dma_adc.count &= s->dma_adc.fragsize - 1; + spin_unlock_irqrestore(&s->lock, flags); + return copy_to_user((void *) arg, &cinfo, sizeof(cinfo)) ? -EFAULT : 0; + + case SNDCTL_DSP_GETOPTR: + if (!(file->f_mode & FMODE_WRITE)) + return -EINVAL; + if(!s->dma_dac.ready && prog_dmabuf_dac(s)) + return 0; + spin_lock_irqsave(&s->lock, flags); + cs4297a_update_ptr(s,CS_FALSE); + cinfo.bytes = s->dma_dac.total_bytes; + if (s->dma_dac.mapped) { + cinfo.blocks = + (cinfo.bytes >> s->dma_dac.fragshift) - + s->dma_dac.blocks; + s->dma_dac.blocks = + cinfo.bytes >> s->dma_dac.fragshift; + } else { + cinfo.blocks = + s->dma_dac.count >> s->dma_dac.fragshift; + } + cinfo.ptr = s->dma_dac.hwptr; + if (s->dma_dac.mapped) + s->dma_dac.count &= s->dma_dac.fragsize - 1; + spin_unlock_irqrestore(&s->lock, flags); + return copy_to_user((void *) arg, &cinfo, sizeof(cinfo)) ? -EFAULT : 0; + + case SNDCTL_DSP_GETBLKSIZE: + if (file->f_mode & FMODE_WRITE) { + if ((val = prog_dmabuf_dac(s))) + return val; + return put_user(s->dma_dac.fragsize, (int *) arg); + } + if ((val = prog_dmabuf_adc(s))) + return val; + if (s->conversion) + return put_user(s->dma_adc.fragsize / 2, + (int *) arg); + else + return put_user(s->dma_adc.fragsize, (int *) arg); + + case SNDCTL_DSP_SETFRAGMENT: + if (get_user(val, (int *) arg)) + return -EFAULT; + return 0; // Say OK, but do nothing. + + case SNDCTL_DSP_SUBDIVIDE: + if ((file->f_mode & FMODE_READ && s->dma_adc.subdivision) + || (file->f_mode & FMODE_WRITE + && s->dma_dac.subdivision)) return -EINVAL; + if (get_user(val, (int *) arg)) + return -EFAULT; + if (val != 1 && val != 2 && val != 4) + return -EINVAL; + if (file->f_mode & FMODE_READ) + s->dma_adc.subdivision = val; + else if (file->f_mode & FMODE_WRITE) + s->dma_dac.subdivision = val; + return 0; + + case SOUND_PCM_READ_RATE: + if (file->f_mode & FMODE_READ) + return put_user(s->prop_adc.rate, (int *) arg); + else if (file->f_mode & FMODE_WRITE) + return put_user(s->prop_dac.rate, (int *) arg); + + case SOUND_PCM_READ_CHANNELS: + if (file->f_mode & FMODE_READ) + return put_user(s->prop_adc.channels, (int *) arg); + else if (file->f_mode & FMODE_WRITE) + return put_user(s->prop_dac.channels, (int *) arg); + + case SOUND_PCM_READ_BITS: + if (file->f_mode & FMODE_READ) + return + put_user( + (s->prop_adc. + fmt & (AFMT_S8 | AFMT_U8)) ? 8 : 16, + (int *) arg); + else if (file->f_mode & FMODE_WRITE) + return + put_user( + (s->prop_dac. + fmt & (AFMT_S8 | AFMT_U8)) ? 8 : 16, + (int *) arg); + + case SOUND_PCM_WRITE_FILTER: + case SNDCTL_DSP_SETSYNCRO: + case SOUND_PCM_READ_FILTER: + return -EINVAL; + } + return mixer_ioctl(s, cmd, arg); +} + +static long cs4297a_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) +{ + int ret; + + mutex_lock(&swarm_cs4297a_mutex); + ret = cs4297a_ioctl(file, cmd, arg); + mutex_unlock(&swarm_cs4297a_mutex); + + return ret; +} + +static int cs4297a_release(struct inode *inode, struct file *file) +{ + struct cs4297a_state *s = + (struct cs4297a_state *) file->private_data; + + CS_DBGOUT(CS_FUNCTION | CS_RELEASE, 2, printk(KERN_INFO + "cs4297a: cs4297a_release(): inode=0x%.8x file=0x%.8x f_mode=0x%x\n", + (unsigned) inode, (unsigned) file, file->f_mode)); + VALIDATE_STATE(s); + + if (file->f_mode & FMODE_WRITE) { + drain_dac(s, file->f_flags & O_NONBLOCK); + mutex_lock(&s->open_sem_dac); + stop_dac(s); + dealloc_dmabuf(s, &s->dma_dac); + s->open_mode &= ~FMODE_WRITE; + mutex_unlock(&s->open_sem_dac); + wake_up(&s->open_wait_dac); + } + if (file->f_mode & FMODE_READ) { + drain_adc(s, file->f_flags & O_NONBLOCK); + mutex_lock(&s->open_sem_adc); + stop_adc(s); + dealloc_dmabuf(s, &s->dma_adc); + s->open_mode &= ~FMODE_READ; + mutex_unlock(&s->open_sem_adc); + wake_up(&s->open_wait_adc); + } + return 0; +} + +static int cs4297a_locked_open(struct inode *inode, struct file *file) +{ + int minor = iminor(inode); + struct cs4297a_state *s=NULL; + struct list_head *entry; + + CS_DBGOUT(CS_FUNCTION | CS_OPEN, 2, printk(KERN_INFO + "cs4297a: cs4297a_open(): inode=0x%.8x file=0x%.8x f_mode=0x%x\n", + (unsigned) inode, (unsigned) file, file->f_mode)); + CS_DBGOUT(CS_FUNCTION | CS_OPEN, 2, printk(KERN_INFO + "cs4297a: status = %08x\n", (int)__raw_readq(SS_CSR(R_SER_STATUS_DEBUG)))); + + list_for_each(entry, &cs4297a_devs) + { + s = list_entry(entry, struct cs4297a_state, list); + + if (!((s->dev_audio ^ minor) & ~0xf)) + break; + } + if (entry == &cs4297a_devs) + return -ENODEV; + if (!s) { + CS_DBGOUT(CS_FUNCTION | CS_OPEN, 2, printk(KERN_INFO + "cs4297a: cs4297a_open(): Error - unable to find audio state struct\n")); + return -ENODEV; + } + VALIDATE_STATE(s); + file->private_data = s; + + // wait for device to become free + if (!(file->f_mode & (FMODE_WRITE | FMODE_READ))) { + CS_DBGOUT(CS_FUNCTION | CS_OPEN | CS_ERROR, 2, printk(KERN_INFO + "cs4297a: cs4297a_open(): Error - must open READ and/or WRITE\n")); + return -ENODEV; + } + if (file->f_mode & FMODE_WRITE) { + if (__raw_readq(SS_CSR(R_SER_DMA_DSCR_COUNT_TX)) != 0) { + printk(KERN_ERR "cs4297a: TX pipe needs to drain\n"); + while (__raw_readq(SS_CSR(R_SER_DMA_DSCR_COUNT_TX))) + ; + } + + mutex_lock(&s->open_sem_dac); + while (s->open_mode & FMODE_WRITE) { + if (file->f_flags & O_NONBLOCK) { + mutex_unlock(&s->open_sem_dac); + return -EBUSY; + } + mutex_unlock(&s->open_sem_dac); + interruptible_sleep_on(&s->open_wait_dac); + + if (signal_pending(current)) { + printk("open - sig pending\n"); + return -ERESTARTSYS; + } + mutex_lock(&s->open_sem_dac); + } + } + if (file->f_mode & FMODE_READ) { + mutex_lock(&s->open_sem_adc); + while (s->open_mode & FMODE_READ) { + if (file->f_flags & O_NONBLOCK) { + mutex_unlock(&s->open_sem_adc); + return -EBUSY; + } + mutex_unlock(&s->open_sem_adc); + interruptible_sleep_on(&s->open_wait_adc); + + if (signal_pending(current)) { + printk("open - sig pending\n"); + return -ERESTARTSYS; + } + mutex_lock(&s->open_sem_adc); + } + } + s->open_mode |= file->f_mode & (FMODE_READ | FMODE_WRITE); + if (file->f_mode & FMODE_READ) { + s->prop_adc.fmt = AFMT_S16_BE; + s->prop_adc.fmt_original = s->prop_adc.fmt; + s->prop_adc.channels = 2; + s->prop_adc.rate = 48000; + s->conversion = 0; + s->ena &= ~FMODE_READ; + s->dma_adc.ossfragshift = s->dma_adc.ossmaxfrags = + s->dma_adc.subdivision = 0; + mutex_unlock(&s->open_sem_adc); + + if (prog_dmabuf_adc(s)) { + CS_DBGOUT(CS_OPEN | CS_ERROR, 2, printk(KERN_ERR + "cs4297a: adc Program dmabufs failed.\n")); + cs4297a_release(inode, file); + return -ENOMEM; + } + } + if (file->f_mode & FMODE_WRITE) { + s->prop_dac.fmt = AFMT_S16_BE; + s->prop_dac.fmt_original = s->prop_dac.fmt; + s->prop_dac.channels = 2; + s->prop_dac.rate = 48000; + s->conversion = 0; + s->ena &= ~FMODE_WRITE; + s->dma_dac.ossfragshift = s->dma_dac.ossmaxfrags = + s->dma_dac.subdivision = 0; + mutex_unlock(&s->open_sem_dac); + + if (prog_dmabuf_dac(s)) { + CS_DBGOUT(CS_OPEN | CS_ERROR, 2, printk(KERN_ERR + "cs4297a: dac Program dmabufs failed.\n")); + cs4297a_release(inode, file); + return -ENOMEM; + } + } + CS_DBGOUT(CS_FUNCTION | CS_OPEN, 2, + printk(KERN_INFO "cs4297a: cs4297a_open()- 0\n")); + return nonseekable_open(inode, file); +} + +static int cs4297a_open(struct inode *inode, struct file *file) +{ + int ret; + + mutex_lock(&swarm_cs4297a_mutex); + ret = cs4297a_open(inode, file); + mutex_unlock(&swarm_cs4297a_mutex); + + return ret; +} + +// ****************************************************************************************** +// Wave (audio) file operations struct. +// ****************************************************************************************** +static const struct file_operations cs4297a_audio_fops = { + .owner = THIS_MODULE, + .llseek = no_llseek, + .read = cs4297a_read, + .write = cs4297a_write, + .poll = cs4297a_poll, + .unlocked_ioctl = cs4297a_unlocked_ioctl, + .mmap = cs4297a_mmap, + .open = cs4297a_open, + .release = cs4297a_release, +}; + +static void cs4297a_interrupt(int irq, void *dev_id) +{ + struct cs4297a_state *s = (struct cs4297a_state *) dev_id; + u32 status; + + status = __raw_readq(SS_CSR(R_SER_STATUS_DEBUG)); + + CS_DBGOUT(CS_INTERRUPT, 6, printk(KERN_INFO + "cs4297a: cs4297a_interrupt() HISR=0x%.8x\n", status)); + +#if 0 + /* XXXKW what check *should* be done here? */ + if (!(status & (M_SYNCSER_RX_EOP_COUNT | M_SYNCSER_RX_OVERRUN | M_SYNCSER_RX_SYNC_ERR))) { + status = __raw_readq(SS_CSR(R_SER_STATUS)); + printk(KERN_ERR "cs4297a: unexpected interrupt (status %08x)\n", status); + return; + } +#endif + + if (status & M_SYNCSER_RX_SYNC_ERR) { + status = __raw_readq(SS_CSR(R_SER_STATUS)); + printk(KERN_ERR "cs4297a: rx sync error (status %08x)\n", status); + return; + } + + if (status & M_SYNCSER_RX_OVERRUN) { + int newptr, i; + s->stats.rx_ovrrn++; + printk(KERN_ERR "cs4297a: receive FIFO overrun\n"); + + /* Fix things up: get the receive descriptor pool + clean and give them back to the hardware */ + while (__raw_readq(SS_CSR(R_SER_DMA_DSCR_COUNT_RX))) + ; + newptr = (unsigned) (((__raw_readq(SS_CSR(R_SER_DMA_CUR_DSCR_ADDR_RX)) & M_DMA_CURDSCR_ADDR) - + s->dma_adc.descrtab_phys) / sizeof(serdma_descr_t)); + for (i=0; i<DMA_DESCR; i++) { + s->dma_adc.descrtab[i].descr_a &= ~M_DMA_SERRX_SOP; + } + s->dma_adc.swptr = s->dma_adc.hwptr = newptr; + s->dma_adc.count = 0; + s->dma_adc.sb_swptr = s->dma_adc.sb_hwptr = s->dma_adc.sample_buf; + __raw_writeq(DMA_DESCR, SS_CSR(R_SER_DMA_DSCR_COUNT_RX)); + } + + spin_lock(&s->lock); + cs4297a_update_ptr(s,CS_TRUE); + spin_unlock(&s->lock); + + CS_DBGOUT(CS_INTERRUPT, 6, printk(KERN_INFO + "cs4297a: cs4297a_interrupt()-\n")); +} + +#if 0 +static struct initvol { + int mixch; + int vol; +} initvol[] __initdata = { + + {SOUND_MIXER_WRITE_VOLUME, 0x4040}, + {SOUND_MIXER_WRITE_PCM, 0x4040}, + {SOUND_MIXER_WRITE_SYNTH, 0x4040}, + {SOUND_MIXER_WRITE_CD, 0x4040}, + {SOUND_MIXER_WRITE_LINE, 0x4040}, + {SOUND_MIXER_WRITE_LINE1, 0x4040}, + {SOUND_MIXER_WRITE_RECLEV, 0x0000}, + {SOUND_MIXER_WRITE_SPEAKER, 0x4040}, + {SOUND_MIXER_WRITE_MIC, 0x0000} +}; +#endif + +static int __init cs4297a_init(void) +{ + struct cs4297a_state *s; + u32 pwr, id; + mm_segment_t fs; + int rval; +#ifndef CONFIG_BCM_CS4297A_CSWARM + u64 cfg; + int mdio_val; +#endif + + CS_DBGOUT(CS_INIT | CS_FUNCTION, 2, printk(KERN_INFO + "cs4297a: cs4297a_init_module()+ \n")); + +#ifndef CONFIG_BCM_CS4297A_CSWARM + mdio_val = __raw_readq(KSEG1 + A_MAC_REGISTER(2, R_MAC_MDIO)) & + (M_MAC_MDIO_DIR|M_MAC_MDIO_OUT); + + /* Check syscfg for synchronous serial on port 1 */ + cfg = __raw_readq(KSEG1 + A_SCD_SYSTEM_CFG); + if (!(cfg & M_SYS_SER1_ENABLE)) { + __raw_writeq(cfg | M_SYS_SER1_ENABLE, KSEG1+A_SCD_SYSTEM_CFG); + cfg = __raw_readq(KSEG1 + A_SCD_SYSTEM_CFG); + if (!(cfg & M_SYS_SER1_ENABLE)) { + printk(KERN_INFO "cs4297a: serial port 1 not configured for synchronous operation\n"); + return -1; + } + + printk(KERN_INFO "cs4297a: serial port 1 switching to synchronous operation\n"); + + /* Force the codec (on SWARM) to reset by clearing + GENO, preserving MDIO (no effect on CSWARM) */ + __raw_writeq(mdio_val, KSEG1+A_MAC_REGISTER(2, R_MAC_MDIO)); + udelay(10); + } + + /* Now set GENO */ + __raw_writeq(mdio_val | M_MAC_GENC, KSEG1+A_MAC_REGISTER(2, R_MAC_MDIO)); + /* Give the codec some time to finish resetting (start the bit clock) */ + udelay(100); +#endif + + if (!(s = kzalloc(sizeof(struct cs4297a_state), GFP_KERNEL))) { + CS_DBGOUT(CS_ERROR, 1, printk(KERN_ERR + "cs4297a: probe() no memory for state struct.\n")); + return -1; + } + s->magic = CS4297a_MAGIC; + init_waitqueue_head(&s->dma_adc.wait); + init_waitqueue_head(&s->dma_dac.wait); + init_waitqueue_head(&s->dma_adc.reg_wait); + init_waitqueue_head(&s->dma_dac.reg_wait); + init_waitqueue_head(&s->open_wait); + init_waitqueue_head(&s->open_wait_adc); + init_waitqueue_head(&s->open_wait_dac); + mutex_init(&s->open_sem_adc); + mutex_init(&s->open_sem_dac); + spin_lock_init(&s->lock); + + s->irq = K_INT_SER_1; + + if (request_irq + (s->irq, cs4297a_interrupt, 0, "Crystal CS4297a", s)) { + CS_DBGOUT(CS_INIT | CS_ERROR, 1, + printk(KERN_ERR "cs4297a: irq %u in use\n", s->irq)); + goto err_irq; + } + if ((s->dev_audio = register_sound_dsp(&cs4297a_audio_fops, -1)) < + 0) { + CS_DBGOUT(CS_INIT | CS_ERROR, 1, printk(KERN_ERR + "cs4297a: probe() register_sound_dsp() failed.\n")); + goto err_dev1; + } + if ((s->dev_mixer = register_sound_mixer(&cs4297a_mixer_fops, -1)) < + 0) { + CS_DBGOUT(CS_INIT | CS_ERROR, 1, printk(KERN_ERR + "cs4297a: probe() register_sound_mixer() failed.\n")); + goto err_dev2; + } + + if (ser_init(s) || dma_init(s)) { + CS_DBGOUT(CS_INIT | CS_ERROR, 1, printk(KERN_ERR + "cs4297a: ser_init failed.\n")); + goto err_dev3; + } + + do { + udelay(4000); + rval = cs4297a_read_ac97(s, AC97_POWER_CONTROL, &pwr); + } while (!rval && (pwr != 0xf)); + + if (!rval) { + char *sb1250_duart_present; + + fs = get_fs(); + set_fs(KERNEL_DS); +#if 0 + val = SOUND_MASK_LINE; + mixer_ioctl(s, SOUND_MIXER_WRITE_RECSRC, (unsigned long) &val); + for (i = 0; i < ARRAY_SIZE(initvol); i++) { + val = initvol[i].vol; + mixer_ioctl(s, initvol[i].mixch, (unsigned long) &val); + } +// cs4297a_write_ac97(s, 0x18, 0x0808); +#else + // cs4297a_write_ac97(s, 0x5e, 0x180); + cs4297a_write_ac97(s, 0x02, 0x0808); + cs4297a_write_ac97(s, 0x18, 0x0808); +#endif + set_fs(fs); + + list_add(&s->list, &cs4297a_devs); + + cs4297a_read_ac97(s, AC97_VENDOR_ID1, &id); + + sb1250_duart_present = symbol_get(sb1250_duart_present); + if (sb1250_duart_present) + sb1250_duart_present[1] = 0; + + printk(KERN_INFO "cs4297a: initialized (vendor id = %x)\n", id); + + CS_DBGOUT(CS_INIT | CS_FUNCTION, 2, + printk(KERN_INFO "cs4297a: cs4297a_init_module()-\n")); + + return 0; + } + + err_dev3: + unregister_sound_mixer(s->dev_mixer); + err_dev2: + unregister_sound_dsp(s->dev_audio); + err_dev1: + free_irq(s->irq, s); + err_irq: + kfree(s); + + printk(KERN_INFO "cs4297a: initialization failed\n"); + + return -1; +} + +static void __exit cs4297a_cleanup(void) +{ + /* + XXXKW + disable_irq, free_irq + drain DMA queue + disable DMA + disable TX/RX + free memory + */ + CS_DBGOUT(CS_INIT | CS_FUNCTION, 2, + printk(KERN_INFO "cs4297a: cleanup_cs4297a() finished\n")); +} + +// --------------------------------------------------------------------- + +MODULE_AUTHOR("Kip Walker, Broadcom Corp."); +MODULE_DESCRIPTION("Cirrus Logic CS4297a Driver for Broadcom SWARM board"); + +// --------------------------------------------------------------------- + +module_init(cs4297a_init); +module_exit(cs4297a_cleanup); diff --git a/sound/oss/sys_timer.c b/sound/oss/sys_timer.c new file mode 100644 index 00000000..8db6aefe --- /dev/null +++ b/sound/oss/sys_timer.c @@ -0,0 +1,285 @@ +/* + * sound/oss/sys_timer.c + * + * The default timer for the Level 2 sequencer interface + * Uses the (1/HZ sec) timer of kernel. + */ +/* + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + */ +/* + * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed) + * Andrew Veliath : adapted tmr2ticks from level 1 sequencer (avoid overflow) + */ +#include <linux/spinlock.h> +#include "sound_config.h" + +static volatile int opened, tmr_running; +static volatile time_t tmr_offs, tmr_ctr; +static volatile unsigned long ticks_offs; +static volatile int curr_tempo, curr_timebase; +static volatile unsigned long curr_ticks; +static volatile unsigned long next_event_time; +static unsigned long prev_event_time; + +static void poll_def_tmr(unsigned long dummy); +static DEFINE_SPINLOCK(lock); +static DEFINE_TIMER(def_tmr, poll_def_tmr, 0, 0); + +static unsigned long +tmr2ticks(int tmr_value) +{ + /* + * Convert timer ticks to MIDI ticks + */ + + unsigned long tmp; + unsigned long scale; + + /* tmr_value (ticks per sec) * + 1000000 (usecs per sec) / HZ (ticks per sec) -=> usecs */ + tmp = tmr_value * (1000000 / HZ); + scale = (60 * 1000000) / (curr_tempo * curr_timebase); /* usecs per MIDI tick */ + return (tmp + scale / 2) / scale; +} + +static void +poll_def_tmr(unsigned long dummy) +{ + + if (opened) + { + + { + def_tmr.expires = (1) + jiffies; + add_timer(&def_tmr); + }; + + if (tmr_running) + { + spin_lock(&lock); + tmr_ctr++; + curr_ticks = ticks_offs + tmr2ticks(tmr_ctr); + + if (curr_ticks >= next_event_time) + { + next_event_time = (unsigned long) -1; + sequencer_timer(0); + } + spin_unlock(&lock); + } + } +} + +static void +tmr_reset(void) +{ + unsigned long flags; + + spin_lock_irqsave(&lock,flags); + tmr_offs = 0; + ticks_offs = 0; + tmr_ctr = 0; + next_event_time = (unsigned long) -1; + prev_event_time = 0; + curr_ticks = 0; + spin_unlock_irqrestore(&lock,flags); +} + +static int +def_tmr_open(int dev, int mode) +{ + if (opened) + return -EBUSY; + + tmr_reset(); + curr_tempo = 60; + curr_timebase = 100; + opened = 1; + { + def_tmr.expires = (1) + jiffies; + add_timer(&def_tmr); + }; + + return 0; +} + +static void +def_tmr_close(int dev) +{ + opened = tmr_running = 0; + del_timer(&def_tmr); +} + +static int +def_tmr_event(int dev, unsigned char *event) +{ + unsigned char cmd = event[1]; + unsigned long parm = *(int *) &event[4]; + + switch (cmd) + { + case TMR_WAIT_REL: + parm += prev_event_time; + case TMR_WAIT_ABS: + if (parm > 0) + { + long time; + + if (parm <= curr_ticks) /* It's the time */ + return TIMER_NOT_ARMED; + + time = parm; + next_event_time = prev_event_time = time; + + return TIMER_ARMED; + } + break; + + case TMR_START: + tmr_reset(); + tmr_running = 1; + break; + + case TMR_STOP: + tmr_running = 0; + break; + + case TMR_CONTINUE: + tmr_running = 1; + break; + + case TMR_TEMPO: + if (parm) + { + if (parm < 8) + parm = 8; + if (parm > 360) + parm = 360; + tmr_offs = tmr_ctr; + ticks_offs += tmr2ticks(tmr_ctr); + tmr_ctr = 0; + curr_tempo = parm; + } + break; + + case TMR_ECHO: + seq_copy_to_input(event, 8); + break; + + default:; + } + + return TIMER_NOT_ARMED; +} + +static unsigned long +def_tmr_get_time(int dev) +{ + if (!opened) + return 0; + + return curr_ticks; +} + +/* same as sound_timer.c:timer_ioctl!? */ +static int def_tmr_ioctl(int dev, unsigned int cmd, void __user *arg) +{ + int __user *p = arg; + int val; + + switch (cmd) { + case SNDCTL_TMR_SOURCE: + return __put_user(TMR_INTERNAL, p); + + case SNDCTL_TMR_START: + tmr_reset(); + tmr_running = 1; + return 0; + + case SNDCTL_TMR_STOP: + tmr_running = 0; + return 0; + + case SNDCTL_TMR_CONTINUE: + tmr_running = 1; + return 0; + + case SNDCTL_TMR_TIMEBASE: + if (__get_user(val, p)) + return -EFAULT; + if (val) { + if (val < 1) + val = 1; + if (val > 1000) + val = 1000; + curr_timebase = val; + } + return __put_user(curr_timebase, p); + + case SNDCTL_TMR_TEMPO: + if (__get_user(val, p)) + return -EFAULT; + if (val) { + if (val < 8) + val = 8; + if (val > 250) + val = 250; + tmr_offs = tmr_ctr; + ticks_offs += tmr2ticks(tmr_ctr); + tmr_ctr = 0; + curr_tempo = val; + reprogram_timer(); + } + return __put_user(curr_tempo, p); + + case SNDCTL_SEQ_CTRLRATE: + if (__get_user(val, p)) + return -EFAULT; + if (val != 0) /* Can't change */ + return -EINVAL; + val = ((curr_tempo * curr_timebase) + 30) / 60; + return __put_user(val, p); + + case SNDCTL_SEQ_GETTIME: + return __put_user(curr_ticks, p); + + case SNDCTL_TMR_METRONOME: + /* NOP */ + break; + + default:; + } + return -EINVAL; +} + +static void +def_tmr_arm(int dev, long time) +{ + if (time < 0) + time = curr_ticks + 1; + else if (time <= curr_ticks) /* It's the time */ + return; + + next_event_time = prev_event_time = time; + + return; +} + +struct sound_timer_operations default_sound_timer = +{ + .owner = THIS_MODULE, + .info = {"System clock", 0}, + .priority = 0, /* Priority */ + .devlink = 0, /* Local device link */ + .open = def_tmr_open, + .close = def_tmr_close, + .event = def_tmr_event, + .get_time = def_tmr_get_time, + .ioctl = def_tmr_ioctl, + .arm_timer = def_tmr_arm +}; diff --git a/sound/oss/trix.c b/sound/oss/trix.c new file mode 100644 index 00000000..e04169e8 --- /dev/null +++ b/sound/oss/trix.c @@ -0,0 +1,525 @@ +/* + * sound/oss/trix.c + * + * Low level driver for the MediaTrix AudioTrix Pro + * (MT-0002-PC Control Chip) + * + * + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * + * Changes + * Alan Cox Modularisation, cleanup. + * Christoph Hellwig Adapted to module_init/module_exit + * Arnaldo C. de Melo Got rid of attach_uart401 + */ + +#include <linux/init.h> +#include <linux/module.h> + +#include "sound_config.h" +#include "sb.h" +#include "sound_firmware.h" + +#include "ad1848.h" +#include "mpu401.h" + +#include "trix_boot.h" + +static int mpu; + +static int joystick; + +static unsigned char trix_read(int addr) +{ + outb(((unsigned char) addr), 0x390); /* MT-0002-PC ASIC address */ + return inb(0x391); /* MT-0002-PC ASIC data */ +} + +static void trix_write(int addr, int data) +{ + outb(((unsigned char) addr), 0x390); /* MT-0002-PC ASIC address */ + outb(((unsigned char) data), 0x391); /* MT-0002-PC ASIC data */ +} + +static void download_boot(int base) +{ + int i = 0, n = trix_boot_len; + + if (trix_boot_len == 0) + return; + + trix_write(0xf8, 0x00); /* ??????? */ + outb((0x01), base + 6); /* Clear the internal data pointer */ + outb((0x00), base + 6); /* Restart */ + + /* + * Write the boot code to the RAM upload/download register. + * Each write increments the internal data pointer. + */ + outb((0x01), base + 6); /* Clear the internal data pointer */ + outb((0x1A), 0x390); /* Select RAM download/upload port */ + + for (i = 0; i < n; i++) + outb((trix_boot[i]), 0x391); + for (i = n; i < 10016; i++) /* Clear up to first 16 bytes of data RAM */ + outb((0x00), 0x391); + outb((0x00), base + 6); /* Reset */ + outb((0x50), 0x390); /* ?????? */ + +} + +static int trix_set_wss_port(struct address_info *hw_config) +{ + unsigned char addr_bits; + + if (trix_read(0x15) != 0x71) /* No ASIC signature */ + { + MDB(printk(KERN_ERR "No AudioTrix ASIC signature found\n")); + return 0; + } + + /* + * Reset some registers. + */ + + trix_write(0x13, 0); + trix_write(0x14, 0); + + /* + * Configure the ASIC to place the codec to the proper I/O location + */ + + switch (hw_config->io_base) + { + case 0x530: + addr_bits = 0; + break; + case 0x604: + addr_bits = 1; + break; + case 0xE80: + addr_bits = 2; + break; + case 0xF40: + addr_bits = 3; + break; + default: + return 0; + } + + trix_write(0x19, (trix_read(0x19) & 0x03) | addr_bits); + return 1; +} + +/* + * Probe and attach routines for the Windows Sound System mode of + * AudioTrix Pro + */ + +static int __init init_trix_wss(struct address_info *hw_config) +{ + static unsigned char dma_bits[4] = { + 1, 2, 0, 3 + }; + struct resource *ports; + int config_port = hw_config->io_base + 0; + int dma1 = hw_config->dma, dma2 = hw_config->dma2; + int old_num_mixers = num_mixers; + u8 config, bits; + int ret; + + switch(hw_config->irq) { + case 7: + bits = 8; + break; + case 9: + bits = 0x10; + break; + case 10: + bits = 0x18; + break; + case 11: + bits = 0x20; + break; + default: + printk(KERN_ERR "AudioTrix: Bad WSS IRQ %d\n", hw_config->irq); + return 0; + } + + switch (dma1) { + case 0: + case 1: + case 3: + break; + default: + printk(KERN_ERR "AudioTrix: Bad WSS DMA %d\n", dma1); + return 0; + } + + switch (dma2) { + case -1: + case 0: + case 1: + case 3: + break; + default: + printk(KERN_ERR "AudioTrix: Bad capture DMA %d\n", dma2); + return 0; + } + + /* + * Check if the IO port returns valid signature. The original MS Sound + * system returns 0x04 while some cards (AudioTrix Pro for example) + * return 0x00. + */ + ports = request_region(hw_config->io_base + 4, 4, "ad1848"); + if (!ports) { + printk(KERN_ERR "AudioTrix: MSS I/O port conflict (%x)\n", hw_config->io_base); + return 0; + } + + if (!request_region(hw_config->io_base, 4, "MSS config")) { + printk(KERN_ERR "AudioTrix: MSS I/O port conflict (%x)\n", hw_config->io_base); + release_region(hw_config->io_base + 4, 4); + return 0; + } + + if (!trix_set_wss_port(hw_config)) + goto fail; + + config = inb(hw_config->io_base + 3); + + if ((config & 0x3f) != 0x00) + { + MDB(printk(KERN_ERR "No MSS signature detected on port 0x%x\n", hw_config->io_base)); + goto fail; + } + + /* + * Check that DMA0 is not in use with a 8 bit board. + */ + + if (dma1 == 0 && config & 0x80) + { + printk(KERN_ERR "AudioTrix: Can't use DMA0 with a 8 bit card slot\n"); + goto fail; + } + if (hw_config->irq > 9 && config & 0x80) + { + printk(KERN_ERR "AudioTrix: Can't use IRQ%d with a 8 bit card slot\n", hw_config->irq); + goto fail; + } + + ret = ad1848_detect(ports, NULL, hw_config->osp); + if (!ret) + goto fail; + + if (joystick==1) + trix_write(0x15, 0x80); + + /* + * Set the IRQ and DMA addresses. + */ + + outb((bits | 0x40), config_port); + + if (dma2 == -1 || dma2 == dma1) + { + bits |= dma_bits[dma1]; + dma2 = dma1; + } + else + { + unsigned char tmp; + + tmp = trix_read(0x13) & ~30; + trix_write(0x13, tmp | 0x80 | (dma1 << 4)); + + tmp = trix_read(0x14) & ~30; + trix_write(0x14, tmp | 0x80 | (dma2 << 4)); + } + + outb((bits), config_port); /* Write IRQ+DMA setup */ + + hw_config->slots[0] = ad1848_init("AudioTrix Pro", ports, + hw_config->irq, + dma1, + dma2, + 0, + hw_config->osp, + THIS_MODULE); + + if (num_mixers > old_num_mixers) /* Mixer got installed */ + { + AD1848_REROUTE(SOUND_MIXER_LINE1, SOUND_MIXER_LINE); /* Line in */ + AD1848_REROUTE(SOUND_MIXER_LINE2, SOUND_MIXER_CD); + AD1848_REROUTE(SOUND_MIXER_LINE3, SOUND_MIXER_SYNTH); /* OPL4 */ + AD1848_REROUTE(SOUND_MIXER_SPEAKER, SOUND_MIXER_ALTPCM); /* SB */ + } + return 1; + +fail: + release_region(hw_config->io_base, 4); + release_region(hw_config->io_base + 4, 4); + return 0; +} + +static int __init probe_trix_sb(struct address_info *hw_config) +{ + + int tmp; + unsigned char conf; + extern int sb_be_quiet; + int old_quiet; + static signed char irq_translate[] = { + -1, -1, -1, 0, 1, 2, -1, 3 + }; + + if (trix_boot_len == 0) + return 0; /* No boot code -> no fun */ + + if ((hw_config->io_base & 0xffffff8f) != 0x200) + return 0; + + tmp = hw_config->irq; + if (tmp > 7) + return 0; + if (irq_translate[tmp] == -1) + return 0; + + tmp = hw_config->dma; + if (tmp != 1 && tmp != 3) + return 0; + + if (!request_region(hw_config->io_base, 16, "soundblaster")) { + printk(KERN_ERR "AudioTrix: SB I/O port conflict (%x)\n", hw_config->io_base); + return 0; + } + + conf = 0x84; /* DMA and IRQ enable */ + conf |= hw_config->io_base & 0x70; /* I/O address bits */ + conf |= irq_translate[hw_config->irq]; + if (hw_config->dma == 3) + conf |= 0x08; + trix_write(0x1b, conf); + + download_boot(hw_config->io_base); + + hw_config->name = "AudioTrix SB"; + if (!sb_dsp_detect(hw_config, 0, 0, NULL)) { + release_region(hw_config->io_base, 16); + return 0; + } + + hw_config->driver_use_1 = SB_NO_MIDI | SB_NO_MIXER | SB_NO_RECORDING; + + /* Prevent false alarms */ + old_quiet = sb_be_quiet; + sb_be_quiet = 1; + + sb_dsp_init(hw_config, THIS_MODULE); + + sb_be_quiet = old_quiet; + return 1; +} + +static int __init probe_trix_mpu(struct address_info *hw_config) +{ + unsigned char conf; + static int irq_bits[] = { + -1, -1, -1, 1, 2, 3, -1, 4, -1, 5 + }; + + if (hw_config->irq > 9) + { + printk(KERN_ERR "AudioTrix: Bad MPU IRQ %d\n", hw_config->irq); + return 0; + } + if (irq_bits[hw_config->irq] == -1) + { + printk(KERN_ERR "AudioTrix: Bad MPU IRQ %d\n", hw_config->irq); + return 0; + } + switch (hw_config->io_base) + { + case 0x330: + conf = 0x00; + break; + case 0x370: + conf = 0x04; + break; + case 0x3b0: + conf = 0x08; + break; + case 0x3f0: + conf = 0x0c; + break; + default: + return 0; /* Invalid port */ + } + + conf |= irq_bits[hw_config->irq] << 4; + trix_write(0x19, (trix_read(0x19) & 0x83) | conf); + hw_config->name = "AudioTrix Pro"; + return probe_uart401(hw_config, THIS_MODULE); +} + +static void __exit unload_trix_wss(struct address_info *hw_config) +{ + int dma2 = hw_config->dma2; + + if (dma2 == -1) + dma2 = hw_config->dma; + + release_region(0x390, 2); + release_region(hw_config->io_base, 4); + + ad1848_unload(hw_config->io_base + 4, + hw_config->irq, + hw_config->dma, + dma2, + 0); + sound_unload_audiodev(hw_config->slots[0]); +} + +static inline void __exit unload_trix_mpu(struct address_info *hw_config) +{ + unload_uart401(hw_config); +} + +static inline void __exit unload_trix_sb(struct address_info *hw_config) +{ + sb_dsp_unload(hw_config, mpu); +} + +static struct address_info cfg; +static struct address_info cfg2; +static struct address_info cfg_mpu; + +static int sb; +static int fw_load; + +static int __initdata io = -1; +static int __initdata irq = -1; +static int __initdata dma = -1; +static int __initdata dma2 = -1; /* Set this for modules that need it */ +static int __initdata sb_io = -1; +static int __initdata sb_dma = -1; +static int __initdata sb_irq = -1; +static int __initdata mpu_io = -1; +static int __initdata mpu_irq = -1; + +module_param(io, int, 0); +module_param(irq, int, 0); +module_param(dma, int, 0); +module_param(dma2, int, 0); +module_param(sb_io, int, 0); +module_param(sb_dma, int, 0); +module_param(sb_irq, int, 0); +module_param(mpu_io, int, 0); +module_param(mpu_irq, int, 0); +module_param(joystick, bool, 0); + +static int __init init_trix(void) +{ + printk(KERN_INFO "MediaTrix audio driver Copyright (C) by Hannu Savolainen 1993-1996\n"); + + cfg.io_base = io; + cfg.irq = irq; + cfg.dma = dma; + cfg.dma2 = dma2; + + cfg2.io_base = sb_io; + cfg2.irq = sb_irq; + cfg2.dma = sb_dma; + + cfg_mpu.io_base = mpu_io; + cfg_mpu.irq = mpu_irq; + + if (cfg.io_base == -1 || cfg.dma == -1 || cfg.irq == -1) { + printk(KERN_INFO "I/O, IRQ, DMA and type are mandatory\n"); + return -EINVAL; + } + + if (cfg2.io_base != -1 && (cfg2.irq == -1 || cfg2.dma == -1)) { + printk(KERN_INFO "CONFIG_SB_IRQ and CONFIG_SB_DMA must be specified if SB_IO is set.\n"); + return -EINVAL; + } + if (cfg_mpu.io_base != -1 && cfg_mpu.irq == -1) { + printk(KERN_INFO "CONFIG_MPU_IRQ must be specified if MPU_IO is set.\n"); + return -EINVAL; + } + if (!trix_boot) + { + fw_load = 1; + trix_boot_len = mod_firmware_load("/etc/sound/trxpro.bin", + (char **) &trix_boot); + } + + if (!request_region(0x390, 2, "AudioTrix")) { + printk(KERN_ERR "AudioTrix: Config port I/O conflict\n"); + return -ENODEV; + } + + if (!init_trix_wss(&cfg)) { + release_region(0x390, 2); + return -ENODEV; + } + + /* + * We must attach in the right order to get the firmware + * loaded up in time. + */ + + if (cfg2.io_base != -1) { + sb = probe_trix_sb(&cfg2); + } + + if (cfg_mpu.io_base != -1) + mpu = probe_trix_mpu(&cfg_mpu); + + return 0; +} + +static void __exit cleanup_trix(void) +{ + if (fw_load && trix_boot) + vfree(trix_boot); + if (sb) + unload_trix_sb(&cfg2); + if (mpu) + unload_trix_mpu(&cfg_mpu); + unload_trix_wss(&cfg); +} + +module_init(init_trix); +module_exit(cleanup_trix); + +#ifndef MODULE +static int __init setup_trix (char *str) +{ + /* io, irq, dma, dma2, sb_io, sb_irq, sb_dma, mpu_io, mpu_irq */ + int ints[9]; + + str = get_options(str, ARRAY_SIZE(ints), ints); + + io = ints[1]; + irq = ints[2]; + dma = ints[3]; + dma2 = ints[4]; + sb_io = ints[5]; + sb_irq = ints[6]; + sb_dma = ints[6]; + mpu_io = ints[7]; + mpu_irq = ints[8]; + + return 1; +} + +__setup("trix=", setup_trix); +#endif +MODULE_LICENSE("GPL"); diff --git a/sound/oss/tuning.h b/sound/oss/tuning.h new file mode 100644 index 00000000..a73e3dd3 --- /dev/null +++ b/sound/oss/tuning.h @@ -0,0 +1,23 @@ +static unsigned short semitone_tuning[24] = +{ +/* 0 */ 10000, 10595, 11225, 11892, 12599, 13348, 14142, 14983, +/* 8 */ 15874, 16818, 17818, 18877, 20000, 21189, 22449, 23784, +/* 16 */ 25198, 26697, 28284, 29966, 31748, 33636, 35636, 37755 +}; + +static unsigned short cent_tuning[100] = +{ +/* 0 */ 10000, 10006, 10012, 10017, 10023, 10029, 10035, 10041, +/* 8 */ 10046, 10052, 10058, 10064, 10070, 10075, 10081, 10087, +/* 16 */ 10093, 10099, 10105, 10110, 10116, 10122, 10128, 10134, +/* 24 */ 10140, 10145, 10151, 10157, 10163, 10169, 10175, 10181, +/* 32 */ 10187, 10192, 10198, 10204, 10210, 10216, 10222, 10228, +/* 40 */ 10234, 10240, 10246, 10251, 10257, 10263, 10269, 10275, +/* 48 */ 10281, 10287, 10293, 10299, 10305, 10311, 10317, 10323, +/* 56 */ 10329, 10335, 10341, 10347, 10353, 10359, 10365, 10371, +/* 64 */ 10377, 10383, 10389, 10395, 10401, 10407, 10413, 10419, +/* 72 */ 10425, 10431, 10437, 10443, 10449, 10455, 10461, 10467, +/* 80 */ 10473, 10479, 10485, 10491, 10497, 10503, 10509, 10515, +/* 88 */ 10521, 10528, 10534, 10540, 10546, 10552, 10558, 10564, +/* 96 */ 10570, 10576, 10582, 10589 +}; diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c new file mode 100644 index 00000000..8e514a67 --- /dev/null +++ b/sound/oss/uart401.c @@ -0,0 +1,482 @@ +/* + * sound/oss/uart401.c + * + * MPU-401 UART driver (formerly uart401_midi.c) + * + * + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * + * Changes: + * Alan Cox Reformatted, removed sound_mem usage, use normal Linux + * interrupt allocation. Protect against bogus unload + * Fixed to allow IRQ > 15 + * Christoph Hellwig Adapted to module_init/module_exit + * Arnaldo C. de Melo got rid of check_region + * + * Status: + * Untested + */ + +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/spinlock.h> +#include "sound_config.h" + +#include "mpu401.h" + +typedef struct uart401_devc +{ + int base; + int irq; + int *osp; + void (*midi_input_intr) (int dev, unsigned char data); + int opened, disabled; + volatile unsigned char input_byte; + int my_dev; + int share_irq; + spinlock_t lock; +} +uart401_devc; + +#define DATAPORT (devc->base) +#define COMDPORT (devc->base+1) +#define STATPORT (devc->base+1) + +static int uart401_status(uart401_devc * devc) +{ + return inb(STATPORT); +} + +#define input_avail(devc) (!(uart401_status(devc)&INPUT_AVAIL)) +#define output_ready(devc) (!(uart401_status(devc)&OUTPUT_READY)) + +static void uart401_cmd(uart401_devc * devc, unsigned char cmd) +{ + outb((cmd), COMDPORT); +} + +static int uart401_read(uart401_devc * devc) +{ + return inb(DATAPORT); +} + +static void uart401_write(uart401_devc * devc, unsigned char byte) +{ + outb((byte), DATAPORT); +} + +#define OUTPUT_READY 0x40 +#define INPUT_AVAIL 0x80 +#define MPU_ACK 0xFE +#define MPU_RESET 0xFF +#define UART_MODE_ON 0x3F + +static int reset_uart401(uart401_devc * devc); +static void enter_uart_mode(uart401_devc * devc); + +static void uart401_input_loop(uart401_devc * devc) +{ + int work_limit=30000; + + while (input_avail(devc) && --work_limit) + { + unsigned char c = uart401_read(devc); + + if (c == MPU_ACK) + devc->input_byte = c; + else if (devc->opened & OPEN_READ && devc->midi_input_intr) + devc->midi_input_intr(devc->my_dev, c); + } + if(work_limit==0) + printk(KERN_WARNING "Too much work in interrupt on uart401 (0x%X). UART jabbering ??\n", devc->base); +} + +irqreturn_t uart401intr(int irq, void *dev_id) +{ + uart401_devc *devc = dev_id; + + if (devc == NULL) + { + printk(KERN_ERR "uart401: bad devc\n"); + return IRQ_NONE; + } + + if (input_avail(devc)) + uart401_input_loop(devc); + return IRQ_HANDLED; +} + +static int +uart401_open(int dev, int mode, + void (*input) (int dev, unsigned char data), + void (*output) (int dev) +) +{ + uart401_devc *devc = (uart401_devc *) midi_devs[dev]->devc; + + if (devc->opened) + return -EBUSY; + + /* Flush the UART */ + + while (input_avail(devc)) + uart401_read(devc); + + devc->midi_input_intr = input; + devc->opened = mode; + enter_uart_mode(devc); + devc->disabled = 0; + + return 0; +} + +static void uart401_close(int dev) +{ + uart401_devc *devc = (uart401_devc *) midi_devs[dev]->devc; + + reset_uart401(devc); + devc->opened = 0; +} + +static int uart401_out(int dev, unsigned char midi_byte) +{ + int timeout; + unsigned long flags; + uart401_devc *devc = (uart401_devc *) midi_devs[dev]->devc; + + if (devc->disabled) + return 1; + /* + * Test for input since pending input seems to block the output. + */ + + spin_lock_irqsave(&devc->lock,flags); + if (input_avail(devc)) + uart401_input_loop(devc); + + spin_unlock_irqrestore(&devc->lock,flags); + + /* + * Sometimes it takes about 13000 loops before the output becomes ready + * (After reset). Normally it takes just about 10 loops. + */ + + for (timeout = 30000; timeout > 0 && !output_ready(devc); timeout--); + + if (!output_ready(devc)) + { + printk(KERN_WARNING "uart401: Timeout - Device not responding\n"); + devc->disabled = 1; + reset_uart401(devc); + enter_uart_mode(devc); + return 1; + } + uart401_write(devc, midi_byte); + return 1; +} + +static inline int uart401_start_read(int dev) +{ + return 0; +} + +static inline int uart401_end_read(int dev) +{ + return 0; +} + +static inline void uart401_kick(int dev) +{ +} + +static inline int uart401_buffer_status(int dev) +{ + return 0; +} + +#define MIDI_SYNTH_NAME "MPU-401 UART" +#define MIDI_SYNTH_CAPS SYNTH_CAP_INPUT +#include "midi_synth.h" + +static const struct midi_operations uart401_operations = +{ + .owner = THIS_MODULE, + .info = {"MPU-401 (UART) MIDI", 0, 0, SNDCARD_MPU401}, + .converter = &std_midi_synth, + .in_info = {0}, + .open = uart401_open, + .close = uart401_close, + .outputc = uart401_out, + .start_read = uart401_start_read, + .end_read = uart401_end_read, + .kick = uart401_kick, + .buffer_status = uart401_buffer_status, +}; + +static void enter_uart_mode(uart401_devc * devc) +{ + int ok, timeout; + unsigned long flags; + + spin_lock_irqsave(&devc->lock,flags); + for (timeout = 30000; timeout > 0 && !output_ready(devc); timeout--); + + devc->input_byte = 0; + uart401_cmd(devc, UART_MODE_ON); + + ok = 0; + for (timeout = 50000; timeout > 0 && !ok; timeout--) + if (devc->input_byte == MPU_ACK) + ok = 1; + else if (input_avail(devc)) + if (uart401_read(devc) == MPU_ACK) + ok = 1; + + spin_unlock_irqrestore(&devc->lock,flags); +} + +static int reset_uart401(uart401_devc * devc) +{ + int ok, timeout, n; + + /* + * Send the RESET command. Try again if no success at the first time. + */ + + ok = 0; + + for (n = 0; n < 2 && !ok; n++) + { + for (timeout = 30000; timeout > 0 && !output_ready(devc); timeout--); + devc->input_byte = 0; + uart401_cmd(devc, MPU_RESET); + + /* + * Wait at least 25 msec. This method is not accurate so let's make the + * loop bit longer. Cannot sleep since this is called during boot. + */ + + for (timeout = 50000; timeout > 0 && !ok; timeout--) + { + if (devc->input_byte == MPU_ACK) /* Interrupt */ + ok = 1; + else if (input_avail(devc)) + { + if (uart401_read(devc) == MPU_ACK) + ok = 1; + } + } + } + + + if (ok) + { + DEB(printk("Reset UART401 OK\n")); + } + else + DDB(printk("Reset UART401 failed - No hardware detected.\n")); + + if (ok) + uart401_input_loop(devc); /* + * Flush input before enabling interrupts + */ + + return ok; +} + +int probe_uart401(struct address_info *hw_config, struct module *owner) +{ + uart401_devc *devc; + char *name = "MPU-401 (UART) MIDI"; + int ok = 0; + unsigned long flags; + + DDB(printk("Entered probe_uart401()\n")); + + /* Default to "not found" */ + hw_config->slots[4] = -1; + + if (!request_region(hw_config->io_base, 4, "MPU-401 UART")) { + printk(KERN_INFO "uart401: could not request_region(%d, 4)\n", hw_config->io_base); + return 0; + } + + devc = kmalloc(sizeof(uart401_devc), GFP_KERNEL); + if (!devc) { + printk(KERN_WARNING "uart401: Can't allocate memory\n"); + goto cleanup_region; + } + + devc->base = hw_config->io_base; + devc->irq = hw_config->irq; + devc->osp = hw_config->osp; + devc->midi_input_intr = NULL; + devc->opened = 0; + devc->input_byte = 0; + devc->my_dev = 0; + devc->share_irq = 0; + spin_lock_init(&devc->lock); + + spin_lock_irqsave(&devc->lock,flags); + ok = reset_uart401(devc); + spin_unlock_irqrestore(&devc->lock,flags); + + if (!ok) + goto cleanup_devc; + + if (hw_config->name) + name = hw_config->name; + + if (devc->irq < 0) { + devc->share_irq = 1; + devc->irq *= -1; + } else + devc->share_irq = 0; + + if (!devc->share_irq) + if (request_irq(devc->irq, uart401intr, 0, "MPU-401 UART", devc) < 0) { + printk(KERN_WARNING "uart401: Failed to allocate IRQ%d\n", devc->irq); + devc->share_irq = 1; + } + devc->my_dev = sound_alloc_mididev(); + enter_uart_mode(devc); + + if (devc->my_dev == -1) { + printk(KERN_INFO "uart401: Too many midi devices detected\n"); + goto cleanup_irq; + } + conf_printf(name, hw_config); + midi_devs[devc->my_dev] = kmalloc(sizeof(struct midi_operations), GFP_KERNEL); + if (!midi_devs[devc->my_dev]) { + printk(KERN_ERR "uart401: Failed to allocate memory\n"); + goto cleanup_unload_mididev; + } + memcpy(midi_devs[devc->my_dev], &uart401_operations, sizeof(struct midi_operations)); + + if (owner) + midi_devs[devc->my_dev]->owner = owner; + + midi_devs[devc->my_dev]->devc = devc; + midi_devs[devc->my_dev]->converter = kmalloc(sizeof(struct synth_operations), GFP_KERNEL); + if (!midi_devs[devc->my_dev]->converter) { + printk(KERN_WARNING "uart401: Failed to allocate memory\n"); + goto cleanup_midi_devs; + } + memcpy(midi_devs[devc->my_dev]->converter, &std_midi_synth, sizeof(struct synth_operations)); + strcpy(midi_devs[devc->my_dev]->info.name, name); + midi_devs[devc->my_dev]->converter->id = "UART401"; + midi_devs[devc->my_dev]->converter->midi_dev = devc->my_dev; + + if (owner) + midi_devs[devc->my_dev]->converter->owner = owner; + + hw_config->slots[4] = devc->my_dev; + sequencer_init(); + devc->opened = 0; + return 1; +cleanup_midi_devs: + kfree(midi_devs[devc->my_dev]); +cleanup_unload_mididev: + sound_unload_mididev(devc->my_dev); +cleanup_irq: + if (!devc->share_irq) + free_irq(devc->irq, devc); +cleanup_devc: + kfree(devc); +cleanup_region: + release_region(hw_config->io_base, 4); + return 0; +} + +void unload_uart401(struct address_info *hw_config) +{ + uart401_devc *devc; + int n=hw_config->slots[4]; + + /* Not set up */ + if(n==-1 || midi_devs[n]==NULL) + return; + + /* Not allocated (erm ??) */ + + devc = midi_devs[hw_config->slots[4]]->devc; + if (devc == NULL) + return; + + reset_uart401(devc); + release_region(hw_config->io_base, 4); + + if (!devc->share_irq) + free_irq(devc->irq, devc); + if (devc) + { + kfree(midi_devs[devc->my_dev]->converter); + kfree(midi_devs[devc->my_dev]); + kfree(devc); + devc = NULL; + } + /* This kills midi_devs[x] */ + sound_unload_mididev(hw_config->slots[4]); +} + +EXPORT_SYMBOL(probe_uart401); +EXPORT_SYMBOL(unload_uart401); +EXPORT_SYMBOL(uart401intr); + +static struct address_info cfg_mpu; + +static int io = -1; +static int irq = -1; + +module_param(io, int, 0444); +module_param(irq, int, 0444); + + +static int __init init_uart401(void) +{ + cfg_mpu.irq = irq; + cfg_mpu.io_base = io; + + /* Can be loaded either for module use or to provide functions + to others */ + if (cfg_mpu.io_base != -1 && cfg_mpu.irq != -1) { + printk(KERN_INFO "MPU-401 UART driver Copyright (C) Hannu Savolainen 1993-1997"); + if (!probe_uart401(&cfg_mpu, THIS_MODULE)) + return -ENODEV; + } + + return 0; +} + +static void __exit cleanup_uart401(void) +{ + if (cfg_mpu.io_base != -1 && cfg_mpu.irq != -1) + unload_uart401(&cfg_mpu); +} + +module_init(init_uart401); +module_exit(cleanup_uart401); + +#ifndef MODULE +static int __init setup_uart401(char *str) +{ + /* io, irq */ + int ints[3]; + + str = get_options(str, ARRAY_SIZE(ints), ints); + + io = ints[1]; + irq = ints[2]; + + return 1; +} + +__setup("uart401=", setup_uart401); +#endif +MODULE_LICENSE("GPL"); diff --git a/sound/oss/uart6850.c b/sound/oss/uart6850.c new file mode 100644 index 00000000..f3f914aa --- /dev/null +++ b/sound/oss/uart6850.c @@ -0,0 +1,361 @@ +/* + * sound/oss/uart6850.c + * + * + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * Extended by Alan Cox for Red Hat Software. Now a loadable MIDI driver. + * 28/4/97 - (C) Copyright Alan Cox. Released under the GPL version 2. + * + * Alan Cox: Updated for new modular code. Removed snd_* irq handling. Now + * uses native linux resources + * Christoph Hellwig: Adapted to module_init/module_exit + * Jeff Garzik: Made it work again, in theory + * FIXME: If the request_irq() succeeds, the probe succeeds. Ug. + * + * Status: Testing required (no shit -jgarzik) + * + * + */ + +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/spinlock.h> +/* Mon Nov 22 22:38:35 MET 1993 marco@driq.home.usn.nl: + * added 6850 support, used with COVOX SoundMaster II and custom cards. + */ + +#include "sound_config.h" + +static int uart6850_base = 0x330; + +static int *uart6850_osp; + +#define DATAPORT (uart6850_base) +#define COMDPORT (uart6850_base+1) +#define STATPORT (uart6850_base+1) + +static int uart6850_status(void) +{ + return inb(STATPORT); +} + +#define input_avail() (uart6850_status()&INPUT_AVAIL) +#define output_ready() (uart6850_status()&OUTPUT_READY) + +static void uart6850_cmd(unsigned char cmd) +{ + outb(cmd, COMDPORT); +} + +static int uart6850_read(void) +{ + return inb(DATAPORT); +} + +static void uart6850_write(unsigned char byte) +{ + outb(byte, DATAPORT); +} + +#define OUTPUT_READY 0x02 /* Mask for data ready Bit */ +#define INPUT_AVAIL 0x01 /* Mask for Data Send Ready Bit */ + +#define UART_RESET 0x95 +#define UART_MODE_ON 0x03 + +static int uart6850_opened; +static int uart6850_irq; +static int uart6850_detected; +static int my_dev; +static DEFINE_SPINLOCK(lock); + +static void (*midi_input_intr) (int dev, unsigned char data); +static void poll_uart6850(unsigned long dummy); + + +static DEFINE_TIMER(uart6850_timer, poll_uart6850, 0, 0); + +static void uart6850_input_loop(void) +{ + int count = 10; + + while (count) + { + /* + * Not timed out + */ + if (input_avail()) + { + unsigned char c = uart6850_read(); + count = 100; + if (uart6850_opened & OPEN_READ) + midi_input_intr(my_dev, c); + } + else + { + while (!input_avail() && count) + count--; + } + } +} + +static irqreturn_t m6850intr(int irq, void *dev_id) +{ + if (input_avail()) + uart6850_input_loop(); + return IRQ_HANDLED; +} + +/* + * It looks like there is no input interrupts in the UART mode. Let's try + * polling. + */ + +static void poll_uart6850(unsigned long dummy) +{ + unsigned long flags; + + if (!(uart6850_opened & OPEN_READ)) + return; /* Device has been closed */ + + spin_lock_irqsave(&lock,flags); + if (input_avail()) + uart6850_input_loop(); + + uart6850_timer.expires = 1 + jiffies; + add_timer(&uart6850_timer); + + /* + * Come back later + */ + + spin_unlock_irqrestore(&lock,flags); +} + +static int uart6850_open(int dev, int mode, + void (*input) (int dev, unsigned char data), + void (*output) (int dev) +) +{ + if (uart6850_opened) + { +/* printk("Midi6850: Midi busy\n");*/ + return -EBUSY; + }; + + uart6850_cmd(UART_RESET); + uart6850_input_loop(); + midi_input_intr = input; + uart6850_opened = mode; + poll_uart6850(0); /* + * Enable input polling + */ + + return 0; +} + +static void uart6850_close(int dev) +{ + uart6850_cmd(UART_MODE_ON); + del_timer(&uart6850_timer); + uart6850_opened = 0; +} + +static int uart6850_out(int dev, unsigned char midi_byte) +{ + int timeout; + unsigned long flags; + + /* + * Test for input since pending input seems to block the output. + */ + + spin_lock_irqsave(&lock,flags); + + if (input_avail()) + uart6850_input_loop(); + + spin_unlock_irqrestore(&lock,flags); + + /* + * Sometimes it takes about 13000 loops before the output becomes ready + * (After reset). Normally it takes just about 10 loops. + */ + + for (timeout = 30000; timeout > 0 && !output_ready(); timeout--); /* + * Wait + */ + if (!output_ready()) + { + printk(KERN_WARNING "Midi6850: Timeout\n"); + return 0; + } + uart6850_write(midi_byte); + return 1; +} + +static inline int uart6850_command(int dev, unsigned char *midi_byte) +{ + return 1; +} + +static inline int uart6850_start_read(int dev) +{ + return 0; +} + +static inline int uart6850_end_read(int dev) +{ + return 0; +} + +static inline void uart6850_kick(int dev) +{ +} + +static inline int uart6850_buffer_status(int dev) +{ + return 0; /* + * No data in buffers + */ +} + +#define MIDI_SYNTH_NAME "6850 UART Midi" +#define MIDI_SYNTH_CAPS SYNTH_CAP_INPUT +#include "midi_synth.h" + +static struct midi_operations uart6850_operations = +{ + .owner = THIS_MODULE, + .info = {"6850 UART", 0, 0, SNDCARD_UART6850}, + .converter = &std_midi_synth, + .in_info = {0}, + .open = uart6850_open, + .close = uart6850_close, + .outputc = uart6850_out, + .start_read = uart6850_start_read, + .end_read = uart6850_end_read, + .kick = uart6850_kick, + .command = uart6850_command, + .buffer_status = uart6850_buffer_status +}; + + +static void __init attach_uart6850(struct address_info *hw_config) +{ + int ok, timeout; + unsigned long flags; + + if (!uart6850_detected) + return; + + if ((my_dev = sound_alloc_mididev()) == -1) + { + printk(KERN_INFO "uart6850: Too many midi devices detected\n"); + return; + } + uart6850_base = hw_config->io_base; + uart6850_osp = hw_config->osp; + uart6850_irq = hw_config->irq; + + spin_lock_irqsave(&lock,flags); + + for (timeout = 30000; timeout > 0 && !output_ready(); timeout--); /* + * Wait + */ + uart6850_cmd(UART_MODE_ON); + ok = 1; + spin_unlock_irqrestore(&lock,flags); + + conf_printf("6850 Midi Interface", hw_config); + + std_midi_synth.midi_dev = my_dev; + hw_config->slots[4] = my_dev; + midi_devs[my_dev] = &uart6850_operations; + sequencer_init(); +} + +static inline int reset_uart6850(void) +{ + uart6850_read(); + return 1; /* + * OK + */ +} + +static int __init probe_uart6850(struct address_info *hw_config) +{ + int ok; + + uart6850_osp = hw_config->osp; + uart6850_base = hw_config->io_base; + uart6850_irq = hw_config->irq; + + if (request_irq(uart6850_irq, m6850intr, 0, "MIDI6850", NULL) < 0) + return 0; + + ok = reset_uart6850(); + uart6850_detected = ok; + return ok; +} + +static void __exit unload_uart6850(struct address_info *hw_config) +{ + free_irq(hw_config->irq, NULL); + sound_unload_mididev(hw_config->slots[4]); +} + +static struct address_info cfg_mpu; + +static int __initdata io = -1; +static int __initdata irq = -1; + +module_param(io, int, 0); +module_param(irq, int, 0); + +static int __init init_uart6850(void) +{ + cfg_mpu.io_base = io; + cfg_mpu.irq = irq; + + if (cfg_mpu.io_base == -1 || cfg_mpu.irq == -1) { + printk(KERN_INFO "uart6850: irq and io must be set.\n"); + return -EINVAL; + } + + if (probe_uart6850(&cfg_mpu)) + return -ENODEV; + attach_uart6850(&cfg_mpu); + + return 0; +} + +static void __exit cleanup_uart6850(void) +{ + unload_uart6850(&cfg_mpu); +} + +module_init(init_uart6850); +module_exit(cleanup_uart6850); + +#ifndef MODULE +static int __init setup_uart6850(char *str) +{ + /* io, irq */ + int ints[3]; + + str = get_options(str, ARRAY_SIZE(ints), ints); + + io = ints[1]; + irq = ints[2]; + + return 1; +} +__setup("uart6850=", setup_uart6850); +#endif +MODULE_LICENSE("GPL"); diff --git a/sound/oss/ulaw.h b/sound/oss/ulaw.h new file mode 100644 index 00000000..0ff8c0a3 --- /dev/null +++ b/sound/oss/ulaw.h @@ -0,0 +1,69 @@ +static unsigned char ulaw_dsp[] = { + 3, 7, 11, 15, 19, 23, 27, 31, + 35, 39, 43, 47, 51, 55, 59, 63, + 66, 68, 70, 72, 74, 76, 78, 80, + 82, 84, 86, 88, 90, 92, 94, 96, + 98, 99, 100, 101, 102, 103, 104, 105, + 106, 107, 108, 109, 110, 111, 112, 113, + 113, 114, 114, 115, 115, 116, 116, 117, + 117, 118, 118, 119, 119, 120, 120, 121, + 121, 121, 122, 122, 122, 122, 123, 123, + 123, 123, 124, 124, 124, 124, 125, 125, + 125, 125, 125, 125, 126, 126, 126, 126, + 126, 126, 126, 126, 127, 127, 127, 127, + 127, 127, 127, 127, 127, 127, 127, 127, + 128, 128, 128, 128, 128, 128, 128, 128, + 128, 128, 128, 128, 128, 128, 128, 128, + 128, 128, 128, 128, 128, 128, 128, 128, + 253, 249, 245, 241, 237, 233, 229, 225, + 221, 217, 213, 209, 205, 201, 197, 193, + 190, 188, 186, 184, 182, 180, 178, 176, + 174, 172, 170, 168, 166, 164, 162, 160, + 158, 157, 156, 155, 154, 153, 152, 151, + 150, 149, 148, 147, 146, 145, 144, 143, + 143, 142, 142, 141, 141, 140, 140, 139, + 139, 138, 138, 137, 137, 136, 136, 135, + 135, 135, 134, 134, 134, 134, 133, 133, + 133, 133, 132, 132, 132, 132, 131, 131, + 131, 131, 131, 131, 130, 130, 130, 130, + 130, 130, 130, 130, 129, 129, 129, 129, + 129, 129, 129, 129, 129, 129, 129, 129, + 128, 128, 128, 128, 128, 128, 128, 128, + 128, 128, 128, 128, 128, 128, 128, 128, + 128, 128, 128, 128, 128, 128, 128, 128, +}; + +static unsigned char dsp_ulaw[] = { + 0, 0, 0, 0, 0, 1, 1, 1, + 1, 2, 2, 2, 2, 3, 3, 3, + 3, 4, 4, 4, 4, 5, 5, 5, + 5, 6, 6, 6, 6, 7, 7, 7, + 7, 8, 8, 8, 8, 9, 9, 9, + 9, 10, 10, 10, 10, 11, 11, 11, + 11, 12, 12, 12, 12, 13, 13, 13, + 13, 14, 14, 14, 14, 15, 15, 15, + 15, 16, 16, 17, 17, 18, 18, 19, + 19, 20, 20, 21, 21, 22, 22, 23, + 23, 24, 24, 25, 25, 26, 26, 27, + 27, 28, 28, 29, 29, 30, 30, 31, + 31, 32, 33, 34, 35, 36, 37, 38, + 39, 40, 41, 42, 43, 44, 45, 46, + 47, 49, 51, 53, 55, 57, 59, 61, + 63, 66, 70, 74, 78, 84, 92, 104, + 254, 231, 219, 211, 205, 201, 197, 193, + 190, 188, 186, 184, 182, 180, 178, 176, + 175, 174, 173, 172, 171, 170, 169, 168, + 167, 166, 165, 164, 163, 162, 161, 160, + 159, 159, 158, 158, 157, 157, 156, 156, + 155, 155, 154, 154, 153, 153, 152, 152, + 151, 151, 150, 150, 149, 149, 148, 148, + 147, 147, 146, 146, 145, 145, 144, 144, + 143, 143, 143, 143, 142, 142, 142, 142, + 141, 141, 141, 141, 140, 140, 140, 140, + 139, 139, 139, 139, 138, 138, 138, 138, + 137, 137, 137, 137, 136, 136, 136, 136, + 135, 135, 135, 135, 134, 134, 134, 134, + 133, 133, 133, 133, 132, 132, 132, 132, + 131, 131, 131, 131, 130, 130, 130, 130, + 129, 129, 129, 129, 128, 128, 128, 128, +}; diff --git a/sound/oss/v_midi.c b/sound/oss/v_midi.c new file mode 100644 index 00000000..f0b4151d --- /dev/null +++ b/sound/oss/v_midi.c @@ -0,0 +1,290 @@ +/* + * sound/oss/v_midi.c + * + * The low level driver for the Sound Blaster DS chips. + * + * + * Copyright (C) by Hannu Savolainen 1993-1996 + * + * USS/Lite for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * ?? + * + * Changes + * Alan Cox Modularisation, changed memory allocations + * Christoph Hellwig Adapted to module_init/module_exit + * + * Status + * Untested + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/spinlock.h> +#include "sound_config.h" + +#include "v_midi.h" + +static vmidi_devc *v_devc[2] = { NULL, NULL}; +static int midi1,midi2; +static void *midi_mem = NULL; + +/* + * The DSP channel can be used either for input or output. Variable + * 'sb_irq_mode' will be set when the program calls read or write first time + * after open. Current version doesn't support mode changes without closing + * and reopening the device. Support for this feature may be implemented in a + * future version of this driver. + */ + + +static int v_midi_open (int dev, int mode, + void (*input) (int dev, unsigned char data), + void (*output) (int dev) +) +{ + vmidi_devc *devc = midi_devs[dev]->devc; + unsigned long flags; + + if (devc == NULL) + return -(ENXIO); + + spin_lock_irqsave(&devc->lock,flags); + if (devc->opened) + { + spin_unlock_irqrestore(&devc->lock,flags); + return -(EBUSY); + } + devc->opened = 1; + spin_unlock_irqrestore(&devc->lock,flags); + + devc->intr_active = 1; + + if (mode & OPEN_READ) + { + devc->input_opened = 1; + devc->midi_input_intr = input; + } + + return 0; +} + +static void v_midi_close (int dev) +{ + vmidi_devc *devc = midi_devs[dev]->devc; + unsigned long flags; + + if (devc == NULL) + return; + + spin_lock_irqsave(&devc->lock,flags); + devc->intr_active = 0; + devc->input_opened = 0; + devc->opened = 0; + spin_unlock_irqrestore(&devc->lock,flags); +} + +static int v_midi_out (int dev, unsigned char midi_byte) +{ + vmidi_devc *devc = midi_devs[dev]->devc; + vmidi_devc *pdevc; + + if (devc == NULL) + return -ENXIO; + + pdevc = midi_devs[devc->pair_mididev]->devc; + if (pdevc->input_opened > 0){ + if (MIDIbuf_avail(pdevc->my_mididev) > 500) + return 0; + pdevc->midi_input_intr (pdevc->my_mididev, midi_byte); + } + return 1; +} + +static inline int v_midi_start_read (int dev) +{ + return 0; +} + +static int v_midi_end_read (int dev) +{ + vmidi_devc *devc = midi_devs[dev]->devc; + if (devc == NULL) + return -ENXIO; + + devc->intr_active = 0; + return 0; +} + +/* why -EPERM and not -EINVAL?? */ + +static inline int v_midi_ioctl (int dev, unsigned cmd, void __user *arg) +{ + return -EPERM; +} + + +#define MIDI_SYNTH_NAME "Loopback MIDI" +#define MIDI_SYNTH_CAPS SYNTH_CAP_INPUT + +#include "midi_synth.h" + +static struct midi_operations v_midi_operations = +{ + .owner = THIS_MODULE, + .info = {"Loopback MIDI Port 1", 0, 0, SNDCARD_VMIDI}, + .converter = &std_midi_synth, + .in_info = {0}, + .open = v_midi_open, + .close = v_midi_close, + .ioctl = v_midi_ioctl, + .outputc = v_midi_out, + .start_read = v_midi_start_read, + .end_read = v_midi_end_read, +}; + +static struct midi_operations v_midi_operations2 = +{ + .owner = THIS_MODULE, + .info = {"Loopback MIDI Port 2", 0, 0, SNDCARD_VMIDI}, + .converter = &std_midi_synth, + .in_info = {0}, + .open = v_midi_open, + .close = v_midi_close, + .ioctl = v_midi_ioctl, + .outputc = v_midi_out, + .start_read = v_midi_start_read, + .end_read = v_midi_end_read, +}; + +/* + * We kmalloc just one of these - it makes life simpler and the code + * cleaner and the memory handling far more efficient + */ + +struct vmidi_memory +{ + /* Must be first */ + struct midi_operations m_ops[2]; + struct synth_operations s_ops[2]; + struct vmidi_devc v_ops[2]; +}; + +static void __init attach_v_midi (struct address_info *hw_config) +{ + struct vmidi_memory *m; + /* printk("Attaching v_midi device.....\n"); */ + + midi1 = sound_alloc_mididev(); + if (midi1 == -1) + { + printk(KERN_ERR "v_midi: Too many midi devices detected\n"); + return; + } + + m = kmalloc(sizeof(struct vmidi_memory), GFP_KERNEL); + if (m == NULL) + { + printk(KERN_WARNING "Loopback MIDI: Failed to allocate memory\n"); + sound_unload_mididev(midi1); + return; + } + + midi_mem = m; + + midi_devs[midi1] = &m->m_ops[0]; + + + midi2 = sound_alloc_mididev(); + if (midi2 == -1) + { + printk (KERN_ERR "v_midi: Too many midi devices detected\n"); + kfree(m); + sound_unload_mididev(midi1); + return; + } + + midi_devs[midi2] = &m->m_ops[1]; + + /* printk("VMIDI1: %d VMIDI2: %d\n",midi1,midi2); */ + + /* for MIDI-1 */ + v_devc[0] = &m->v_ops[0]; + memcpy ((char *) midi_devs[midi1], (char *) &v_midi_operations, + sizeof (struct midi_operations)); + + v_devc[0]->my_mididev = midi1; + v_devc[0]->pair_mididev = midi2; + v_devc[0]->opened = v_devc[0]->input_opened = 0; + v_devc[0]->intr_active = 0; + v_devc[0]->midi_input_intr = NULL; + spin_lock_init(&v_devc[0]->lock); + + midi_devs[midi1]->devc = v_devc[0]; + + midi_devs[midi1]->converter = &m->s_ops[0]; + std_midi_synth.midi_dev = midi1; + memcpy ((char *) midi_devs[midi1]->converter, (char *) &std_midi_synth, + sizeof (struct synth_operations)); + midi_devs[midi1]->converter->id = "V_MIDI 1"; + + /* for MIDI-2 */ + v_devc[1] = &m->v_ops[1]; + + memcpy ((char *) midi_devs[midi2], (char *) &v_midi_operations2, + sizeof (struct midi_operations)); + + v_devc[1]->my_mididev = midi2; + v_devc[1]->pair_mididev = midi1; + v_devc[1]->opened = v_devc[1]->input_opened = 0; + v_devc[1]->intr_active = 0; + v_devc[1]->midi_input_intr = NULL; + spin_lock_init(&v_devc[1]->lock); + + midi_devs[midi2]->devc = v_devc[1]; + midi_devs[midi2]->converter = &m->s_ops[1]; + + std_midi_synth.midi_dev = midi2; + memcpy ((char *) midi_devs[midi2]->converter, (char *) &std_midi_synth, + sizeof (struct synth_operations)); + midi_devs[midi2]->converter->id = "V_MIDI 2"; + + sequencer_init(); + /* printk("Attached v_midi device\n"); */ +} + +static inline int __init probe_v_midi(struct address_info *hw_config) +{ + return(1); /* always OK */ +} + + +static void __exit unload_v_midi(struct address_info *hw_config) +{ + sound_unload_mididev(midi1); + sound_unload_mididev(midi2); + kfree(midi_mem); +} + +static struct address_info cfg; /* dummy */ + +static int __init init_vmidi(void) +{ + printk("MIDI Loopback device driver\n"); + if (!probe_v_midi(&cfg)) + return -ENODEV; + attach_v_midi(&cfg); + + return 0; +} + +static void __exit cleanup_vmidi(void) +{ + unload_v_midi(&cfg); +} + +module_init(init_vmidi); +module_exit(cleanup_vmidi); +MODULE_LICENSE("GPL"); diff --git a/sound/oss/v_midi.h b/sound/oss/v_midi.h new file mode 100644 index 00000000..08e2185e --- /dev/null +++ b/sound/oss/v_midi.h @@ -0,0 +1,14 @@ +typedef struct vmidi_devc { + int dev; + + /* State variables */ + int opened; + spinlock_t lock; + + /* MIDI fields */ + int my_mididev; + int pair_mididev; + int input_opened; + int intr_active; + void (*midi_input_intr) (int dev, unsigned char data); + } vmidi_devc; diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c new file mode 100644 index 00000000..12ba28e7 --- /dev/null +++ b/sound/oss/vidc.c @@ -0,0 +1,558 @@ +/* + * linux/drivers/sound/vidc.c + * + * Copyright (C) 1997-2000 by Russell King <rmk@arm.linux.org.uk> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * VIDC20 audio driver. + * + * The VIDC20 sound hardware consists of the VIDC20 itself, a DAC and a DMA + * engine. The DMA transfers fixed-format (16-bit little-endian linear) + * samples to the VIDC20, which then transfers this data serially to the + * DACs. The samplerate is controlled by the VIDC. + * + * We currently support a mixer device, but it is currently non-functional. + */ + +#include <linux/gfp.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/interrupt.h> + +#include <mach/hardware.h> +#include <asm/dma.h> +#include <asm/io.h> +#include <asm/hardware/iomd.h> +#include <asm/irq.h> +#include <asm/system.h> + +#include "sound_config.h" +#include "vidc.h" + +#ifndef _SIOC_TYPE +#define _SIOC_TYPE(x) _IOC_TYPE(x) +#endif +#ifndef _SIOC_NR +#define _SIOC_NR(x) _IOC_NR(x) +#endif + +#define VIDC_SOUND_CLOCK (250000) +#define VIDC_SOUND_CLOCK_EXT (176400) + +/* + * When using SERIAL SOUND mode (external DAC), the number of physical + * channels is fixed at 2. + */ +static int vidc_busy; +static int vidc_adev; +static int vidc_audio_rate; +static char vidc_audio_format; +static char vidc_audio_channels; + +static unsigned char vidc_level_l[SOUND_MIXER_NRDEVICES] = { + 85, /* master */ + 50, /* bass */ + 50, /* treble */ + 0, /* synth */ + 75, /* pcm */ + 0, /* speaker */ + 100, /* ext line */ + 0, /* mic */ + 100, /* CD */ + 0, +}; + +static unsigned char vidc_level_r[SOUND_MIXER_NRDEVICES] = { + 85, /* master */ + 50, /* bass */ + 50, /* treble */ + 0, /* synth */ + 75, /* pcm */ + 0, /* speaker */ + 100, /* ext line */ + 0, /* mic */ + 100, /* CD */ + 0, +}; + +static unsigned int vidc_audio_volume_l; /* left PCM vol, 0 - 65536 */ +static unsigned int vidc_audio_volume_r; /* right PCM vol, 0 - 65536 */ + +extern void vidc_update_filler(int bits, int channels); +extern int softoss_dev; + +static void +vidc_mixer_set(int mdev, unsigned int level) +{ + unsigned int lev_l = level & 0x007f; + unsigned int lev_r = (level & 0x7f00) >> 8; + unsigned int mlev_l, mlev_r; + + if (lev_l > 100) + lev_l = 100; + if (lev_r > 100) + lev_r = 100; + +#define SCALE(lev,master) ((lev) * (master) * 65536 / 10000) + + mlev_l = vidc_level_l[SOUND_MIXER_VOLUME]; + mlev_r = vidc_level_r[SOUND_MIXER_VOLUME]; + + switch (mdev) { + case SOUND_MIXER_VOLUME: + case SOUND_MIXER_PCM: + vidc_level_l[mdev] = lev_l; + vidc_level_r[mdev] = lev_r; + + vidc_audio_volume_l = SCALE(lev_l, mlev_l); + vidc_audio_volume_r = SCALE(lev_r, mlev_r); +/*printk("VIDC: PCM vol %05X %05X\n", vidc_audio_volume_l, vidc_audio_volume_r);*/ + break; + } +#undef SCALE +} + +static int vidc_mixer_ioctl(int dev, unsigned int cmd, void __user *arg) +{ + unsigned int val; + unsigned int mdev; + + if (_SIOC_TYPE(cmd) != 'M') + return -EINVAL; + + mdev = _SIOC_NR(cmd); + + if (_SIOC_DIR(cmd) & _SIOC_WRITE) { + if (get_user(val, (unsigned int __user *)arg)) + return -EFAULT; + + if (mdev < SOUND_MIXER_NRDEVICES) + vidc_mixer_set(mdev, val); + else + return -EINVAL; + } + + /* + * Return parameters + */ + switch (mdev) { + case SOUND_MIXER_RECSRC: + val = 0; + break; + + case SOUND_MIXER_DEVMASK: + val = SOUND_MASK_VOLUME | SOUND_MASK_PCM | SOUND_MASK_SYNTH; + break; + + case SOUND_MIXER_STEREODEVS: + val = SOUND_MASK_VOLUME | SOUND_MASK_PCM | SOUND_MASK_SYNTH; + break; + + case SOUND_MIXER_RECMASK: + val = 0; + break; + + case SOUND_MIXER_CAPS: + val = 0; + break; + + default: + if (mdev < SOUND_MIXER_NRDEVICES) + val = vidc_level_l[mdev] | vidc_level_r[mdev] << 8; + else + return -EINVAL; + } + + return put_user(val, (unsigned int __user *)arg) ? -EFAULT : 0; +} + +static unsigned int vidc_audio_set_format(int dev, unsigned int fmt) +{ + switch (fmt) { + default: + fmt = AFMT_S16_LE; + case AFMT_U8: + case AFMT_S8: + case AFMT_S16_LE: + vidc_audio_format = fmt; + vidc_update_filler(vidc_audio_format, vidc_audio_channels); + case AFMT_QUERY: + break; + } + return vidc_audio_format; +} + +#define my_abs(i) ((i)<0 ? -(i) : (i)) + +static int vidc_audio_set_speed(int dev, int rate) +{ + if (rate) { + unsigned int hwctrl, hwrate, hwrate_ext, rate_int, rate_ext; + unsigned int diff_int, diff_ext; + unsigned int newsize, new2size; + + hwctrl = 0x00000003; + + /* Using internal clock */ + hwrate = (((VIDC_SOUND_CLOCK * 2) / rate) + 1) >> 1; + if (hwrate < 3) + hwrate = 3; + if (hwrate > 255) + hwrate = 255; + + /* Using exernal clock */ + hwrate_ext = (((VIDC_SOUND_CLOCK_EXT * 2) / rate) + 1) >> 1; + if (hwrate_ext < 3) + hwrate_ext = 3; + if (hwrate_ext > 255) + hwrate_ext = 255; + + rate_int = VIDC_SOUND_CLOCK / hwrate; + rate_ext = VIDC_SOUND_CLOCK_EXT / hwrate_ext; + + /* Chose between external and internal clock */ + diff_int = my_abs(rate_ext-rate); + diff_ext = my_abs(rate_int-rate); + if (diff_ext < diff_int) { + /*printk("VIDC: external %d %d %d\n", rate, rate_ext, hwrate_ext);*/ + hwrate=hwrate_ext; + hwctrl=0x00000002; + /* Allow roughly 0.4% tolerance */ + if (diff_ext > (rate/256)) + rate=rate_ext; + } else { + /*printk("VIDC: internal %d %d %d\n", rate, rate_int, hwrate);*/ + hwctrl=0x00000003; + /* Allow roughly 0.4% tolerance */ + if (diff_int > (rate/256)) + rate=rate_int; + } + + vidc_writel(0xb0000000 | (hwrate - 2)); + vidc_writel(0xb1000000 | hwctrl); + + newsize = (10000 / hwrate) & ~3; + if (newsize < 208) + newsize = 208; + if (newsize > 4096) + newsize = 4096; + for (new2size = 128; new2size < newsize; new2size <<= 1); + if (new2size - newsize > newsize - (new2size >> 1)) + new2size >>= 1; + if (new2size > 4096) { + printk(KERN_ERR "VIDC: error: dma buffer (%d) %d > 4K\n", + newsize, new2size); + new2size = 4096; + } + /*printk("VIDC: dma size %d\n", new2size);*/ + dma_bufsize = new2size; + vidc_audio_rate = rate; + } + return vidc_audio_rate; +} + +static short vidc_audio_set_channels(int dev, short channels) +{ + switch (channels) { + default: + channels = 2; + case 1: + case 2: + vidc_audio_channels = channels; + vidc_update_filler(vidc_audio_format, vidc_audio_channels); + case 0: + break; + } + return vidc_audio_channels; +} + +/* + * Open the device + */ +static int vidc_audio_open(int dev, int mode) +{ + /* This audio device does not have recording capability */ + if (mode == OPEN_READ) + return -EPERM; + + if (vidc_busy) + return -EBUSY; + + vidc_busy = 1; + return 0; +} + +/* + * Close the device + */ +static void vidc_audio_close(int dev) +{ + vidc_busy = 0; +} + +/* + * Output a block via DMA to sound device. + * + * We just set the DMA start and count; the DMA interrupt routine + * will take care of formatting the samples (via the appropriate + * vidc_filler routine), and flag via vidc_audio_dma_interrupt when + * more data is required. + */ +static void +vidc_audio_output_block(int dev, unsigned long buf, int total_count, int one) +{ + struct dma_buffparms *dmap = audio_devs[dev]->dmap_out; + unsigned long flags; + + local_irq_save(flags); + dma_start = buf - (unsigned long)dmap->raw_buf_phys + (unsigned long)dmap->raw_buf; + dma_count = total_count; + local_irq_restore(flags); +} + +static void +vidc_audio_start_input(int dev, unsigned long buf, int count, int intrflag) +{ +} + +static int vidc_audio_prepare_for_input(int dev, int bsize, int bcount) +{ + return -EINVAL; +} + +static irqreturn_t vidc_audio_dma_interrupt(void) +{ + DMAbuf_outputintr(vidc_adev, 1); + return IRQ_HANDLED; +} + +/* + * Prepare for outputting samples. + * + * Each buffer that will be passed will be `bsize' bytes long, + * with a total of `bcount' buffers. + */ +static int vidc_audio_prepare_for_output(int dev, int bsize, int bcount) +{ + struct audio_operations *adev = audio_devs[dev]; + + dma_interrupt = NULL; + adev->dmap_out->flags |= DMA_NODMA; + + return 0; +} + +/* + * Stop our current operation. + */ +static void vidc_audio_reset(int dev) +{ + dma_interrupt = NULL; +} + +static int vidc_audio_local_qlen(int dev) +{ + return /*dma_count !=*/ 0; +} + +static void vidc_audio_trigger(int dev, int enable_bits) +{ + struct audio_operations *adev = audio_devs[dev]; + + if (enable_bits & PCM_ENABLE_OUTPUT) { + if (!(adev->dmap_out->flags & DMA_ACTIVE)) { + unsigned long flags; + + local_irq_save(flags); + + /* prevent recusion */ + adev->dmap_out->flags |= DMA_ACTIVE; + + dma_interrupt = vidc_audio_dma_interrupt; + vidc_sound_dma_irq(0, NULL); + iomd_writeb(DMA_CR_E | 0x10, IOMD_SD0CR); + + local_irq_restore(flags); + } + } +} + +static struct audio_driver vidc_audio_driver = +{ + .owner = THIS_MODULE, + .open = vidc_audio_open, + .close = vidc_audio_close, + .output_block = vidc_audio_output_block, + .start_input = vidc_audio_start_input, + .prepare_for_input = vidc_audio_prepare_for_input, + .prepare_for_output = vidc_audio_prepare_for_output, + .halt_io = vidc_audio_reset, + .local_qlen = vidc_audio_local_qlen, + .trigger = vidc_audio_trigger, + .set_speed = vidc_audio_set_speed, + .set_bits = vidc_audio_set_format, + .set_channels = vidc_audio_set_channels +}; + +static struct mixer_operations vidc_mixer_operations = { + .owner = THIS_MODULE, + .id = "VIDC", + .name = "VIDCsound", + .ioctl = vidc_mixer_ioctl +}; + +void vidc_update_filler(int format, int channels) +{ +#define TYPE(fmt,ch) (((fmt)<<2) | ((ch)&3)) + + switch (TYPE(format, channels)) { + default: + case TYPE(AFMT_U8, 1): + vidc_filler = vidc_fill_1x8_u; + break; + + case TYPE(AFMT_U8, 2): + vidc_filler = vidc_fill_2x8_u; + break; + + case TYPE(AFMT_S8, 1): + vidc_filler = vidc_fill_1x8_s; + break; + + case TYPE(AFMT_S8, 2): + vidc_filler = vidc_fill_2x8_s; + break; + + case TYPE(AFMT_S16_LE, 1): + vidc_filler = vidc_fill_1x16_s; + break; + + case TYPE(AFMT_S16_LE, 2): + vidc_filler = vidc_fill_2x16_s; + break; + } +} + +static void __init attach_vidc(struct address_info *hw_config) +{ + char name[32]; + int i, adev; + + sprintf(name, "VIDC %d-bit sound", hw_config->card_subtype); + conf_printf(name, hw_config); + memset(dma_buf, 0, sizeof(dma_buf)); + + adev = sound_install_audiodrv(AUDIO_DRIVER_VERSION, name, + &vidc_audio_driver, sizeof(vidc_audio_driver), + DMA_AUTOMODE, AFMT_U8 | AFMT_S8 | AFMT_S16_LE, + NULL, hw_config->dma, hw_config->dma2); + + if (adev < 0) + goto audio_failed; + + /* + * 1024 bytes => 64 buffers + */ + audio_devs[adev]->min_fragment = 10; + audio_devs[adev]->mixer_dev = num_mixers; + + audio_devs[adev]->mixer_dev = + sound_install_mixer(MIXER_DRIVER_VERSION, + name, &vidc_mixer_operations, + sizeof(vidc_mixer_operations), NULL); + + if (audio_devs[adev]->mixer_dev < 0) + goto mixer_failed; + + for (i = 0; i < 2; i++) { + dma_buf[i] = get_zeroed_page(GFP_KERNEL); + if (!dma_buf[i]) { + printk(KERN_ERR "%s: can't allocate required buffers\n", + name); + goto mem_failed; + } + dma_pbuf[i] = virt_to_phys((void *)dma_buf[i]); + } + + if (sound_alloc_dma(hw_config->dma, hw_config->name)) { + printk(KERN_ERR "%s: DMA %d is in use\n", name, hw_config->dma); + goto dma_failed; + } + + if (request_irq(hw_config->irq, vidc_sound_dma_irq, 0, + hw_config->name, &dma_start)) { + printk(KERN_ERR "%s: IRQ %d is in use\n", name, hw_config->irq); + goto irq_failed; + } + vidc_adev = adev; + vidc_mixer_set(SOUND_MIXER_VOLUME, (85 | 85 << 8)); + + return; + +irq_failed: + sound_free_dma(hw_config->dma); +dma_failed: +mem_failed: + for (i = 0; i < 2; i++) + free_page(dma_buf[i]); + sound_unload_mixerdev(audio_devs[adev]->mixer_dev); +mixer_failed: + sound_unload_audiodev(adev); +audio_failed: + return; +} + +static int __init probe_vidc(struct address_info *hw_config) +{ + hw_config->irq = IRQ_DMAS0; + hw_config->dma = DMA_VIRTUAL_SOUND; + hw_config->dma2 = -1; + hw_config->card_subtype = 16; + hw_config->name = "VIDC20"; + return 1; +} + +static void __exit unload_vidc(struct address_info *hw_config) +{ + int i, adev = vidc_adev; + + vidc_adev = -1; + + free_irq(hw_config->irq, &dma_start); + sound_free_dma(hw_config->dma); + + if (adev >= 0) { + sound_unload_mixerdev(audio_devs[adev]->mixer_dev); + sound_unload_audiodev(adev); + for (i = 0; i < 2; i++) + free_page(dma_buf[i]); + } +} + +static struct address_info cfg; + +static int __init init_vidc(void) +{ + if (probe_vidc(&cfg) == 0) + return -ENODEV; + + attach_vidc(&cfg); + + return 0; +} + +static void __exit cleanup_vidc(void) +{ + unload_vidc(&cfg); +} + +module_init(init_vidc); +module_exit(cleanup_vidc); + +MODULE_AUTHOR("Russell King"); +MODULE_DESCRIPTION("VIDC20 audio driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/oss/vidc.h b/sound/oss/vidc.h new file mode 100644 index 00000000..0d142475 --- /dev/null +++ b/sound/oss/vidc.h @@ -0,0 +1,63 @@ +/* + * linux/drivers/sound/vidc.h + * + * Copyright (C) 1997 Russell King <rmk@arm.linux.org.uk> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * VIDC sound function prototypes + */ + +/* vidc_fill.S */ + +/* + * Filler routines for different channels and sample sizes + */ + +extern unsigned long vidc_fill_1x8_u(unsigned long ibuf, unsigned long iend, + unsigned long obuf, int mask); +extern unsigned long vidc_fill_2x8_u(unsigned long ibuf, unsigned long iend, + unsigned long obuf, int mask); +extern unsigned long vidc_fill_1x8_s(unsigned long ibuf, unsigned long iend, + unsigned long obuf, int mask); +extern unsigned long vidc_fill_2x8_s(unsigned long ibuf, unsigned long iend, + unsigned long obuf, int mask); +extern unsigned long vidc_fill_1x16_s(unsigned long ibuf, unsigned long iend, + unsigned long obuf, int mask); +extern unsigned long vidc_fill_2x16_s(unsigned long ibuf, unsigned long iend, + unsigned long obuf, int mask); + +/* + * DMA Interrupt handler + */ + +extern irqreturn_t vidc_sound_dma_irq(int irqnr, void *ref); + +/* + * Filler routine pointer + */ + +extern unsigned long (*vidc_filler) (unsigned long ibuf, unsigned long iend, + unsigned long obuf, int mask); + +/* + * Virtual DMA buffer exhausted + */ + +extern irqreturn_t (*dma_interrupt) (void); + +/* + * Virtual DMA buffer addresses + */ + +extern unsigned long dma_start, dma_count, dma_bufsize; +extern unsigned long dma_buf[2], dma_pbuf[2]; + +/* vidc_synth.c */ + +extern void vidc_synth_init(struct address_info *hw_config); +extern void vidc_synth_exit(struct address_info *hw_config); +extern int vidc_synth_get_volume(void); +extern int vidc_synth_set_volume(int vol); diff --git a/sound/oss/vidc_fill.S b/sound/oss/vidc_fill.S new file mode 100644 index 00000000..bed34921 --- /dev/null +++ b/sound/oss/vidc_fill.S @@ -0,0 +1,218 @@ +/* + * linux/drivers/sound/vidc_fill.S + * + * Copyright (C) 1997 Russell King + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Filler routines for DMA buffers + */ +#include <linux/linkage.h> +#include <asm/assembler.h> +#include <mach/hardware.h> +#include <asm/hardware/iomd.h> + + .text + +ENTRY(vidc_fill_1x8_u) + mov ip, #0xff00 +1: cmp r0, r1 + bge vidc_clear + ldrb r4, [r0], #1 + eor r4, r4, #0x80 + and r4, ip, r4, lsl #8 + orr r4, r4, r4, lsl #16 + str r4, [r2], #4 + cmp r2, r3 + blt 1b + mov pc, lr + +ENTRY(vidc_fill_2x8_u) + mov ip, #0xff00 +1: cmp r0, r1 + bge vidc_clear + ldr r4, [r0], #2 + and r5, r4, ip + and r4, ip, r4, lsl #8 + orr r4, r4, r5, lsl #16 + orr r4, r4, r4, lsr #8 + str r4, [r2], #4 + cmp r2, r3 + blt 1b + mov pc, lr + +ENTRY(vidc_fill_1x8_s) + mov ip, #0xff00 +1: cmp r0, r1 + bge vidc_clear + ldrb r4, [r0], #1 + and r4, ip, r4, lsl #8 + orr r4, r4, r4, lsl #16 + str r4, [r2], #4 + cmp r2, r3 + blt 1b + mov pc, lr + +ENTRY(vidc_fill_2x8_s) + mov ip, #0xff00 +1: cmp r0, r1 + bge vidc_clear + ldr r4, [r0], #2 + and r5, r4, ip + and r4, ip, r4, lsl #8 + orr r4, r4, r5, lsl #16 + orr r4, r4, r4, lsr #8 + str r4, [r2], #4 + cmp r2, r3 + blt 1b + mov pc, lr + +ENTRY(vidc_fill_1x16_s) + mov ip, #0xff00 + orr ip, ip, ip, lsr #8 +1: cmp r0, r1 + bge vidc_clear + ldr r5, [r0], #2 + and r4, r5, ip + orr r4, r4, r4, lsl #16 + str r4, [r2], #4 + cmp r0, r1 + addlt r0, r0, #2 + andlt r4, r5, ip, lsl #16 + orrlt r4, r4, r4, lsr #16 + strlt r4, [r2], #4 + cmp r2, r3 + blt 1b + mov pc, lr + +ENTRY(vidc_fill_2x16_s) + mov ip, #0xff00 + orr ip, ip, ip, lsr #8 +1: cmp r0, r1 + bge vidc_clear + ldr r4, [r0], #4 + str r4, [r2], #4 + cmp r0, r1 + ldrlt r4, [r0], #4 + strlt r4, [r2], #4 + cmp r2, r3 + blt 1b + mov pc, lr + +ENTRY(vidc_fill_noaudio) + mov r0, #0 + mov r1, #0 +2: mov r4, #0 + mov r5, #0 +1: cmp r2, r3 + stmltia r2!, {r0, r1, r4, r5} + blt 1b + mov pc, lr + +ENTRY(vidc_clear) + mov r0, #0 + mov r1, #0 + tst r2, #4 + str r0, [r2], #4 + tst r2, #8 + stmia r2!, {r0, r1} + b 2b + +/* + * Call filler routines with: + * r0 = phys address + * r1 = phys end + * r2 = buffer + * Returns: + * r0 = new buffer address + * r2 = new buffer finish + * r4 = corrupted + * r5 = corrupted + * ip = corrupted + */ + +ENTRY(vidc_sound_dma_irq) + stmfd sp!, {r4 - r8, lr} + ldr r8, =dma_start + ldmia r8, {r0, r1, r2, r3, r4, r5} + teq r1, #0 + adreq r4, vidc_fill_noaudio + moveq r7, #1 << 31 + movne r7, #0 + mov ip, #IOMD_BASE & 0xff000000 + orr ip, ip, #IOMD_BASE & 0x00ff0000 + ldrb r6, [ip, #IOMD_SD0ST] + tst r6, #DMA_ST_OFL @ Check for overrun + eorne r6, r6, #DMA_ST_AB + tst r6, #DMA_ST_AB + moveq r2, r3 @ DMAing A, update B + add r3, r2, r5 @ End of DMA buffer + add r1, r1, r0 @ End of virtual DMA buffer + mov lr, pc + mov pc, r4 @ Call fill routine (uses r4, ip) + sub r1, r1, r0 @ Remaining length + stmia r8, {r0, r1} + mov r0, #0 + tst r2, #4 @ Round buffer up to 4 words + strne r0, [r2], #4 + tst r2, #8 + strne r0, [r2], #4 + strne r0, [r2], #4 + sub r2, r2, #16 + mov r2, r2, lsl #20 + movs r2, r2, lsr #20 + orreq r2, r2, #1 << 30 @ Set L bit + orr r2, r2, r7 + ldmdb r8, {r3, r4, r5} + tst r6, #DMA_ST_AB + mov ip, #IOMD_BASE & 0xff000000 + orr ip, ip, #IOMD_BASE & 0x00ff0000 + streq r4, [ip, #IOMD_SD0CURB] + strne r5, [ip, #IOMD_SD0CURA] + streq r2, [ip, #IOMD_SD0ENDB] + strne r2, [ip, #IOMD_SD0ENDA] + ldr lr, [ip, #IOMD_SD0ST] + tst lr, #DMA_ST_OFL + bne 1f + tst r6, #DMA_ST_AB + strne r4, [ip, #IOMD_SD0CURB] + streq r5, [ip, #IOMD_SD0CURA] + strne r2, [ip, #IOMD_SD0ENDB] + streq r2, [ip, #IOMD_SD0ENDA] +1: teq r7, #0 + mov r0, #0x10 + strneb r0, [ip, #IOMD_SD0CR] + ldmfd sp!, {r4 - r8, lr} + mov r0, #1 @ IRQ_HANDLED + teq r1, #0 @ If we have no more + movne pc, lr + teq r3, #0 + movne pc, r3 @ Call interrupt routine + mov pc, lr + + .data + .globl dma_interrupt +dma_interrupt: + .long 0 @ r3 + .globl dma_pbuf +dma_pbuf: + .long 0 @ r4 + .long 0 @ r5 + .globl dma_start +dma_start: + .long 0 @ r0 + .globl dma_count +dma_count: + .long 0 @ r1 + .globl dma_buf +dma_buf: + .long 0 @ r2 + .long 0 @ r3 + .globl vidc_filler +vidc_filler: + .long vidc_fill_noaudio @ r4 + .globl dma_bufsize +dma_bufsize: + .long 0x1000 @ r5 diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c new file mode 100644 index 00000000..643f1113 --- /dev/null +++ b/sound/oss/vwsnd.c @@ -0,0 +1,3498 @@ +/* + * Sound driver for Silicon Graphics 320 and 540 Visual Workstations' + * onboard audio. See notes in Documentation/sound/oss/vwsnd . + * + * Copyright 1999 Silicon Graphics, Inc. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#undef VWSND_DEBUG /* define for debugging */ + +/* + * XXX to do - + * + * External sync. + * Rename swbuf, hwbuf, u&i, hwptr&swptr to something rational. + * Bug - if select() called before read(), pcm_setup() not called. + * Bug - output doesn't stop soon enough if process killed. + */ + +/* + * Things to test - + * + * Will readv/writev work? Write a test. + * + * insmod/rmmod 100 million times. + * + * Run I/O until int ptrs wrap around (roughly 6.2 hours @ DAT + * rate). + * + * Concurrent threads banging on mixer simultaneously, both UP + * and SMP kernels. Especially, watch for thread A changing + * OUTSRC while thread B changes gain -- both write to the same + * ad1843 register. + * + * What happens if a client opens /dev/audio then forks? + * Do two procs have /dev/audio open? Test. + * + * Pump audio through the CD, MIC and line inputs and verify that + * they mix/mute into the output. + * + * Apps: + * amp + * mpg123 + * x11amp + * mxv + * kmedia + * esound + * need more input apps + * + * Run tests while bombarding with signals. setitimer(2) will do it... */ + +/* + * This driver is organized in nine sections. + * The nine sections are: + * + * debug stuff + * low level lithium access + * high level lithium access + * AD1843 access + * PCM I/O + * audio driver + * mixer driver + * probe/attach/unload + * initialization and loadable kernel module interface + * + * That is roughly the order of increasing abstraction, so forward + * dependencies are minimal. + */ + +/* + * Locking Notes + * + * INC_USE_COUNT and DEC_USE_COUNT keep track of the number of + * open descriptors to this driver. They store it in vwsnd_use_count. + * The global device list, vwsnd_dev_list, is immutable when the IN_USE + * is true. + * + * devc->open_lock is a semaphore that is used to enforce the + * single reader/single writer rule for /dev/audio. The rule is + * that each device may have at most one reader and one writer. + * Open will block until the previous client has closed the + * device, unless O_NONBLOCK is specified. + * + * The semaphore devc->io_mutex serializes PCM I/O syscalls. This + * is unnecessary in Linux 2.2, because the kernel lock + * serializes read, write, and ioctl globally, but it's there, + * ready for the brave, new post-kernel-lock world. + * + * Locking between interrupt and baselevel is handled by the + * "lock" spinlock in vwsnd_port (one lock each for read and + * write). Each half holds the lock just long enough to see what + * area it owns and update its pointers. See pcm_output() and + * pcm_input() for most of the gory stuff. + * + * devc->mix_mutex serializes all mixer ioctls. This is also + * redundant because of the kernel lock. + * + * The lowest level lock is lith->lithium_lock. It is a + * spinlock which is held during the two-register tango of + * reading/writing an AD1843 register. See + * li_{read,write}_ad1843_reg(). + */ + +/* + * Sample Format Notes + * + * Lithium's DMA engine has two formats: 16-bit 2's complement + * and 8-bit unsigned . 16-bit transfers the data unmodified, 2 + * bytes per sample. 8-bit unsigned transfers 1 byte per sample + * and XORs each byte with 0x80. Lithium can input or output + * either mono or stereo in either format. + * + * The AD1843 has four formats: 16-bit 2's complement, 8-bit + * unsigned, 8-bit mu-Law and 8-bit A-Law. + * + * This driver supports five formats: AFMT_S8, AFMT_U8, + * AFMT_MU_LAW, AFMT_A_LAW, and AFMT_S16_LE. + * + * For AFMT_U8 output, we keep the AD1843 in 16-bit mode, and + * rely on Lithium's XOR to translate between U8 and S8. + * + * For AFMT_S8, AFMT_MU_LAW and AFMT_A_LAW output, we have to XOR + * the 0x80 bit in software to compensate for Lithium's XOR. + * This happens in pcm_copy_{in,out}(). + * + * Changes: + * 11-10-2000 Bartlomiej Zolnierkiewicz <bkz@linux-ide.org> + * Added some __init/__exit + */ + +#include <linux/module.h> +#include <linux/init.h> + +#include <linux/spinlock.h> +#include <linux/wait.h> +#include <linux/interrupt.h> +#include <linux/mutex.h> +#include <linux/slab.h> + +#include <asm/visws/cobalt.h> + +#include "sound_config.h" + +/*****************************************************************************/ +/* debug stuff */ + +#ifdef VWSND_DEBUG + +static DEFINE_MUTEX(vwsnd_mutex); +static int shut_up = 1; + +/* + * dbgassert - called when an assertion fails. + */ + +static void dbgassert(const char *fcn, int line, const char *expr) +{ + if (in_interrupt()) + panic("ASSERTION FAILED IN INTERRUPT, %s:%s:%d %s\n", + __FILE__, fcn, line, expr); + else { + int x; + printk(KERN_ERR "ASSERTION FAILED, %s:%s:%d %s\n", + __FILE__, fcn, line, expr); + x = * (volatile int *) 0; /* force proc to exit */ + } +} + +/* + * Bunch of useful debug macros: + * + * ASSERT - print unless e nonzero (panic if in interrupt) + * DBGDO - include arbitrary code if debugging + * DBGX - debug print raw (w/o function name) + * DBGP - debug print w/ function name + * DBGE - debug print function entry + * DBGC - debug print function call + * DBGR - debug print function return + * DBGXV - debug print raw when verbose + * DBGPV - debug print when verbose + * DBGEV - debug print function entry when verbose + * DBGRV - debug print function return when verbose + */ + +#define ASSERT(e) ((e) ? (void) 0 : dbgassert(__func__, __LINE__, #e)) +#define DBGDO(x) x +#define DBGX(fmt, args...) (in_interrupt() ? 0 : printk(KERN_ERR fmt, ##args)) +#define DBGP(fmt, args...) (DBGX("%s: " fmt, __func__ , ##args)) +#define DBGE(fmt, args...) (DBGX("%s" fmt, __func__ , ##args)) +#define DBGC(rtn) (DBGP("calling %s\n", rtn)) +#define DBGR() (DBGP("returning\n")) +#define DBGXV(fmt, args...) (shut_up ? 0 : DBGX(fmt, ##args)) +#define DBGPV(fmt, args...) (shut_up ? 0 : DBGP(fmt, ##args)) +#define DBGEV(fmt, args...) (shut_up ? 0 : DBGE(fmt, ##args)) +#define DBGCV(rtn) (shut_up ? 0 : DBGC(rtn)) +#define DBGRV() (shut_up ? 0 : DBGR()) + +#else /* !VWSND_DEBUG */ + +#define ASSERT(e) ((void) 0) +#define DBGDO(x) /* don't */ +#define DBGX(fmt, args...) ((void) 0) +#define DBGP(fmt, args...) ((void) 0) +#define DBGE(fmt, args...) ((void) 0) +#define DBGC(rtn) ((void) 0) +#define DBGR() ((void) 0) +#define DBGPV(fmt, args...) ((void) 0) +#define DBGXV(fmt, args...) ((void) 0) +#define DBGEV(fmt, args...) ((void) 0) +#define DBGCV(rtn) ((void) 0) +#define DBGRV() ((void) 0) + +#endif /* !VWSND_DEBUG */ + +/*****************************************************************************/ +/* low level lithium access */ + +/* + * We need to talk to Lithium registers on three pages. Here are + * the pages' offsets from the base address (0xFF001000). + */ + +enum { + LI_PAGE0_OFFSET = 0x01000 - 0x1000, /* FF001000 */ + LI_PAGE1_OFFSET = 0x0F000 - 0x1000, /* FF00F000 */ + LI_PAGE2_OFFSET = 0x10000 - 0x1000, /* FF010000 */ +}; + +/* low-level lithium data */ + +typedef struct lithium { + void * page0; /* virtual addresses */ + void * page1; + void * page2; + spinlock_t lock; /* protects codec and UST/MSC access */ +} lithium_t; + +/* + * li_destroy destroys the lithium_t structure and vm mappings. + */ + +static void li_destroy(lithium_t *lith) +{ + if (lith->page0) { + iounmap(lith->page0); + lith->page0 = NULL; + } + if (lith->page1) { + iounmap(lith->page1); + lith->page1 = NULL; + } + if (lith->page2) { + iounmap(lith->page2); + lith->page2 = NULL; + } +} + +/* + * li_create initializes the lithium_t structure and sets up vm mappings + * to access the registers. + * Returns 0 on success, -errno on failure. + */ + +static int __init li_create(lithium_t *lith, unsigned long baseaddr) +{ + spin_lock_init(&lith->lock); + lith->page0 = ioremap_nocache(baseaddr + LI_PAGE0_OFFSET, PAGE_SIZE); + lith->page1 = ioremap_nocache(baseaddr + LI_PAGE1_OFFSET, PAGE_SIZE); + lith->page2 = ioremap_nocache(baseaddr + LI_PAGE2_OFFSET, PAGE_SIZE); + if (!lith->page0 || !lith->page1 || !lith->page2) { + li_destroy(lith); + return -ENOMEM; + } + return 0; +} + +/* + * basic register accessors - read/write long/byte + */ + +static __inline__ unsigned long li_readl(lithium_t *lith, int off) +{ + return * (volatile unsigned long *) (lith->page0 + off); +} + +static __inline__ unsigned char li_readb(lithium_t *lith, int off) +{ + return * (volatile unsigned char *) (lith->page0 + off); +} + +static __inline__ void li_writel(lithium_t *lith, int off, unsigned long val) +{ + * (volatile unsigned long *) (lith->page0 + off) = val; +} + +static __inline__ void li_writeb(lithium_t *lith, int off, unsigned char val) +{ + * (volatile unsigned char *) (lith->page0 + off) = val; +} + +/*****************************************************************************/ +/* High Level Lithium Access */ + +/* + * Lithium DMA Notes + * + * Lithium has two dedicated DMA channels for audio. They are known + * as comm1 and comm2 (communication areas 1 and 2). Comm1 is for + * input, and comm2 is for output. Each is controlled by three + * registers: BASE (base address), CFG (config) and CCTL + * (config/control). + * + * Each DMA channel points to a physically contiguous ring buffer in + * main memory of up to 8 Kbytes. (This driver always uses 8 Kb.) + * There are three pointers into the ring buffer: read, write, and + * trigger. The pointers are 8 bits each. Each pointer points to + * 32-byte "chunks" of data. The DMA engine moves 32 bytes at a time, + * so there is no finer-granularity control. + * + * In comm1, the hardware updates the write ptr, and software updates + * the read ptr. In comm2, it's the opposite: hardware updates the + * read ptr, and software updates the write ptr. I designate the + * hardware-updated ptr as the hwptr, and the software-updated ptr as + * the swptr. + * + * The trigger ptr and trigger mask are used to trigger interrupts. + * From the Lithium spec, section 5.6.8, revision of 12/15/1998: + * + * Trigger Mask Value + * + * A three bit wide field that represents a power of two mask + * that is used whenever the trigger pointer is compared to its + * respective read or write pointer. A value of zero here + * implies a mask of 0xFF and a value of seven implies a mask + * 0x01. This value can be used to sub-divide the ring buffer + * into pie sections so that interrupts monitor the progress of + * hardware from section to section. + * + * My interpretation of that is, whenever the hw ptr is updated, it is + * compared with the trigger ptr, and the result is masked by the + * trigger mask. (Actually, by the complement of the trigger mask.) + * If the result is zero, an interrupt is triggered. I.e., interrupt + * if ((hwptr & ~mask) == (trptr & ~mask)). The mask is formed from + * the trigger register value as mask = (1 << (8 - tmreg)) - 1. + * + * In yet different words, setting tmreg to 0 causes an interrupt after + * every 256 DMA chunks (8192 bytes) or once per traversal of the + * ring buffer. Setting it to 7 caues an interrupt every 2 DMA chunks + * (64 bytes) or 128 times per traversal of the ring buffer. + */ + +/* Lithium register offsets and bit definitions */ + +#define LI_HOST_CONTROLLER 0x000 +# define LI_HC_RESET 0x00008000 +# define LI_HC_LINK_ENABLE 0x00004000 +# define LI_HC_LINK_FAILURE 0x00000004 +# define LI_HC_LINK_CODEC 0x00000002 +# define LI_HC_LINK_READY 0x00000001 + +#define LI_INTR_STATUS 0x010 +#define LI_INTR_MASK 0x014 +# define LI_INTR_LINK_ERR 0x00008000 +# define LI_INTR_COMM2_TRIG 0x00000008 +# define LI_INTR_COMM2_UNDERFLOW 0x00000004 +# define LI_INTR_COMM1_TRIG 0x00000002 +# define LI_INTR_COMM1_OVERFLOW 0x00000001 + +#define LI_CODEC_COMMAND 0x018 +# define LI_CC_BUSY 0x00008000 +# define LI_CC_DIR 0x00000080 +# define LI_CC_DIR_RD LI_CC_DIR +# define LI_CC_DIR_WR (!LI_CC_DIR) +# define LI_CC_ADDR_MASK 0x0000007F + +#define LI_CODEC_DATA 0x01C + +#define LI_COMM1_BASE 0x100 +#define LI_COMM1_CTL 0x104 +# define LI_CCTL_RESET 0x80000000 +# define LI_CCTL_SIZE 0x70000000 +# define LI_CCTL_DMA_ENABLE 0x08000000 +# define LI_CCTL_TMASK 0x07000000 /* trigger mask */ +# define LI_CCTL_TPTR 0x00FF0000 /* trigger pointer */ +# define LI_CCTL_RPTR 0x0000FF00 +# define LI_CCTL_WPTR 0x000000FF +#define LI_COMM1_CFG 0x108 +# define LI_CCFG_LOCK 0x00008000 +# define LI_CCFG_SLOT 0x00000070 +# define LI_CCFG_DIRECTION 0x00000008 +# define LI_CCFG_DIR_IN (!LI_CCFG_DIRECTION) +# define LI_CCFG_DIR_OUT LI_CCFG_DIRECTION +# define LI_CCFG_MODE 0x00000004 +# define LI_CCFG_MODE_MONO (!LI_CCFG_MODE) +# define LI_CCFG_MODE_STEREO LI_CCFG_MODE +# define LI_CCFG_FORMAT 0x00000003 +# define LI_CCFG_FMT_8BIT 0x00000000 +# define LI_CCFG_FMT_16BIT 0x00000001 +#define LI_COMM2_BASE 0x10C +#define LI_COMM2_CTL 0x110 + /* bit definitions are the same as LI_COMM1_CTL */ +#define LI_COMM2_CFG 0x114 + /* bit definitions are the same as LI_COMM1_CFG */ + +#define LI_UST_LOW 0x200 /* 64-bit Unadjusted System Time is */ +#define LI_UST_HIGH 0x204 /* microseconds since boot */ + +#define LI_AUDIO1_UST 0x300 /* UST-MSC pairs */ +#define LI_AUDIO1_MSC 0x304 /* MSC (Media Stream Counter) */ +#define LI_AUDIO2_UST 0x308 /* counts samples actually */ +#define LI_AUDIO2_MSC 0x30C /* processed as of time UST */ + +/* + * Lithium's DMA engine operates on chunks of 32 bytes. We call that + * a DMACHUNK. + */ + +#define DMACHUNK_SHIFT 5 +#define DMACHUNK_SIZE (1 << DMACHUNK_SHIFT) +#define BYTES_TO_CHUNKS(bytes) ((bytes) >> DMACHUNK_SHIFT) +#define CHUNKS_TO_BYTES(chunks) ((chunks) << DMACHUNK_SHIFT) + +/* + * Two convenient macros to shift bitfields into/out of position. + * + * Observe that (mask & -mask) is (1 << low_set_bit_of(mask)). + * As long as mask is constant, we trust the compiler will change the + * multipy and divide into shifts. + */ + +#define SHIFT_FIELD(val, mask) (((val) * ((mask) & -(mask))) & (mask)) +#define UNSHIFT_FIELD(val, mask) (((val) & (mask)) / ((mask) & -(mask))) + +/* + * dma_chan_desc is invariant information about a Lithium + * DMA channel. There are two instances, li_comm1 and li_comm2. + * + * Note that the CCTL register fields are write ptr and read ptr, but what + * we care about are which pointer is updated by software and which by + * hardware. + */ + +typedef struct dma_chan_desc { + int basereg; + int cfgreg; + int ctlreg; + int hwptrreg; + int swptrreg; + int ustreg; + int mscreg; + unsigned long swptrmask; + int ad1843_slot; + int direction; /* LI_CCTL_DIR_IN/OUT */ +} dma_chan_desc_t; + +static const dma_chan_desc_t li_comm1 = { + LI_COMM1_BASE, /* base register offset */ + LI_COMM1_CFG, /* config register offset */ + LI_COMM1_CTL, /* control register offset */ + LI_COMM1_CTL + 0, /* hw ptr reg offset (write ptr) */ + LI_COMM1_CTL + 1, /* sw ptr reg offset (read ptr) */ + LI_AUDIO1_UST, /* ust reg offset */ + LI_AUDIO1_MSC, /* msc reg offset */ + LI_CCTL_RPTR, /* sw ptr bitmask in ctlval */ + 2, /* ad1843 serial slot */ + LI_CCFG_DIR_IN /* direction */ +}; + +static const dma_chan_desc_t li_comm2 = { + LI_COMM2_BASE, /* base register offset */ + LI_COMM2_CFG, /* config register offset */ + LI_COMM2_CTL, /* control register offset */ + LI_COMM2_CTL + 1, /* hw ptr reg offset (read ptr) */ + LI_COMM2_CTL + 0, /* sw ptr reg offset (writr ptr) */ + LI_AUDIO2_UST, /* ust reg offset */ + LI_AUDIO2_MSC, /* msc reg offset */ + LI_CCTL_WPTR, /* sw ptr bitmask in ctlval */ + 2, /* ad1843 serial slot */ + LI_CCFG_DIR_OUT /* direction */ +}; + +/* + * dma_chan is variable information about a Lithium DMA channel. + * + * The desc field points to invariant information. + * The lith field points to a lithium_t which is passed + * to li_read* and li_write* to access the registers. + * The *val fields shadow the lithium registers' contents. + */ + +typedef struct dma_chan { + const dma_chan_desc_t *desc; + lithium_t *lith; + unsigned long baseval; + unsigned long cfgval; + unsigned long ctlval; +} dma_chan_t; + +/* + * ustmsc is a UST/MSC pair (Unadjusted System Time/Media Stream Counter). + * UST is time in microseconds since the system booted, and MSC is a + * counter that increments with every audio sample. + */ + +typedef struct ustmsc { + unsigned long long ust; + unsigned long msc; +} ustmsc_t; + +/* + * li_ad1843_wait waits until lithium says the AD1843 register + * exchange is not busy. Returns 0 on success, -EBUSY on timeout. + * + * Locking: must be called with lithium_lock held. + */ + +static int li_ad1843_wait(lithium_t *lith) +{ + unsigned long later = jiffies + 2; + while (li_readl(lith, LI_CODEC_COMMAND) & LI_CC_BUSY) + if (time_after_eq(jiffies, later)) + return -EBUSY; + return 0; +} + +/* + * li_read_ad1843_reg returns the current contents of a 16 bit AD1843 register. + * + * Returns unsigned register value on success, -errno on failure. + */ + +static int li_read_ad1843_reg(lithium_t *lith, int reg) +{ + int val; + + ASSERT(!in_interrupt()); + spin_lock(&lith->lock); + { + val = li_ad1843_wait(lith); + if (val == 0) { + li_writel(lith, LI_CODEC_COMMAND, LI_CC_DIR_RD | reg); + val = li_ad1843_wait(lith); + } + if (val == 0) + val = li_readl(lith, LI_CODEC_DATA); + } + spin_unlock(&lith->lock); + + DBGXV("li_read_ad1843_reg(lith=0x%p, reg=%d) returns 0x%04x\n", + lith, reg, val); + + return val; +} + +/* + * li_write_ad1843_reg writes the specified value to a 16 bit AD1843 register. + */ + +static void li_write_ad1843_reg(lithium_t *lith, int reg, int newval) +{ + spin_lock(&lith->lock); + { + if (li_ad1843_wait(lith) == 0) { + li_writel(lith, LI_CODEC_DATA, newval); + li_writel(lith, LI_CODEC_COMMAND, LI_CC_DIR_WR | reg); + } + } + spin_unlock(&lith->lock); +} + +/* + * li_setup_dma calculates all the register settings for DMA in a particular + * mode. It takes too many arguments. + */ + +static void li_setup_dma(dma_chan_t *chan, + const dma_chan_desc_t *desc, + lithium_t *lith, + unsigned long buffer_paddr, + int bufshift, + int fragshift, + int channels, + int sampsize) +{ + unsigned long mode, format; + unsigned long size, tmask; + + DBGEV("(chan=0x%p, desc=0x%p, lith=0x%p, buffer_paddr=0x%lx, " + "bufshift=%d, fragshift=%d, channels=%d, sampsize=%d)\n", + chan, desc, lith, buffer_paddr, + bufshift, fragshift, channels, sampsize); + + /* Reset the channel first. */ + + li_writel(lith, desc->ctlreg, LI_CCTL_RESET); + + ASSERT(channels == 1 || channels == 2); + if (channels == 2) + mode = LI_CCFG_MODE_STEREO; + else + mode = LI_CCFG_MODE_MONO; + ASSERT(sampsize == 1 || sampsize == 2); + if (sampsize == 2) + format = LI_CCFG_FMT_16BIT; + else + format = LI_CCFG_FMT_8BIT; + chan->desc = desc; + chan->lith = lith; + + /* + * Lithium DMA address register takes a 40-bit physical + * address, right-shifted by 8 so it fits in 32 bits. Bit 37 + * must be set -- it enables cache coherence. + */ + + ASSERT(!(buffer_paddr & 0xFF)); + chan->baseval = (buffer_paddr >> 8) | 1 << (37 - 8); + + chan->cfgval = ((chan->cfgval & ~LI_CCFG_LOCK) | + SHIFT_FIELD(desc->ad1843_slot, LI_CCFG_SLOT) | + desc->direction | + mode | + format); + + size = bufshift - 6; + tmask = 13 - fragshift; /* See Lithium DMA Notes above. */ + ASSERT(size >= 2 && size <= 7); + ASSERT(tmask >= 1 && tmask <= 7); + chan->ctlval = ((chan->ctlval & ~LI_CCTL_RESET) | + SHIFT_FIELD(size, LI_CCTL_SIZE) | + (chan->ctlval & ~LI_CCTL_DMA_ENABLE) | + SHIFT_FIELD(tmask, LI_CCTL_TMASK) | + SHIFT_FIELD(0, LI_CCTL_TPTR)); + + DBGPV("basereg 0x%x = 0x%lx\n", desc->basereg, chan->baseval); + DBGPV("cfgreg 0x%x = 0x%lx\n", desc->cfgreg, chan->cfgval); + DBGPV("ctlreg 0x%x = 0x%lx\n", desc->ctlreg, chan->ctlval); + + li_writel(lith, desc->basereg, chan->baseval); + li_writel(lith, desc->cfgreg, chan->cfgval); + li_writel(lith, desc->ctlreg, chan->ctlval); + + DBGRV(); +} + +static void li_shutdown_dma(dma_chan_t *chan) +{ + lithium_t *lith = chan->lith; + void * lith1 = lith->page1; + + DBGEV("(chan=0x%p)\n", chan); + + chan->ctlval &= ~LI_CCTL_DMA_ENABLE; + DBGPV("ctlreg 0x%x = 0x%lx\n", chan->desc->ctlreg, chan->ctlval); + li_writel(lith, chan->desc->ctlreg, chan->ctlval); + + /* + * Offset 0x500 on Lithium page 1 is an undocumented, + * unsupported register that holds the zero sample value. + * Lithium is supposed to output zero samples when DMA is + * inactive, and repeat the last sample when DMA underflows. + * But it has a bug, where, after underflow occurs, the zero + * sample is not reset. + * + * I expect this to break in a future rev of Lithium. + */ + + if (lith1 && chan->desc->direction == LI_CCFG_DIR_OUT) + * (volatile unsigned long *) (lith1 + 0x500) = 0; +} + +/* + * li_activate_dma always starts dma at the beginning of the buffer. + * + * N.B., these may be called from interrupt. + */ + +static __inline__ void li_activate_dma(dma_chan_t *chan) +{ + chan->ctlval |= LI_CCTL_DMA_ENABLE; + DBGPV("ctlval = 0x%lx\n", chan->ctlval); + li_writel(chan->lith, chan->desc->ctlreg, chan->ctlval); +} + +static void li_deactivate_dma(dma_chan_t *chan) +{ + lithium_t *lith = chan->lith; + void * lith2 = lith->page2; + + chan->ctlval &= ~(LI_CCTL_DMA_ENABLE | LI_CCTL_RPTR | LI_CCTL_WPTR); + DBGPV("ctlval = 0x%lx\n", chan->ctlval); + DBGPV("ctlreg 0x%x = 0x%lx\n", chan->desc->ctlreg, chan->ctlval); + li_writel(lith, chan->desc->ctlreg, chan->ctlval); + + /* + * Offsets 0x98 and 0x9C on Lithium page 2 are undocumented, + * unsupported registers that are internal copies of the DMA + * read and write pointers. Because of a Lithium bug, these + * registers aren't zeroed correctly when DMA is shut off. So + * we whack them directly. + * + * I expect this to break in a future rev of Lithium. + */ + + if (lith2 && chan->desc->direction == LI_CCFG_DIR_OUT) { + * (volatile unsigned long *) (lith2 + 0x98) = 0; + * (volatile unsigned long *) (lith2 + 0x9C) = 0; + } +} + +/* + * read/write the ring buffer pointers. These routines' arguments and results + * are byte offsets from the beginning of the ring buffer. + */ + +static __inline__ int li_read_swptr(dma_chan_t *chan) +{ + const unsigned long mask = chan->desc->swptrmask; + + return CHUNKS_TO_BYTES(UNSHIFT_FIELD(chan->ctlval, mask)); +} + +static __inline__ int li_read_hwptr(dma_chan_t *chan) +{ + return CHUNKS_TO_BYTES(li_readb(chan->lith, chan->desc->hwptrreg)); +} + +static __inline__ void li_write_swptr(dma_chan_t *chan, int val) +{ + const unsigned long mask = chan->desc->swptrmask; + + ASSERT(!(val & ~CHUNKS_TO_BYTES(0xFF))); + val = BYTES_TO_CHUNKS(val); + chan->ctlval = (chan->ctlval & ~mask) | SHIFT_FIELD(val, mask); + li_writeb(chan->lith, chan->desc->swptrreg, val); +} + +/* li_read_USTMSC() returns a UST/MSC pair for the given channel. */ + +static void li_read_USTMSC(dma_chan_t *chan, ustmsc_t *ustmsc) +{ + lithium_t *lith = chan->lith; + const dma_chan_desc_t *desc = chan->desc; + unsigned long now_low, now_high0, now_high1, chan_ust; + + spin_lock(&lith->lock); + { + /* + * retry until we do all five reads without the + * high word changing. (High word increments + * every 2^32 microseconds, i.e., not often) + */ + do { + now_high0 = li_readl(lith, LI_UST_HIGH); + now_low = li_readl(lith, LI_UST_LOW); + + /* + * Lithium guarantees these two reads will be + * atomic -- ust will not increment after msc + * is read. + */ + + ustmsc->msc = li_readl(lith, desc->mscreg); + chan_ust = li_readl(lith, desc->ustreg); + + now_high1 = li_readl(lith, LI_UST_HIGH); + } while (now_high0 != now_high1); + } + spin_unlock(&lith->lock); + ustmsc->ust = ((unsigned long long) now_high0 << 32 | chan_ust); +} + +static void li_enable_interrupts(lithium_t *lith, unsigned int mask) +{ + DBGEV("(lith=0x%p, mask=0x%x)\n", lith, mask); + + /* clear any already-pending interrupts. */ + + li_writel(lith, LI_INTR_STATUS, mask); + + /* enable the interrupts. */ + + mask |= li_readl(lith, LI_INTR_MASK); + li_writel(lith, LI_INTR_MASK, mask); +} + +static void li_disable_interrupts(lithium_t *lith, unsigned int mask) +{ + unsigned int keepmask; + + DBGEV("(lith=0x%p, mask=0x%x)\n", lith, mask); + + /* disable the interrupts */ + + keepmask = li_readl(lith, LI_INTR_MASK) & ~mask; + li_writel(lith, LI_INTR_MASK, keepmask); + + /* clear any pending interrupts. */ + + li_writel(lith, LI_INTR_STATUS, mask); +} + +/* Get the interrupt status and clear all pending interrupts. */ + +static unsigned int li_get_clear_intr_status(lithium_t *lith) +{ + unsigned int status; + + status = li_readl(lith, LI_INTR_STATUS); + li_writel(lith, LI_INTR_STATUS, ~0); + return status & li_readl(lith, LI_INTR_MASK); +} + +static int li_init(lithium_t *lith) +{ + /* 1. System power supplies stabilize. */ + + /* 2. Assert the ~RESET signal. */ + + li_writel(lith, LI_HOST_CONTROLLER, LI_HC_RESET); + udelay(1); + + /* 3. Deassert the ~RESET signal and enter a wait period to allow + the AD1843 internal clocks and the external crystal oscillator + to stabilize. */ + + li_writel(lith, LI_HOST_CONTROLLER, LI_HC_LINK_ENABLE); + udelay(1); + + return 0; +} + +/*****************************************************************************/ +/* AD1843 access */ + +/* + * AD1843 bitfield definitions. All are named as in the AD1843 data + * sheet, with ad1843_ prepended and individual bit numbers removed. + * + * E.g., bits LSS0 through LSS2 become ad1843_LSS. + * + * Only the bitfields we need are defined. + */ + +typedef struct ad1843_bitfield { + char reg; + char lo_bit; + char nbits; +} ad1843_bitfield_t; + +static const ad1843_bitfield_t + ad1843_PDNO = { 0, 14, 1 }, /* Converter Power-Down Flag */ + ad1843_INIT = { 0, 15, 1 }, /* Clock Initialization Flag */ + ad1843_RIG = { 2, 0, 4 }, /* Right ADC Input Gain */ + ad1843_RMGE = { 2, 4, 1 }, /* Right ADC Mic Gain Enable */ + ad1843_RSS = { 2, 5, 3 }, /* Right ADC Source Select */ + ad1843_LIG = { 2, 8, 4 }, /* Left ADC Input Gain */ + ad1843_LMGE = { 2, 12, 1 }, /* Left ADC Mic Gain Enable */ + ad1843_LSS = { 2, 13, 3 }, /* Left ADC Source Select */ + ad1843_RX1M = { 4, 0, 5 }, /* Right Aux 1 Mix Gain/Atten */ + ad1843_RX1MM = { 4, 7, 1 }, /* Right Aux 1 Mix Mute */ + ad1843_LX1M = { 4, 8, 5 }, /* Left Aux 1 Mix Gain/Atten */ + ad1843_LX1MM = { 4, 15, 1 }, /* Left Aux 1 Mix Mute */ + ad1843_RX2M = { 5, 0, 5 }, /* Right Aux 2 Mix Gain/Atten */ + ad1843_RX2MM = { 5, 7, 1 }, /* Right Aux 2 Mix Mute */ + ad1843_LX2M = { 5, 8, 5 }, /* Left Aux 2 Mix Gain/Atten */ + ad1843_LX2MM = { 5, 15, 1 }, /* Left Aux 2 Mix Mute */ + ad1843_RMCM = { 7, 0, 5 }, /* Right Mic Mix Gain/Atten */ + ad1843_RMCMM = { 7, 7, 1 }, /* Right Mic Mix Mute */ + ad1843_LMCM = { 7, 8, 5 }, /* Left Mic Mix Gain/Atten */ + ad1843_LMCMM = { 7, 15, 1 }, /* Left Mic Mix Mute */ + ad1843_HPOS = { 8, 4, 1 }, /* Headphone Output Voltage Swing */ + ad1843_HPOM = { 8, 5, 1 }, /* Headphone Output Mute */ + ad1843_RDA1G = { 9, 0, 6 }, /* Right DAC1 Analog/Digital Gain */ + ad1843_RDA1GM = { 9, 7, 1 }, /* Right DAC1 Analog Mute */ + ad1843_LDA1G = { 9, 8, 6 }, /* Left DAC1 Analog/Digital Gain */ + ad1843_LDA1GM = { 9, 15, 1 }, /* Left DAC1 Analog Mute */ + ad1843_RDA1AM = { 11, 7, 1 }, /* Right DAC1 Digital Mute */ + ad1843_LDA1AM = { 11, 15, 1 }, /* Left DAC1 Digital Mute */ + ad1843_ADLC = { 15, 0, 2 }, /* ADC Left Sample Rate Source */ + ad1843_ADRC = { 15, 2, 2 }, /* ADC Right Sample Rate Source */ + ad1843_DA1C = { 15, 8, 2 }, /* DAC1 Sample Rate Source */ + ad1843_C1C = { 17, 0, 16 }, /* Clock 1 Sample Rate Select */ + ad1843_C2C = { 20, 0, 16 }, /* Clock 1 Sample Rate Select */ + ad1843_DAADL = { 25, 4, 2 }, /* Digital ADC Left Source Select */ + ad1843_DAADR = { 25, 6, 2 }, /* Digital ADC Right Source Select */ + ad1843_DRSFLT = { 25, 15, 1 }, /* Digital Reampler Filter Mode */ + ad1843_ADLF = { 26, 0, 2 }, /* ADC Left Channel Data Format */ + ad1843_ADRF = { 26, 2, 2 }, /* ADC Right Channel Data Format */ + ad1843_ADTLK = { 26, 4, 1 }, /* ADC Transmit Lock Mode Select */ + ad1843_SCF = { 26, 7, 1 }, /* SCLK Frequency Select */ + ad1843_DA1F = { 26, 8, 2 }, /* DAC1 Data Format Select */ + ad1843_DA1SM = { 26, 14, 1 }, /* DAC1 Stereo/Mono Mode Select */ + ad1843_ADLEN = { 27, 0, 1 }, /* ADC Left Channel Enable */ + ad1843_ADREN = { 27, 1, 1 }, /* ADC Right Channel Enable */ + ad1843_AAMEN = { 27, 4, 1 }, /* Analog to Analog Mix Enable */ + ad1843_ANAEN = { 27, 7, 1 }, /* Analog Channel Enable */ + ad1843_DA1EN = { 27, 8, 1 }, /* DAC1 Enable */ + ad1843_DA2EN = { 27, 9, 1 }, /* DAC2 Enable */ + ad1843_C1EN = { 28, 11, 1 }, /* Clock Generator 1 Enable */ + ad1843_C2EN = { 28, 12, 1 }, /* Clock Generator 2 Enable */ + ad1843_PDNI = { 28, 15, 1 }; /* Converter Power Down */ + +/* + * The various registers of the AD1843 use three different formats for + * specifying gain. The ad1843_gain structure parameterizes the + * formats. + */ + +typedef struct ad1843_gain { + + int negative; /* nonzero if gain is negative. */ + const ad1843_bitfield_t *lfield; + const ad1843_bitfield_t *rfield; + +} ad1843_gain_t; + +static const ad1843_gain_t ad1843_gain_RECLEV + = { 0, &ad1843_LIG, &ad1843_RIG }; +static const ad1843_gain_t ad1843_gain_LINE + = { 1, &ad1843_LX1M, &ad1843_RX1M }; +static const ad1843_gain_t ad1843_gain_CD + = { 1, &ad1843_LX2M, &ad1843_RX2M }; +static const ad1843_gain_t ad1843_gain_MIC + = { 1, &ad1843_LMCM, &ad1843_RMCM }; +static const ad1843_gain_t ad1843_gain_PCM + = { 1, &ad1843_LDA1G, &ad1843_RDA1G }; + +/* read the current value of an AD1843 bitfield. */ + +static int ad1843_read_bits(lithium_t *lith, const ad1843_bitfield_t *field) +{ + int w = li_read_ad1843_reg(lith, field->reg); + int val = w >> field->lo_bit & ((1 << field->nbits) - 1); + + DBGXV("ad1843_read_bits(lith=0x%p, field->{%d %d %d}) returns 0x%x\n", + lith, field->reg, field->lo_bit, field->nbits, val); + + return val; +} + +/* + * write a new value to an AD1843 bitfield and return the old value. + */ + +static int ad1843_write_bits(lithium_t *lith, + const ad1843_bitfield_t *field, + int newval) +{ + int w = li_read_ad1843_reg(lith, field->reg); + int mask = ((1 << field->nbits) - 1) << field->lo_bit; + int oldval = (w & mask) >> field->lo_bit; + int newbits = (newval << field->lo_bit) & mask; + w = (w & ~mask) | newbits; + (void) li_write_ad1843_reg(lith, field->reg, w); + + DBGXV("ad1843_write_bits(lith=0x%p, field->{%d %d %d}, val=0x%x) " + "returns 0x%x\n", + lith, field->reg, field->lo_bit, field->nbits, newval, + oldval); + + return oldval; +} + +/* + * ad1843_read_multi reads multiple bitfields from the same AD1843 + * register. It uses a single read cycle to do it. (Reading the + * ad1843 requires 256 bit times at 12.288 MHz, or nearly 20 + * microseconds.) + * + * Called ike this. + * + * ad1843_read_multi(lith, nfields, + * &ad1843_FIELD1, &val1, + * &ad1843_FIELD2, &val2, ...); + */ + +static void ad1843_read_multi(lithium_t *lith, int argcount, ...) +{ + va_list ap; + const ad1843_bitfield_t *fp; + int w = 0, mask, *value, reg = -1; + + va_start(ap, argcount); + while (--argcount >= 0) { + fp = va_arg(ap, const ad1843_bitfield_t *); + value = va_arg(ap, int *); + if (reg == -1) { + reg = fp->reg; + w = li_read_ad1843_reg(lith, reg); + } + ASSERT(reg == fp->reg); + mask = (1 << fp->nbits) - 1; + *value = w >> fp->lo_bit & mask; + } + va_end(ap); +} + +/* + * ad1843_write_multi stores multiple bitfields into the same AD1843 + * register. It uses one read and one write cycle to do it. + * + * Called like this. + * + * ad1843_write_multi(lith, nfields, + * &ad1843_FIELD1, val1, + * &ad1843_FIELF2, val2, ...); + */ + +static void ad1843_write_multi(lithium_t *lith, int argcount, ...) +{ + va_list ap; + int reg; + const ad1843_bitfield_t *fp; + int value; + int w, m, mask, bits; + + mask = 0; + bits = 0; + reg = -1; + + va_start(ap, argcount); + while (--argcount >= 0) { + fp = va_arg(ap, const ad1843_bitfield_t *); + value = va_arg(ap, int); + if (reg == -1) + reg = fp->reg; + ASSERT(fp->reg == reg); + m = ((1 << fp->nbits) - 1) << fp->lo_bit; + mask |= m; + bits |= (value << fp->lo_bit) & m; + } + va_end(ap); + ASSERT(!(bits & ~mask)); + if (~mask & 0xFFFF) + w = li_read_ad1843_reg(lith, reg); + else + w = 0; + w = (w & ~mask) | bits; + (void) li_write_ad1843_reg(lith, reg, w); +} + +/* + * ad1843_get_gain reads the specified register and extracts the gain value + * using the supplied gain type. It returns the gain in OSS format. + */ + +static int ad1843_get_gain(lithium_t *lith, const ad1843_gain_t *gp) +{ + int lg, rg; + unsigned short mask = (1 << gp->lfield->nbits) - 1; + + ad1843_read_multi(lith, 2, gp->lfield, &lg, gp->rfield, &rg); + if (gp->negative) { + lg = mask - lg; + rg = mask - rg; + } + lg = (lg * 100 + (mask >> 1)) / mask; + rg = (rg * 100 + (mask >> 1)) / mask; + return lg << 0 | rg << 8; +} + +/* + * Set an audio channel's gain. Converts from OSS format to AD1843's + * format. + * + * Returns the new gain, which may be lower than the old gain. + */ + +static int ad1843_set_gain(lithium_t *lith, + const ad1843_gain_t *gp, + int newval) +{ + unsigned short mask = (1 << gp->lfield->nbits) - 1; + + int lg = newval >> 0 & 0xFF; + int rg = newval >> 8; + if (lg < 0 || lg > 100 || rg < 0 || rg > 100) + return -EINVAL; + lg = (lg * mask + (mask >> 1)) / 100; + rg = (rg * mask + (mask >> 1)) / 100; + if (gp->negative) { + lg = mask - lg; + rg = mask - rg; + } + ad1843_write_multi(lith, 2, gp->lfield, lg, gp->rfield, rg); + return ad1843_get_gain(lith, gp); +} + +/* Returns the current recording source, in OSS format. */ + +static int ad1843_get_recsrc(lithium_t *lith) +{ + int ls = ad1843_read_bits(lith, &ad1843_LSS); + + switch (ls) { + case 1: + return SOUND_MASK_MIC; + case 2: + return SOUND_MASK_LINE; + case 3: + return SOUND_MASK_CD; + case 6: + return SOUND_MASK_PCM; + default: + ASSERT(0); + return -1; + } +} + +/* + * Enable/disable digital resample mode in the AD1843. + * + * The AD1843 requires that ADL, ADR, DA1 and DA2 be powered down + * while switching modes. So we save DA1's state (DA2's state is not + * interesting), power them down, switch into/out of resample mode, + * power them up, and restore state. + * + * This will cause audible glitches if D/A or A/D is going on, so the + * driver disallows that (in mixer_write_ioctl()). + * + * The open question is, is this worth doing? I'm leaving it in, + * because it's written, but... + */ + +static void ad1843_set_resample_mode(lithium_t *lith, int onoff) +{ + /* Save DA1 mute and gain (addr 9 is DA1 analog gain/attenuation) */ + int save_da1 = li_read_ad1843_reg(lith, 9); + + /* Power down A/D and D/A. */ + ad1843_write_multi(lith, 4, + &ad1843_DA1EN, 0, + &ad1843_DA2EN, 0, + &ad1843_ADLEN, 0, + &ad1843_ADREN, 0); + + /* Switch mode */ + ASSERT(onoff == 0 || onoff == 1); + ad1843_write_bits(lith, &ad1843_DRSFLT, onoff); + + /* Power up A/D and D/A. */ + ad1843_write_multi(lith, 3, + &ad1843_DA1EN, 1, + &ad1843_ADLEN, 1, + &ad1843_ADREN, 1); + + /* Restore DA1 mute and gain. */ + li_write_ad1843_reg(lith, 9, save_da1); +} + +/* + * Set recording source. Arg newsrc specifies an OSS channel mask. + * + * The complication is that when we switch into/out of loopback mode + * (i.e., src = SOUND_MASK_PCM), we change the AD1843 into/out of + * digital resampling mode. + * + * Returns newsrc on success, -errno on failure. + */ + +static int ad1843_set_recsrc(lithium_t *lith, int newsrc) +{ + int bits; + int oldbits; + + switch (newsrc) { + case SOUND_MASK_PCM: + bits = 6; + break; + + case SOUND_MASK_MIC: + bits = 1; + break; + + case SOUND_MASK_LINE: + bits = 2; + break; + + case SOUND_MASK_CD: + bits = 3; + break; + + default: + return -EINVAL; + } + oldbits = ad1843_read_bits(lith, &ad1843_LSS); + if (newsrc == SOUND_MASK_PCM && oldbits != 6) { + DBGP("enabling digital resample mode\n"); + ad1843_set_resample_mode(lith, 1); + ad1843_write_multi(lith, 2, + &ad1843_DAADL, 2, + &ad1843_DAADR, 2); + } else if (newsrc != SOUND_MASK_PCM && oldbits == 6) { + DBGP("disabling digital resample mode\n"); + ad1843_set_resample_mode(lith, 0); + ad1843_write_multi(lith, 2, + &ad1843_DAADL, 0, + &ad1843_DAADR, 0); + } + ad1843_write_multi(lith, 2, &ad1843_LSS, bits, &ad1843_RSS, bits); + return newsrc; +} + +/* + * Return current output sources, in OSS format. + */ + +static int ad1843_get_outsrc(lithium_t *lith) +{ + int pcm, line, mic, cd; + + pcm = ad1843_read_bits(lith, &ad1843_LDA1GM) ? 0 : SOUND_MASK_PCM; + line = ad1843_read_bits(lith, &ad1843_LX1MM) ? 0 : SOUND_MASK_LINE; + cd = ad1843_read_bits(lith, &ad1843_LX2MM) ? 0 : SOUND_MASK_CD; + mic = ad1843_read_bits(lith, &ad1843_LMCMM) ? 0 : SOUND_MASK_MIC; + + return pcm | line | cd | mic; +} + +/* + * Set output sources. Arg is a mask of active sources in OSS format. + * + * Returns source mask on success, -errno on failure. + */ + +static int ad1843_set_outsrc(lithium_t *lith, int mask) +{ + int pcm, line, mic, cd; + + if (mask & ~(SOUND_MASK_PCM | SOUND_MASK_LINE | + SOUND_MASK_CD | SOUND_MASK_MIC)) + return -EINVAL; + pcm = (mask & SOUND_MASK_PCM) ? 0 : 1; + line = (mask & SOUND_MASK_LINE) ? 0 : 1; + mic = (mask & SOUND_MASK_MIC) ? 0 : 1; + cd = (mask & SOUND_MASK_CD) ? 0 : 1; + + ad1843_write_multi(lith, 2, &ad1843_LDA1GM, pcm, &ad1843_RDA1GM, pcm); + ad1843_write_multi(lith, 2, &ad1843_LX1MM, line, &ad1843_RX1MM, line); + ad1843_write_multi(lith, 2, &ad1843_LX2MM, cd, &ad1843_RX2MM, cd); + ad1843_write_multi(lith, 2, &ad1843_LMCMM, mic, &ad1843_RMCMM, mic); + + return mask; +} + +/* Setup ad1843 for D/A conversion. */ + +static void ad1843_setup_dac(lithium_t *lith, + int framerate, + int fmt, + int channels) +{ + int ad_fmt = 0, ad_mode = 0; + + DBGEV("(lith=0x%p, framerate=%d, fmt=%d, channels=%d)\n", + lith, framerate, fmt, channels); + + switch (fmt) { + case AFMT_S8: ad_fmt = 1; break; + case AFMT_U8: ad_fmt = 1; break; + case AFMT_S16_LE: ad_fmt = 1; break; + case AFMT_MU_LAW: ad_fmt = 2; break; + case AFMT_A_LAW: ad_fmt = 3; break; + default: ASSERT(0); + } + + switch (channels) { + case 2: ad_mode = 0; break; + case 1: ad_mode = 1; break; + default: ASSERT(0); + } + + DBGPV("ad_mode = %d, ad_fmt = %d\n", ad_mode, ad_fmt); + ASSERT(framerate >= 4000 && framerate <= 49000); + ad1843_write_bits(lith, &ad1843_C1C, framerate); + ad1843_write_multi(lith, 2, + &ad1843_DA1SM, ad_mode, &ad1843_DA1F, ad_fmt); +} + +static void ad1843_shutdown_dac(lithium_t *lith) +{ + ad1843_write_bits(lith, &ad1843_DA1F, 1); +} + +static void ad1843_setup_adc(lithium_t *lith, int framerate, int fmt, int channels) +{ + int da_fmt = 0; + + DBGEV("(lith=0x%p, framerate=%d, fmt=%d, channels=%d)\n", + lith, framerate, fmt, channels); + + switch (fmt) { + case AFMT_S8: da_fmt = 1; break; + case AFMT_U8: da_fmt = 1; break; + case AFMT_S16_LE: da_fmt = 1; break; + case AFMT_MU_LAW: da_fmt = 2; break; + case AFMT_A_LAW: da_fmt = 3; break; + default: ASSERT(0); + } + + DBGPV("da_fmt = %d\n", da_fmt); + ASSERT(framerate >= 4000 && framerate <= 49000); + ad1843_write_bits(lith, &ad1843_C2C, framerate); + ad1843_write_multi(lith, 2, + &ad1843_ADLF, da_fmt, &ad1843_ADRF, da_fmt); +} + +static void ad1843_shutdown_adc(lithium_t *lith) +{ + /* nothing to do */ +} + +/* + * Fully initialize the ad1843. As described in the AD1843 data + * sheet, section "START-UP SEQUENCE". The numbered comments are + * subsection headings from the data sheet. See the data sheet, pages + * 52-54, for more info. + * + * return 0 on success, -errno on failure. */ + +static int __init ad1843_init(lithium_t *lith) +{ + unsigned long later; + int err; + + err = li_init(lith); + if (err) + return err; + + if (ad1843_read_bits(lith, &ad1843_INIT) != 0) { + printk(KERN_ERR "vwsnd sound: AD1843 won't initialize\n"); + return -EIO; + } + + ad1843_write_bits(lith, &ad1843_SCF, 1); + + /* 4. Put the conversion resources into standby. */ + + ad1843_write_bits(lith, &ad1843_PDNI, 0); + later = jiffies + HZ / 2; /* roughly half a second */ + DBGDO(shut_up++); + while (ad1843_read_bits(lith, &ad1843_PDNO)) { + if (time_after(jiffies, later)) { + printk(KERN_ERR + "vwsnd audio: AD1843 won't power up\n"); + return -EIO; + } + schedule(); + } + DBGDO(shut_up--); + + /* 5. Power up the clock generators and enable clock output pins. */ + + ad1843_write_multi(lith, 2, &ad1843_C1EN, 1, &ad1843_C2EN, 1); + + /* 6. Configure conversion resources while they are in standby. */ + + /* DAC1 uses clock 1 as source, ADC uses clock 2. Always. */ + + ad1843_write_multi(lith, 3, + &ad1843_DA1C, 1, + &ad1843_ADLC, 2, + &ad1843_ADRC, 2); + + /* 7. Enable conversion resources. */ + + ad1843_write_bits(lith, &ad1843_ADTLK, 1); + ad1843_write_multi(lith, 5, + &ad1843_ANAEN, 1, + &ad1843_AAMEN, 1, + &ad1843_DA1EN, 1, + &ad1843_ADLEN, 1, + &ad1843_ADREN, 1); + + /* 8. Configure conversion resources while they are enabled. */ + + ad1843_write_bits(lith, &ad1843_DA1C, 1); + + /* Unmute all channels. */ + + ad1843_set_outsrc(lith, + (SOUND_MASK_PCM | SOUND_MASK_LINE | + SOUND_MASK_MIC | SOUND_MASK_CD)); + ad1843_write_multi(lith, 2, &ad1843_LDA1AM, 0, &ad1843_RDA1AM, 0); + + /* Set default recording source to Line In and set + * mic gain to +20 dB. + */ + + ad1843_set_recsrc(lith, SOUND_MASK_LINE); + ad1843_write_multi(lith, 2, &ad1843_LMGE, 1, &ad1843_RMGE, 1); + + /* Set Speaker Out level to +/- 4V and unmute it. */ + + ad1843_write_multi(lith, 2, &ad1843_HPOS, 1, &ad1843_HPOM, 0); + + return 0; +} + +/*****************************************************************************/ +/* PCM I/O */ + +#define READ_INTR_MASK (LI_INTR_COMM1_TRIG | LI_INTR_COMM1_OVERFLOW) +#define WRITE_INTR_MASK (LI_INTR_COMM2_TRIG | LI_INTR_COMM2_UNDERFLOW) + +typedef enum vwsnd_port_swstate { /* software state */ + SW_OFF, + SW_INITIAL, + SW_RUN, + SW_DRAIN, +} vwsnd_port_swstate_t; + +typedef enum vwsnd_port_hwstate { /* hardware state */ + HW_STOPPED, + HW_RUNNING, +} vwsnd_port_hwstate_t; + +/* + * These flags are read by ISR, but only written at baseline. + */ + +typedef enum vwsnd_port_flags { + DISABLED = 1 << 0, + ERFLOWN = 1 << 1, /* overflown or underflown */ + HW_BUSY = 1 << 2, +} vwsnd_port_flags_t; + +/* + * vwsnd_port is the per-port data structure. Each device has two + * ports, one for input and one for output. + * + * Locking: + * + * port->lock protects: hwstate, flags, swb_[iu]_avail. + * + * devc->io_mutex protects: swstate, sw_*, swb_[iu]_idx. + * + * everything else is only written by open/release or + * pcm_{setup,shutdown}(), which are serialized by a + * combination of devc->open_mutex and devc->io_mutex. + */ + +typedef struct vwsnd_port { + + spinlock_t lock; + wait_queue_head_t queue; + vwsnd_port_swstate_t swstate; + vwsnd_port_hwstate_t hwstate; + vwsnd_port_flags_t flags; + + int sw_channels; + int sw_samplefmt; + int sw_framerate; + int sample_size; + int frame_size; + unsigned int zero_word; /* zero for the sample format */ + + int sw_fragshift; + int sw_fragcount; + int sw_subdivshift; + + unsigned int hw_fragshift; + unsigned int hw_fragsize; + unsigned int hw_fragcount; + + int hwbuf_size; + unsigned long hwbuf_paddr; + unsigned long hwbuf_vaddr; + void * hwbuf; /* hwbuf == hwbuf_vaddr */ + int hwbuf_max; /* max bytes to preload */ + + void * swbuf; + unsigned int swbuf_size; /* size in bytes */ + unsigned int swb_u_idx; /* index of next user byte */ + unsigned int swb_i_idx; /* index of next intr byte */ + unsigned int swb_u_avail; /* # bytes avail to user */ + unsigned int swb_i_avail; /* # bytes avail to intr */ + + dma_chan_t chan; + + /* Accounting */ + + int byte_count; + int frag_count; + int MSC_offset; + +} vwsnd_port_t; + +/* vwsnd_dev is the per-device data structure. */ + +typedef struct vwsnd_dev { + struct vwsnd_dev *next_dev; + int audio_minor; /* minor number of audio device */ + int mixer_minor; /* minor number of mixer device */ + + struct mutex open_mutex; + struct mutex io_mutex; + struct mutex mix_mutex; + fmode_t open_mode; + wait_queue_head_t open_wait; + + lithium_t lith; + + vwsnd_port_t rport; + vwsnd_port_t wport; +} vwsnd_dev_t; + +static vwsnd_dev_t *vwsnd_dev_list; /* linked list of all devices */ + +static atomic_t vwsnd_use_count = ATOMIC_INIT(0); + +# define INC_USE_COUNT (atomic_inc(&vwsnd_use_count)) +# define DEC_USE_COUNT (atomic_dec(&vwsnd_use_count)) +# define IN_USE (atomic_read(&vwsnd_use_count) != 0) + +/* + * Lithium can only DMA multiples of 32 bytes. Its DMA buffer may + * be up to 8 Kb. This driver always uses 8 Kb. + * + * Memory bug workaround -- I'm not sure what's going on here, but + * somehow pcm_copy_out() was triggering segv's going on to the next + * page of the hw buffer. So, I make the hw buffer one size bigger + * than we actually use. That way, the following page is allocated + * and mapped, and no error. I suspect that something is broken + * in Cobalt, but haven't really investigated. HBO is the actual + * size of the buffer, and HWBUF_ORDER is what we allocate. + */ + +#define HWBUF_SHIFT 13 +#define HWBUF_SIZE (1 << HWBUF_SHIFT) +# define HBO (HWBUF_SHIFT > PAGE_SHIFT ? HWBUF_SHIFT - PAGE_SHIFT : 0) +# define HWBUF_ORDER (HBO + 1) /* next size bigger */ +#define MIN_SPEED 4000 +#define MAX_SPEED 49000 + +#define MIN_FRAGSHIFT (DMACHUNK_SHIFT + 1) +#define MAX_FRAGSHIFT (PAGE_SHIFT) +#define MIN_FRAGSIZE (1 << MIN_FRAGSHIFT) +#define MAX_FRAGSIZE (1 << MAX_FRAGSHIFT) +#define MIN_FRAGCOUNT(fragsize) 3 +#define MAX_FRAGCOUNT(fragsize) (32 * PAGE_SIZE / (fragsize)) +#define DEFAULT_FRAGSHIFT 12 +#define DEFAULT_FRAGCOUNT 16 +#define DEFAULT_SUBDIVSHIFT 0 + +/* + * The software buffer (swbuf) is a ring buffer shared between user + * level and interrupt level. Each level owns some of the bytes in + * the buffer, and may give bytes away by calling swb_inc_{u,i}(). + * User level calls _u for user, and interrupt level calls _i for + * interrupt. + * + * port->swb_{u,i}_avail is the number of bytes available to that level. + * + * port->swb_{u,i}_idx is the index of the first available byte in the + * buffer. + * + * Each level calls swb_inc_{u,i}() to atomically increment its index, + * recalculate the number of bytes available for both sides, and + * return the number of bytes available. Since each side can only + * give away bytes, the other side can only increase the number of + * bytes available to this side. Each side updates its own index + * variable, swb_{u,i}_idx, so no lock is needed to read it. + * + * To query the number of bytes available, call swb_inc_{u,i} with an + * increment of zero. + */ + +static __inline__ unsigned int __swb_inc_u(vwsnd_port_t *port, int inc) +{ + if (inc) { + port->swb_u_idx += inc; + port->swb_u_idx %= port->swbuf_size; + port->swb_u_avail -= inc; + port->swb_i_avail += inc; + } + return port->swb_u_avail; +} + +static __inline__ unsigned int swb_inc_u(vwsnd_port_t *port, int inc) +{ + unsigned long flags; + unsigned int ret; + + spin_lock_irqsave(&port->lock, flags); + { + ret = __swb_inc_u(port, inc); + } + spin_unlock_irqrestore(&port->lock, flags); + return ret; +} + +static __inline__ unsigned int __swb_inc_i(vwsnd_port_t *port, int inc) +{ + if (inc) { + port->swb_i_idx += inc; + port->swb_i_idx %= port->swbuf_size; + port->swb_i_avail -= inc; + port->swb_u_avail += inc; + } + return port->swb_i_avail; +} + +static __inline__ unsigned int swb_inc_i(vwsnd_port_t *port, int inc) +{ + unsigned long flags; + unsigned int ret; + + spin_lock_irqsave(&port->lock, flags); + { + ret = __swb_inc_i(port, inc); + } + spin_unlock_irqrestore(&port->lock, flags); + return ret; +} + +/* + * pcm_setup - this routine initializes all port state after + * mode-setting ioctls have been done, but before the first I/O is + * done. + * + * Locking: called with devc->io_mutex held. + * + * Returns 0 on success, -errno on failure. + */ + +static int pcm_setup(vwsnd_dev_t *devc, + vwsnd_port_t *rport, + vwsnd_port_t *wport) +{ + vwsnd_port_t *aport = rport ? rport : wport; + int sample_size; + unsigned int zero_word; + + DBGEV("(devc=0x%p, rport=0x%p, wport=0x%p)\n", devc, rport, wport); + + ASSERT(aport != NULL); + if (aport->swbuf != NULL) + return 0; + switch (aport->sw_samplefmt) { + case AFMT_MU_LAW: + sample_size = 1; + zero_word = 0xFFFFFFFF ^ 0x80808080; + break; + + case AFMT_A_LAW: + sample_size = 1; + zero_word = 0xD5D5D5D5 ^ 0x80808080; + break; + + case AFMT_U8: + sample_size = 1; + zero_word = 0x80808080; + break; + + case AFMT_S8: + sample_size = 1; + zero_word = 0x00000000; + break; + + case AFMT_S16_LE: + sample_size = 2; + zero_word = 0x00000000; + break; + + default: + sample_size = 0; /* prevent compiler warning */ + zero_word = 0; + ASSERT(0); + } + aport->sample_size = sample_size; + aport->zero_word = zero_word; + aport->frame_size = aport->sw_channels * aport->sample_size; + aport->hw_fragshift = aport->sw_fragshift - aport->sw_subdivshift; + aport->hw_fragsize = 1 << aport->hw_fragshift; + aport->hw_fragcount = aport->sw_fragcount << aport->sw_subdivshift; + ASSERT(aport->hw_fragsize >= MIN_FRAGSIZE); + ASSERT(aport->hw_fragsize <= MAX_FRAGSIZE); + ASSERT(aport->hw_fragcount >= MIN_FRAGCOUNT(aport->hw_fragsize)); + ASSERT(aport->hw_fragcount <= MAX_FRAGCOUNT(aport->hw_fragsize)); + if (rport) { + int hwfrags, swfrags; + rport->hwbuf_max = aport->hwbuf_size - DMACHUNK_SIZE; + hwfrags = rport->hwbuf_max >> aport->hw_fragshift; + swfrags = aport->hw_fragcount - hwfrags; + if (swfrags < 2) + swfrags = 2; + rport->swbuf_size = swfrags * aport->hw_fragsize; + DBGPV("hwfrags = %d, swfrags = %d\n", hwfrags, swfrags); + DBGPV("read hwbuf_max = %d, swbuf_size = %d\n", + rport->hwbuf_max, rport->swbuf_size); + } + if (wport) { + int hwfrags, swfrags; + int total_bytes = aport->hw_fragcount * aport->hw_fragsize; + wport->hwbuf_max = aport->hwbuf_size - DMACHUNK_SIZE; + if (wport->hwbuf_max > total_bytes) + wport->hwbuf_max = total_bytes; + hwfrags = wport->hwbuf_max >> aport->hw_fragshift; + DBGPV("hwfrags = %d\n", hwfrags); + swfrags = aport->hw_fragcount - hwfrags; + if (swfrags < 2) + swfrags = 2; + wport->swbuf_size = swfrags * aport->hw_fragsize; + DBGPV("hwfrags = %d, swfrags = %d\n", hwfrags, swfrags); + DBGPV("write hwbuf_max = %d, swbuf_size = %d\n", + wport->hwbuf_max, wport->swbuf_size); + } + + aport->swb_u_idx = 0; + aport->swb_i_idx = 0; + aport->byte_count = 0; + + /* + * Is this a Cobalt bug? We need to make this buffer extend + * one page further than we actually use -- somehow memcpy + * causes an exceptoin otherwise. I suspect there's a bug in + * Cobalt (or somewhere) where it's generating a fault on a + * speculative load or something. Obviously, I haven't taken + * the time to track it down. + */ + + aport->swbuf = vmalloc(aport->swbuf_size + PAGE_SIZE); + if (!aport->swbuf) + return -ENOMEM; + if (rport && wport) { + ASSERT(aport == rport); + ASSERT(wport->swbuf == NULL); + /* One extra page - see comment above. */ + wport->swbuf = vmalloc(aport->swbuf_size + PAGE_SIZE); + if (!wport->swbuf) { + vfree(aport->swbuf); + aport->swbuf = NULL; + return -ENOMEM; + } + wport->sample_size = rport->sample_size; + wport->zero_word = rport->zero_word; + wport->frame_size = rport->frame_size; + wport->hw_fragshift = rport->hw_fragshift; + wport->hw_fragsize = rport->hw_fragsize; + wport->hw_fragcount = rport->hw_fragcount; + wport->swbuf_size = rport->swbuf_size; + wport->hwbuf_max = rport->hwbuf_max; + wport->swb_u_idx = rport->swb_u_idx; + wport->swb_i_idx = rport->swb_i_idx; + wport->byte_count = rport->byte_count; + } + if (rport) { + rport->swb_u_avail = 0; + rport->swb_i_avail = rport->swbuf_size; + rport->swstate = SW_RUN; + li_setup_dma(&rport->chan, + &li_comm1, + &devc->lith, + rport->hwbuf_paddr, + HWBUF_SHIFT, + rport->hw_fragshift, + rport->sw_channels, + rport->sample_size); + ad1843_setup_adc(&devc->lith, + rport->sw_framerate, + rport->sw_samplefmt, + rport->sw_channels); + li_enable_interrupts(&devc->lith, READ_INTR_MASK); + if (!(rport->flags & DISABLED)) { + ustmsc_t ustmsc; + rport->hwstate = HW_RUNNING; + li_activate_dma(&rport->chan); + li_read_USTMSC(&rport->chan, &ustmsc); + rport->MSC_offset = ustmsc.msc; + } + } + if (wport) { + if (wport->hwbuf_max > wport->swbuf_size) + wport->hwbuf_max = wport->swbuf_size; + wport->flags &= ~ERFLOWN; + wport->swb_u_avail = wport->swbuf_size; + wport->swb_i_avail = 0; + wport->swstate = SW_RUN; + li_setup_dma(&wport->chan, + &li_comm2, + &devc->lith, + wport->hwbuf_paddr, + HWBUF_SHIFT, + wport->hw_fragshift, + wport->sw_channels, + wport->sample_size); + ad1843_setup_dac(&devc->lith, + wport->sw_framerate, + wport->sw_samplefmt, + wport->sw_channels); + li_enable_interrupts(&devc->lith, WRITE_INTR_MASK); + } + DBGRV(); + return 0; +} + +/* + * pcm_shutdown_port - shut down one port (direction) for PCM I/O. + * Only called from pcm_shutdown. + */ + +static void pcm_shutdown_port(vwsnd_dev_t *devc, + vwsnd_port_t *aport, + unsigned int mask) +{ + unsigned long flags; + vwsnd_port_hwstate_t hwstate; + DECLARE_WAITQUEUE(wait, current); + + aport->swstate = SW_INITIAL; + add_wait_queue(&aport->queue, &wait); + while (1) { + set_current_state(TASK_UNINTERRUPTIBLE); + spin_lock_irqsave(&aport->lock, flags); + { + hwstate = aport->hwstate; + } + spin_unlock_irqrestore(&aport->lock, flags); + if (hwstate == HW_STOPPED) + break; + schedule(); + } + current->state = TASK_RUNNING; + remove_wait_queue(&aport->queue, &wait); + li_disable_interrupts(&devc->lith, mask); + if (aport == &devc->rport) + ad1843_shutdown_adc(&devc->lith); + else /* aport == &devc->wport) */ + ad1843_shutdown_dac(&devc->lith); + li_shutdown_dma(&aport->chan); + vfree(aport->swbuf); + aport->swbuf = NULL; + aport->byte_count = 0; +} + +/* + * pcm_shutdown undoes what pcm_setup did. + * Also sets the ports' swstate to newstate. + */ + +static void pcm_shutdown(vwsnd_dev_t *devc, + vwsnd_port_t *rport, + vwsnd_port_t *wport) +{ + DBGEV("(devc=0x%p, rport=0x%p, wport=0x%p)\n", devc, rport, wport); + + if (rport && rport->swbuf) { + DBGPV("shutting down rport\n"); + pcm_shutdown_port(devc, rport, READ_INTR_MASK); + } + if (wport && wport->swbuf) { + DBGPV("shutting down wport\n"); + pcm_shutdown_port(devc, wport, WRITE_INTR_MASK); + } + DBGRV(); +} + +static void pcm_copy_in(vwsnd_port_t *rport, int swidx, int hwidx, int nb) +{ + char *src = rport->hwbuf + hwidx; + char *dst = rport->swbuf + swidx; + int fmt = rport->sw_samplefmt; + + DBGPV("swidx = %d, hwidx = %d\n", swidx, hwidx); + ASSERT(rport->hwbuf != NULL); + ASSERT(rport->swbuf != NULL); + ASSERT(nb > 0 && (nb % 32) == 0); + ASSERT(swidx % 32 == 0 && hwidx % 32 == 0); + ASSERT(swidx >= 0 && swidx + nb <= rport->swbuf_size); + ASSERT(hwidx >= 0 && hwidx + nb <= rport->hwbuf_size); + + if (fmt == AFMT_MU_LAW || fmt == AFMT_A_LAW || fmt == AFMT_S8) { + + /* See Sample Format Notes above. */ + + char *end = src + nb; + while (src < end) + *dst++ = *src++ ^ 0x80; + } else + memcpy(dst, src, nb); +} + +static void pcm_copy_out(vwsnd_port_t *wport, int swidx, int hwidx, int nb) +{ + char *src = wport->swbuf + swidx; + char *dst = wport->hwbuf + hwidx; + int fmt = wport->sw_samplefmt; + + ASSERT(nb > 0 && (nb % 32) == 0); + ASSERT(wport->hwbuf != NULL); + ASSERT(wport->swbuf != NULL); + ASSERT(swidx % 32 == 0 && hwidx % 32 == 0); + ASSERT(swidx >= 0 && swidx + nb <= wport->swbuf_size); + ASSERT(hwidx >= 0 && hwidx + nb <= wport->hwbuf_size); + if (fmt == AFMT_MU_LAW || fmt == AFMT_A_LAW || fmt == AFMT_S8) { + + /* See Sample Format Notes above. */ + + char *end = src + nb; + while (src < end) + *dst++ = *src++ ^ 0x80; + } else + memcpy(dst, src, nb); +} + +/* + * pcm_output() is called both from baselevel and from interrupt level. + * This is where audio frames are copied into the hardware-accessible + * ring buffer. + * + * Locking note: The part of this routine that figures out what to do + * holds wport->lock. The longer part releases wport->lock, but sets + * wport->flags & HW_BUSY. Afterward, it reacquires wport->lock, and + * checks for more work to do. + * + * If another thread calls pcm_output() while HW_BUSY is set, it + * returns immediately, knowing that the thread that set HW_BUSY will + * look for more work to do before returning. + * + * This has the advantage that port->lock is held for several short + * periods instead of one long period. Also, when pcm_output is + * called from base level, it reenables interrupts. + */ + +static void pcm_output(vwsnd_dev_t *devc, int erflown, int nb) +{ + vwsnd_port_t *wport = &devc->wport; + const int hwmax = wport->hwbuf_max; + const int hwsize = wport->hwbuf_size; + const int swsize = wport->swbuf_size; + const int fragsize = wport->hw_fragsize; + unsigned long iflags; + + DBGEV("(devc=0x%p, erflown=%d, nb=%d)\n", devc, erflown, nb); + spin_lock_irqsave(&wport->lock, iflags); + if (erflown) + wport->flags |= ERFLOWN; + (void) __swb_inc_u(wport, nb); + if (wport->flags & HW_BUSY) { + spin_unlock_irqrestore(&wport->lock, iflags); + DBGPV("returning: HW BUSY\n"); + return; + } + if (wport->flags & DISABLED) { + spin_unlock_irqrestore(&wport->lock, iflags); + DBGPV("returning: DISABLED\n"); + return; + } + wport->flags |= HW_BUSY; + while (1) { + int swptr, hwptr, hw_avail, sw_avail, swidx; + vwsnd_port_hwstate_t hwstate = wport->hwstate; + vwsnd_port_swstate_t swstate = wport->swstate; + int hw_unavail; + ustmsc_t ustmsc; + + hwptr = li_read_hwptr(&wport->chan); + swptr = li_read_swptr(&wport->chan); + hw_unavail = (swptr - hwptr + hwsize) % hwsize; + hw_avail = (hwmax - hw_unavail) & -fragsize; + sw_avail = wport->swb_i_avail & -fragsize; + if (sw_avail && swstate == SW_RUN) { + if (wport->flags & ERFLOWN) { + wport->flags &= ~ERFLOWN; + } + } else if (swstate == SW_INITIAL || + swstate == SW_OFF || + (swstate == SW_DRAIN && + !sw_avail && + (wport->flags & ERFLOWN))) { + DBGP("stopping. hwstate = %d\n", hwstate); + if (hwstate != HW_STOPPED) { + li_deactivate_dma(&wport->chan); + wport->hwstate = HW_STOPPED; + } + wake_up(&wport->queue); + break; + } + if (!sw_avail || !hw_avail) + break; + spin_unlock_irqrestore(&wport->lock, iflags); + + /* + * We gave up the port lock, but we have the HW_BUSY flag. + * Proceed without accessing any nonlocal state. + * Do not exit the loop -- must check for more work. + */ + + swidx = wport->swb_i_idx; + nb = hw_avail; + if (nb > sw_avail) + nb = sw_avail; + if (nb > hwsize - swptr) + nb = hwsize - swptr; /* don't overflow hwbuf */ + if (nb > swsize - swidx) + nb = swsize - swidx; /* don't overflow swbuf */ + ASSERT(nb > 0); + if (nb % fragsize) { + DBGP("nb = %d, fragsize = %d\n", nb, fragsize); + DBGP("hw_avail = %d\n", hw_avail); + DBGP("sw_avail = %d\n", sw_avail); + DBGP("hwsize = %d, swptr = %d\n", hwsize, swptr); + DBGP("swsize = %d, swidx = %d\n", swsize, swidx); + } + ASSERT(!(nb % fragsize)); + DBGPV("copying swb[%d..%d] to hwb[%d..%d]\n", + swidx, swidx + nb, swptr, swptr + nb); + pcm_copy_out(wport, swidx, swptr, nb); + li_write_swptr(&wport->chan, (swptr + nb) % hwsize); + spin_lock_irqsave(&wport->lock, iflags); + if (hwstate == HW_STOPPED) { + DBGPV("starting\n"); + li_activate_dma(&wport->chan); + wport->hwstate = HW_RUNNING; + li_read_USTMSC(&wport->chan, &ustmsc); + ASSERT(wport->byte_count % wport->frame_size == 0); + wport->MSC_offset = ustmsc.msc - wport->byte_count / wport->frame_size; + } + __swb_inc_i(wport, nb); + wport->byte_count += nb; + wport->frag_count += nb / fragsize; + ASSERT(nb % fragsize == 0); + wake_up(&wport->queue); + } + wport->flags &= ~HW_BUSY; + spin_unlock_irqrestore(&wport->lock, iflags); + DBGRV(); +} + +/* + * pcm_input() is called both from baselevel and from interrupt level. + * This is where audio frames are copied out of the hardware-accessible + * ring buffer. + * + * Locking note: The part of this routine that figures out what to do + * holds rport->lock. The longer part releases rport->lock, but sets + * rport->flags & HW_BUSY. Afterward, it reacquires rport->lock, and + * checks for more work to do. + * + * If another thread calls pcm_input() while HW_BUSY is set, it + * returns immediately, knowing that the thread that set HW_BUSY will + * look for more work to do before returning. + * + * This has the advantage that port->lock is held for several short + * periods instead of one long period. Also, when pcm_input is + * called from base level, it reenables interrupts. + */ + +static void pcm_input(vwsnd_dev_t *devc, int erflown, int nb) +{ + vwsnd_port_t *rport = &devc->rport; + const int hwmax = rport->hwbuf_max; + const int hwsize = rport->hwbuf_size; + const int swsize = rport->swbuf_size; + const int fragsize = rport->hw_fragsize; + unsigned long iflags; + + DBGEV("(devc=0x%p, erflown=%d, nb=%d)\n", devc, erflown, nb); + + spin_lock_irqsave(&rport->lock, iflags); + if (erflown) + rport->flags |= ERFLOWN; + (void) __swb_inc_u(rport, nb); + if (rport->flags & HW_BUSY || !rport->swbuf) { + spin_unlock_irqrestore(&rport->lock, iflags); + DBGPV("returning: HW BUSY or !swbuf\n"); + return; + } + if (rport->flags & DISABLED) { + spin_unlock_irqrestore(&rport->lock, iflags); + DBGPV("returning: DISABLED\n"); + return; + } + rport->flags |= HW_BUSY; + while (1) { + int swptr, hwptr, hw_avail, sw_avail, swidx; + vwsnd_port_hwstate_t hwstate = rport->hwstate; + vwsnd_port_swstate_t swstate = rport->swstate; + + hwptr = li_read_hwptr(&rport->chan); + swptr = li_read_swptr(&rport->chan); + hw_avail = (hwptr - swptr + hwsize) % hwsize & -fragsize; + if (hw_avail > hwmax) + hw_avail = hwmax; + sw_avail = rport->swb_i_avail & -fragsize; + if (swstate != SW_RUN) { + DBGP("stopping. hwstate = %d\n", hwstate); + if (hwstate != HW_STOPPED) { + li_deactivate_dma(&rport->chan); + rport->hwstate = HW_STOPPED; + } + wake_up(&rport->queue); + break; + } + if (!sw_avail || !hw_avail) + break; + spin_unlock_irqrestore(&rport->lock, iflags); + + /* + * We gave up the port lock, but we have the HW_BUSY flag. + * Proceed without accessing any nonlocal state. + * Do not exit the loop -- must check for more work. + */ + + swidx = rport->swb_i_idx; + nb = hw_avail; + if (nb > sw_avail) + nb = sw_avail; + if (nb > hwsize - swptr) + nb = hwsize - swptr; /* don't overflow hwbuf */ + if (nb > swsize - swidx) + nb = swsize - swidx; /* don't overflow swbuf */ + ASSERT(nb > 0); + if (nb % fragsize) { + DBGP("nb = %d, fragsize = %d\n", nb, fragsize); + DBGP("hw_avail = %d\n", hw_avail); + DBGP("sw_avail = %d\n", sw_avail); + DBGP("hwsize = %d, swptr = %d\n", hwsize, swptr); + DBGP("swsize = %d, swidx = %d\n", swsize, swidx); + } + ASSERT(!(nb % fragsize)); + DBGPV("copying hwb[%d..%d] to swb[%d..%d]\n", + swptr, swptr + nb, swidx, swidx + nb); + pcm_copy_in(rport, swidx, swptr, nb); + li_write_swptr(&rport->chan, (swptr + nb) % hwsize); + spin_lock_irqsave(&rport->lock, iflags); + __swb_inc_i(rport, nb); + rport->byte_count += nb; + rport->frag_count += nb / fragsize; + ASSERT(nb % fragsize == 0); + wake_up(&rport->queue); + } + rport->flags &= ~HW_BUSY; + spin_unlock_irqrestore(&rport->lock, iflags); + DBGRV(); +} + +/* + * pcm_flush_frag() writes zero samples to fill the current fragment, + * then flushes it to the hardware. + * + * It is only meaningful to flush output, not input. + */ + +static void pcm_flush_frag(vwsnd_dev_t *devc) +{ + vwsnd_port_t *wport = &devc->wport; + + DBGPV("swstate = %d\n", wport->swstate); + if (wport->swstate == SW_RUN) { + int idx = wport->swb_u_idx; + int end = (idx + wport->hw_fragsize - 1) + >> wport->hw_fragshift + << wport->hw_fragshift; + int nb = end - idx; + DBGPV("clearing %d bytes\n", nb); + if (nb) + memset(wport->swbuf + idx, + (char) wport->zero_word, + nb); + wport->swstate = SW_DRAIN; + pcm_output(devc, 0, nb); + } + DBGRV(); +} + +/* + * Wait for output to drain. This sleeps uninterruptibly because + * there is nothing intelligent we can do if interrupted. This + * means the process will be delayed in responding to the signal. + */ + +static void pcm_write_sync(vwsnd_dev_t *devc) +{ + vwsnd_port_t *wport = &devc->wport; + DECLARE_WAITQUEUE(wait, current); + unsigned long flags; + vwsnd_port_hwstate_t hwstate; + + DBGEV("(devc=0x%p)\n", devc); + add_wait_queue(&wport->queue, &wait); + while (1) { + set_current_state(TASK_UNINTERRUPTIBLE); + spin_lock_irqsave(&wport->lock, flags); + { + hwstate = wport->hwstate; + } + spin_unlock_irqrestore(&wport->lock, flags); + if (hwstate == HW_STOPPED) + break; + schedule(); + } + current->state = TASK_RUNNING; + remove_wait_queue(&wport->queue, &wait); + DBGPV("swstate = %d, hwstate = %d\n", wport->swstate, wport->hwstate); + DBGRV(); +} + +/*****************************************************************************/ +/* audio driver */ + +/* + * seek on an audio device always fails. + */ + +static void vwsnd_audio_read_intr(vwsnd_dev_t *devc, unsigned int status) +{ + int overflown = status & LI_INTR_COMM1_OVERFLOW; + + if (status & READ_INTR_MASK) + pcm_input(devc, overflown, 0); +} + +static void vwsnd_audio_write_intr(vwsnd_dev_t *devc, unsigned int status) +{ + int underflown = status & LI_INTR_COMM2_UNDERFLOW; + + if (status & WRITE_INTR_MASK) + pcm_output(devc, underflown, 0); +} + +static irqreturn_t vwsnd_audio_intr(int irq, void *dev_id) +{ + vwsnd_dev_t *devc = dev_id; + unsigned int status; + + DBGEV("(irq=%d, dev_id=0x%p)\n", irq, dev_id); + + status = li_get_clear_intr_status(&devc->lith); + vwsnd_audio_read_intr(devc, status); + vwsnd_audio_write_intr(devc, status); + return IRQ_HANDLED; +} + +static ssize_t vwsnd_audio_do_read(struct file *file, + char *buffer, + size_t count, + loff_t *ppos) +{ + vwsnd_dev_t *devc = file->private_data; + vwsnd_port_t *rport = ((file->f_mode & FMODE_READ) ? + &devc->rport : NULL); + int ret, nb; + + DBGEV("(file=0x%p, buffer=0x%p, count=%d, ppos=0x%p)\n", + file, buffer, count, ppos); + + if (!rport) + return -EINVAL; + + if (rport->swbuf == NULL) { + vwsnd_port_t *wport = (file->f_mode & FMODE_WRITE) ? + &devc->wport : NULL; + ret = pcm_setup(devc, rport, wport); + if (ret < 0) + return ret; + } + + if (!access_ok(VERIFY_READ, buffer, count)) + return -EFAULT; + ret = 0; + while (count) { + DECLARE_WAITQUEUE(wait, current); + add_wait_queue(&rport->queue, &wait); + while ((nb = swb_inc_u(rport, 0)) == 0) { + DBGPV("blocking\n"); + set_current_state(TASK_INTERRUPTIBLE); + if (rport->flags & DISABLED || + file->f_flags & O_NONBLOCK) { + current->state = TASK_RUNNING; + remove_wait_queue(&rport->queue, &wait); + return ret ? ret : -EAGAIN; + } + schedule(); + if (signal_pending(current)) { + current->state = TASK_RUNNING; + remove_wait_queue(&rport->queue, &wait); + return ret ? ret : -ERESTARTSYS; + } + } + current->state = TASK_RUNNING; + remove_wait_queue(&rport->queue, &wait); + pcm_input(devc, 0, 0); + /* nb bytes are available in userbuf. */ + if (nb > count) + nb = count; + DBGPV("nb = %d\n", nb); + if (copy_to_user(buffer, rport->swbuf + rport->swb_u_idx, nb)) + return -EFAULT; + (void) swb_inc_u(rport, nb); + buffer += nb; + count -= nb; + ret += nb; + } + DBGPV("returning %d\n", ret); + return ret; +} + +static ssize_t vwsnd_audio_read(struct file *file, + char *buffer, + size_t count, + loff_t *ppos) +{ + vwsnd_dev_t *devc = file->private_data; + ssize_t ret; + + mutex_lock(&devc->io_mutex); + ret = vwsnd_audio_do_read(file, buffer, count, ppos); + mutex_unlock(&devc->io_mutex); + return ret; +} + +static ssize_t vwsnd_audio_do_write(struct file *file, + const char *buffer, + size_t count, + loff_t *ppos) +{ + vwsnd_dev_t *devc = file->private_data; + vwsnd_port_t *wport = ((file->f_mode & FMODE_WRITE) ? + &devc->wport : NULL); + int ret, nb; + + DBGEV("(file=0x%p, buffer=0x%p, count=%d, ppos=0x%p)\n", + file, buffer, count, ppos); + + if (!wport) + return -EINVAL; + + if (wport->swbuf == NULL) { + vwsnd_port_t *rport = (file->f_mode & FMODE_READ) ? + &devc->rport : NULL; + ret = pcm_setup(devc, rport, wport); + if (ret < 0) + return ret; + } + if (!access_ok(VERIFY_WRITE, buffer, count)) + return -EFAULT; + ret = 0; + while (count) { + DECLARE_WAITQUEUE(wait, current); + add_wait_queue(&wport->queue, &wait); + while ((nb = swb_inc_u(wport, 0)) == 0) { + set_current_state(TASK_INTERRUPTIBLE); + if (wport->flags & DISABLED || + file->f_flags & O_NONBLOCK) { + current->state = TASK_RUNNING; + remove_wait_queue(&wport->queue, &wait); + return ret ? ret : -EAGAIN; + } + schedule(); + if (signal_pending(current)) { + current->state = TASK_RUNNING; + remove_wait_queue(&wport->queue, &wait); + return ret ? ret : -ERESTARTSYS; + } + } + current->state = TASK_RUNNING; + remove_wait_queue(&wport->queue, &wait); + /* nb bytes are available in userbuf. */ + if (nb > count) + nb = count; + DBGPV("nb = %d\n", nb); + if (copy_from_user(wport->swbuf + wport->swb_u_idx, buffer, nb)) + return -EFAULT; + pcm_output(devc, 0, nb); + buffer += nb; + count -= nb; + ret += nb; + } + DBGPV("returning %d\n", ret); + return ret; +} + +static ssize_t vwsnd_audio_write(struct file *file, + const char *buffer, + size_t count, + loff_t *ppos) +{ + vwsnd_dev_t *devc = file->private_data; + ssize_t ret; + + mutex_lock(&devc->io_mutex); + ret = vwsnd_audio_do_write(file, buffer, count, ppos); + mutex_unlock(&devc->io_mutex); + return ret; +} + +/* No kernel lock - fine */ +static unsigned int vwsnd_audio_poll(struct file *file, + struct poll_table_struct *wait) +{ + vwsnd_dev_t *devc = (vwsnd_dev_t *) file->private_data; + vwsnd_port_t *rport = (file->f_mode & FMODE_READ) ? + &devc->rport : NULL; + vwsnd_port_t *wport = (file->f_mode & FMODE_WRITE) ? + &devc->wport : NULL; + unsigned int mask = 0; + + DBGEV("(file=0x%p, wait=0x%p)\n", file, wait); + + ASSERT(rport || wport); + if (rport) { + poll_wait(file, &rport->queue, wait); + if (swb_inc_u(rport, 0)) + mask |= (POLLIN | POLLRDNORM); + } + if (wport) { + poll_wait(file, &wport->queue, wait); + if (wport->swbuf == NULL || swb_inc_u(wport, 0)) + mask |= (POLLOUT | POLLWRNORM); + } + + DBGPV("returning 0x%x\n", mask); + return mask; +} + +static int vwsnd_audio_do_ioctl(struct file *file, + unsigned int cmd, + unsigned long arg) +{ + vwsnd_dev_t *devc = (vwsnd_dev_t *) file->private_data; + vwsnd_port_t *rport = (file->f_mode & FMODE_READ) ? + &devc->rport : NULL; + vwsnd_port_t *wport = (file->f_mode & FMODE_WRITE) ? + &devc->wport : NULL; + vwsnd_port_t *aport = rport ? rport : wport; + struct audio_buf_info buf_info; + struct count_info info; + unsigned long flags; + int ival; + + + DBGEV("(file=0x%p, cmd=0x%x, arg=0x%lx)\n", + file, cmd, arg); + switch (cmd) { + case OSS_GETVERSION: /* _SIOR ('M', 118, int) */ + DBGX("OSS_GETVERSION\n"); + ival = SOUND_VERSION; + return put_user(ival, (int *) arg); + + case SNDCTL_DSP_GETCAPS: /* _SIOR ('P',15, int) */ + DBGX("SNDCTL_DSP_GETCAPS\n"); + ival = DSP_CAP_DUPLEX | DSP_CAP_REALTIME | DSP_CAP_TRIGGER; + return put_user(ival, (int *) arg); + + case SNDCTL_DSP_GETFMTS: /* _SIOR ('P',11, int) */ + DBGX("SNDCTL_DSP_GETFMTS\n"); + ival = (AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | + AFMT_U8 | AFMT_S8); + return put_user(ival, (int *) arg); + break; + + case SOUND_PCM_READ_RATE: /* _SIOR ('P', 2, int) */ + DBGX("SOUND_PCM_READ_RATE\n"); + ival = aport->sw_framerate; + return put_user(ival, (int *) arg); + + case SOUND_PCM_READ_CHANNELS: /* _SIOR ('P', 6, int) */ + DBGX("SOUND_PCM_READ_CHANNELS\n"); + ival = aport->sw_channels; + return put_user(ival, (int *) arg); + + case SNDCTL_DSP_SPEED: /* _SIOWR('P', 2, int) */ + if (get_user(ival, (int *) arg)) + return -EFAULT; + DBGX("SNDCTL_DSP_SPEED %d\n", ival); + if (ival) { + if (aport->swstate != SW_INITIAL) { + DBGX("SNDCTL_DSP_SPEED failed: swstate = %d\n", + aport->swstate); + return -EINVAL; + } + if (ival < MIN_SPEED) + ival = MIN_SPEED; + if (ival > MAX_SPEED) + ival = MAX_SPEED; + if (rport) + rport->sw_framerate = ival; + if (wport) + wport->sw_framerate = ival; + } else + ival = aport->sw_framerate; + return put_user(ival, (int *) arg); + + case SNDCTL_DSP_STEREO: /* _SIOWR('P', 3, int) */ + if (get_user(ival, (int *) arg)) + return -EFAULT; + DBGX("SNDCTL_DSP_STEREO %d\n", ival); + if (ival != 0 && ival != 1) + return -EINVAL; + if (aport->swstate != SW_INITIAL) + return -EINVAL; + if (rport) + rport->sw_channels = ival + 1; + if (wport) + wport->sw_channels = ival + 1; + return put_user(ival, (int *) arg); + + case SNDCTL_DSP_CHANNELS: /* _SIOWR('P', 6, int) */ + if (get_user(ival, (int *) arg)) + return -EFAULT; + DBGX("SNDCTL_DSP_CHANNELS %d\n", ival); + if (ival != 1 && ival != 2) + return -EINVAL; + if (aport->swstate != SW_INITIAL) + return -EINVAL; + if (rport) + rport->sw_channels = ival; + if (wport) + wport->sw_channels = ival; + return put_user(ival, (int *) arg); + + case SNDCTL_DSP_GETBLKSIZE: /* _SIOWR('P', 4, int) */ + ival = pcm_setup(devc, rport, wport); + if (ival < 0) { + DBGX("SNDCTL_DSP_GETBLKSIZE failed, errno %d\n", ival); + return ival; + } + ival = 1 << aport->sw_fragshift; + DBGX("SNDCTL_DSP_GETBLKSIZE returning %d\n", ival); + return put_user(ival, (int *) arg); + + case SNDCTL_DSP_SETFRAGMENT: /* _SIOWR('P',10, int) */ + if (get_user(ival, (int *) arg)) + return -EFAULT; + DBGX("SNDCTL_DSP_SETFRAGMENT %d:%d\n", + ival >> 16, ival & 0xFFFF); + if (aport->swstate != SW_INITIAL) + return -EINVAL; + { + int sw_fragshift = ival & 0xFFFF; + int sw_subdivshift = aport->sw_subdivshift; + int hw_fragshift = sw_fragshift - sw_subdivshift; + int sw_fragcount = (ival >> 16) & 0xFFFF; + int hw_fragsize; + if (hw_fragshift < MIN_FRAGSHIFT) + hw_fragshift = MIN_FRAGSHIFT; + if (hw_fragshift > MAX_FRAGSHIFT) + hw_fragshift = MAX_FRAGSHIFT; + sw_fragshift = hw_fragshift + aport->sw_subdivshift; + hw_fragsize = 1 << hw_fragshift; + if (sw_fragcount < MIN_FRAGCOUNT(hw_fragsize)) + sw_fragcount = MIN_FRAGCOUNT(hw_fragsize); + if (sw_fragcount > MAX_FRAGCOUNT(hw_fragsize)) + sw_fragcount = MAX_FRAGCOUNT(hw_fragsize); + DBGPV("sw_fragshift = %d\n", sw_fragshift); + DBGPV("rport = 0x%p, wport = 0x%p\n", rport, wport); + if (rport) { + rport->sw_fragshift = sw_fragshift; + rport->sw_fragcount = sw_fragcount; + } + if (wport) { + wport->sw_fragshift = sw_fragshift; + wport->sw_fragcount = sw_fragcount; + } + ival = sw_fragcount << 16 | sw_fragshift; + } + DBGX("SNDCTL_DSP_SETFRAGMENT returns %d:%d\n", + ival >> 16, ival & 0xFFFF); + return put_user(ival, (int *) arg); + + case SNDCTL_DSP_SUBDIVIDE: /* _SIOWR('P', 9, int) */ + if (get_user(ival, (int *) arg)) + return -EFAULT; + DBGX("SNDCTL_DSP_SUBDIVIDE %d\n", ival); + if (aport->swstate != SW_INITIAL) + return -EINVAL; + { + int subdivshift; + int hw_fragshift, hw_fragsize, hw_fragcount; + switch (ival) { + case 1: subdivshift = 0; break; + case 2: subdivshift = 1; break; + case 4: subdivshift = 2; break; + default: return -EINVAL; + } + hw_fragshift = aport->sw_fragshift - subdivshift; + if (hw_fragshift < MIN_FRAGSHIFT || + hw_fragshift > MAX_FRAGSHIFT) + return -EINVAL; + hw_fragsize = 1 << hw_fragshift; + hw_fragcount = aport->sw_fragcount >> subdivshift; + if (hw_fragcount < MIN_FRAGCOUNT(hw_fragsize) || + hw_fragcount > MAX_FRAGCOUNT(hw_fragsize)) + return -EINVAL; + if (rport) + rport->sw_subdivshift = subdivshift; + if (wport) + wport->sw_subdivshift = subdivshift; + } + return 0; + + case SNDCTL_DSP_SETFMT: /* _SIOWR('P',5, int) */ + if (get_user(ival, (int *) arg)) + return -EFAULT; + DBGX("SNDCTL_DSP_SETFMT %d\n", ival); + if (ival != AFMT_QUERY) { + if (aport->swstate != SW_INITIAL) { + DBGP("SETFMT failed, swstate = %d\n", + aport->swstate); + return -EINVAL; + } + switch (ival) { + case AFMT_MU_LAW: + case AFMT_A_LAW: + case AFMT_U8: + case AFMT_S8: + case AFMT_S16_LE: + if (rport) + rport->sw_samplefmt = ival; + if (wport) + wport->sw_samplefmt = ival; + break; + default: + return -EINVAL; + } + } + ival = aport->sw_samplefmt; + return put_user(ival, (int *) arg); + + case SNDCTL_DSP_GETOSPACE: /* _SIOR ('P',12, audio_buf_info) */ + DBGXV("SNDCTL_DSP_GETOSPACE\n"); + if (!wport) + return -EINVAL; + ival = pcm_setup(devc, rport, wport); + if (ival < 0) + return ival; + ival = swb_inc_u(wport, 0); + buf_info.fragments = ival >> wport->sw_fragshift; + buf_info.fragstotal = wport->sw_fragcount; + buf_info.fragsize = 1 << wport->sw_fragshift; + buf_info.bytes = ival; + DBGXV("SNDCTL_DSP_GETOSPACE returns { %d %d %d %d }\n", + buf_info.fragments, buf_info.fragstotal, + buf_info.fragsize, buf_info.bytes); + if (copy_to_user((void *) arg, &buf_info, sizeof buf_info)) + return -EFAULT; + return 0; + + case SNDCTL_DSP_GETISPACE: /* _SIOR ('P',13, audio_buf_info) */ + DBGX("SNDCTL_DSP_GETISPACE\n"); + if (!rport) + return -EINVAL; + ival = pcm_setup(devc, rport, wport); + if (ival < 0) + return ival; + ival = swb_inc_u(rport, 0); + buf_info.fragments = ival >> rport->sw_fragshift; + buf_info.fragstotal = rport->sw_fragcount; + buf_info.fragsize = 1 << rport->sw_fragshift; + buf_info.bytes = ival; + DBGX("SNDCTL_DSP_GETISPACE returns { %d %d %d %d }\n", + buf_info.fragments, buf_info.fragstotal, + buf_info.fragsize, buf_info.bytes); + if (copy_to_user((void *) arg, &buf_info, sizeof buf_info)) + return -EFAULT; + return 0; + + case SNDCTL_DSP_NONBLOCK: /* _SIO ('P',14) */ + DBGX("SNDCTL_DSP_NONBLOCK\n"); + spin_lock(&file->f_lock); + file->f_flags |= O_NONBLOCK; + spin_unlock(&file->f_lock); + return 0; + + case SNDCTL_DSP_RESET: /* _SIO ('P', 0) */ + DBGX("SNDCTL_DSP_RESET\n"); + /* + * Nothing special needs to be done for input. Input + * samples sit in swbuf, but it will be reinitialized + * to empty when pcm_setup() is called. + */ + if (wport && wport->swbuf) { + wport->swstate = SW_INITIAL; + pcm_output(devc, 0, 0); + pcm_write_sync(devc); + } + pcm_shutdown(devc, rport, wport); + return 0; + + case SNDCTL_DSP_SYNC: /* _SIO ('P', 1) */ + DBGX("SNDCTL_DSP_SYNC\n"); + if (wport) { + pcm_flush_frag(devc); + pcm_write_sync(devc); + } + pcm_shutdown(devc, rport, wport); + return 0; + + case SNDCTL_DSP_POST: /* _SIO ('P', 8) */ + DBGX("SNDCTL_DSP_POST\n"); + if (!wport) + return -EINVAL; + pcm_flush_frag(devc); + return 0; + + case SNDCTL_DSP_GETIPTR: /* _SIOR ('P', 17, count_info) */ + DBGX("SNDCTL_DSP_GETIPTR\n"); + if (!rport) + return -EINVAL; + spin_lock_irqsave(&rport->lock, flags); + { + ustmsc_t ustmsc; + if (rport->hwstate == HW_RUNNING) { + ASSERT(rport->swstate == SW_RUN); + li_read_USTMSC(&rport->chan, &ustmsc); + info.bytes = ustmsc.msc - rport->MSC_offset; + info.bytes *= rport->frame_size; + } else { + info.bytes = rport->byte_count; + } + info.blocks = rport->frag_count; + info.ptr = 0; /* not implemented */ + rport->frag_count = 0; + } + spin_unlock_irqrestore(&rport->lock, flags); + if (copy_to_user((void *) arg, &info, sizeof info)) + return -EFAULT; + return 0; + + case SNDCTL_DSP_GETOPTR: /* _SIOR ('P',18, count_info) */ + DBGX("SNDCTL_DSP_GETOPTR\n"); + if (!wport) + return -EINVAL; + spin_lock_irqsave(&wport->lock, flags); + { + ustmsc_t ustmsc; + if (wport->hwstate == HW_RUNNING) { + ASSERT(wport->swstate == SW_RUN); + li_read_USTMSC(&wport->chan, &ustmsc); + info.bytes = ustmsc.msc - wport->MSC_offset; + info.bytes *= wport->frame_size; + } else { + info.bytes = wport->byte_count; + } + info.blocks = wport->frag_count; + info.ptr = 0; /* not implemented */ + wport->frag_count = 0; + } + spin_unlock_irqrestore(&wport->lock, flags); + if (copy_to_user((void *) arg, &info, sizeof info)) + return -EFAULT; + return 0; + + case SNDCTL_DSP_GETODELAY: /* _SIOR ('P', 23, int) */ + DBGX("SNDCTL_DSP_GETODELAY\n"); + if (!wport) + return -EINVAL; + spin_lock_irqsave(&wport->lock, flags); + { + int fsize = wport->frame_size; + ival = wport->swb_i_avail / fsize; + if (wport->hwstate == HW_RUNNING) { + int swptr, hwptr, hwframes, hwbytes, hwsize; + int totalhwbytes; + ustmsc_t ustmsc; + + hwsize = wport->hwbuf_size; + swptr = li_read_swptr(&wport->chan); + li_read_USTMSC(&wport->chan, &ustmsc); + hwframes = ustmsc.msc - wport->MSC_offset; + totalhwbytes = hwframes * fsize; + hwptr = totalhwbytes % hwsize; + hwbytes = (swptr - hwptr + hwsize) % hwsize; + ival += hwbytes / fsize; + } + } + spin_unlock_irqrestore(&wport->lock, flags); + return put_user(ival, (int *) arg); + + case SNDCTL_DSP_PROFILE: /* _SIOW ('P', 23, int) */ + DBGX("SNDCTL_DSP_PROFILE\n"); + + /* + * Thomas Sailer explains SNDCTL_DSP_PROFILE + * (private email, March 24, 1999): + * + * This gives the sound driver a hint on what it + * should do with partial fragments + * (i.e. fragments partially filled with write). + * This can direct the driver to zero them or + * leave them alone. But don't ask me what this + * is good for, my driver just zeroes the last + * fragment before the receiver stops, no idea + * what good for any other behaviour could + * be. Implementing it as NOP seems safe. + */ + + break; + + case SNDCTL_DSP_GETTRIGGER: /* _SIOR ('P',16, int) */ + DBGX("SNDCTL_DSP_GETTRIGGER\n"); + ival = 0; + if (rport) { + spin_lock_irqsave(&rport->lock, flags); + { + if (!(rport->flags & DISABLED)) + ival |= PCM_ENABLE_INPUT; + } + spin_unlock_irqrestore(&rport->lock, flags); + } + if (wport) { + spin_lock_irqsave(&wport->lock, flags); + { + if (!(wport->flags & DISABLED)) + ival |= PCM_ENABLE_OUTPUT; + } + spin_unlock_irqrestore(&wport->lock, flags); + } + return put_user(ival, (int *) arg); + + case SNDCTL_DSP_SETTRIGGER: /* _SIOW ('P',16, int) */ + if (get_user(ival, (int *) arg)) + return -EFAULT; + DBGX("SNDCTL_DSP_SETTRIGGER %d\n", ival); + + /* + * If user is disabling I/O and port is not in initial + * state, fail with EINVAL. + */ + + if (((rport && !(ival & PCM_ENABLE_INPUT)) || + (wport && !(ival & PCM_ENABLE_OUTPUT))) && + aport->swstate != SW_INITIAL) + return -EINVAL; + + if (rport) { + vwsnd_port_hwstate_t hwstate; + spin_lock_irqsave(&rport->lock, flags); + { + hwstate = rport->hwstate; + if (ival & PCM_ENABLE_INPUT) + rport->flags &= ~DISABLED; + else + rport->flags |= DISABLED; + } + spin_unlock_irqrestore(&rport->lock, flags); + if (hwstate != HW_RUNNING && ival & PCM_ENABLE_INPUT) { + + if (rport->swstate == SW_INITIAL) + pcm_setup(devc, rport, wport); + else + li_activate_dma(&rport->chan); + } + } + if (wport) { + vwsnd_port_flags_t pflags; + spin_lock_irqsave(&wport->lock, flags); + { + pflags = wport->flags; + if (ival & PCM_ENABLE_OUTPUT) + wport->flags &= ~DISABLED; + else + wport->flags |= DISABLED; + } + spin_unlock_irqrestore(&wport->lock, flags); + if (pflags & DISABLED && ival & PCM_ENABLE_OUTPUT) { + if (wport->swstate == SW_RUN) + pcm_output(devc, 0, 0); + } + } + return 0; + + default: + DBGP("unknown ioctl 0x%x\n", cmd); + return -EINVAL; + } + DBGP("unimplemented ioctl 0x%x\n", cmd); + return -EINVAL; +} + +static long vwsnd_audio_ioctl(struct file *file, + unsigned int cmd, + unsigned long arg) +{ + vwsnd_dev_t *devc = (vwsnd_dev_t *) file->private_data; + int ret; + + mutex_lock(&vwsnd_mutex); + mutex_lock(&devc->io_mutex); + ret = vwsnd_audio_do_ioctl(file, cmd, arg); + mutex_unlock(&devc->io_mutex); + mutex_unlock(&vwsnd_mutex); + + return ret; +} + +/* No mmap. */ + +static int vwsnd_audio_mmap(struct file *file, struct vm_area_struct *vma) +{ + DBGE("(file=0x%p, vma=0x%p)\n", file, vma); + return -ENODEV; +} + +/* + * Open the audio device for read and/or write. + * + * Returns 0 on success, -errno on failure. + */ + +static int vwsnd_audio_open(struct inode *inode, struct file *file) +{ + vwsnd_dev_t *devc; + int minor = iminor(inode); + int sw_samplefmt; + + DBGE("(inode=0x%p, file=0x%p)\n", inode, file); + + mutex_lock(&vwsnd_mutex); + INC_USE_COUNT; + for (devc = vwsnd_dev_list; devc; devc = devc->next_dev) + if ((devc->audio_minor & ~0x0F) == (minor & ~0x0F)) + break; + + if (devc == NULL) { + DEC_USE_COUNT; + mutex_unlock(&vwsnd_mutex); + return -ENODEV; + } + + mutex_lock(&devc->open_mutex); + while (devc->open_mode & file->f_mode) { + mutex_unlock(&devc->open_mutex); + if (file->f_flags & O_NONBLOCK) { + DEC_USE_COUNT; + mutex_unlock(&vwsnd_mutex); + return -EBUSY; + } + interruptible_sleep_on(&devc->open_wait); + if (signal_pending(current)) { + DEC_USE_COUNT; + mutex_unlock(&vwsnd_mutex); + return -ERESTARTSYS; + } + mutex_lock(&devc->open_mutex); + } + devc->open_mode |= file->f_mode & (FMODE_READ | FMODE_WRITE); + mutex_unlock(&devc->open_mutex); + + /* get default sample format from minor number. */ + + sw_samplefmt = 0; + if ((minor & 0xF) == SND_DEV_DSP) + sw_samplefmt = AFMT_U8; + else if ((minor & 0xF) == SND_DEV_AUDIO) + sw_samplefmt = AFMT_MU_LAW; + else if ((minor & 0xF) == SND_DEV_DSP16) + sw_samplefmt = AFMT_S16_LE; + else + ASSERT(0); + + /* Initialize vwsnd_ports. */ + + mutex_lock(&devc->io_mutex); + { + if (file->f_mode & FMODE_READ) { + devc->rport.swstate = SW_INITIAL; + devc->rport.flags = 0; + devc->rport.sw_channels = 1; + devc->rport.sw_samplefmt = sw_samplefmt; + devc->rport.sw_framerate = 8000; + devc->rport.sw_fragshift = DEFAULT_FRAGSHIFT; + devc->rport.sw_fragcount = DEFAULT_FRAGCOUNT; + devc->rport.sw_subdivshift = DEFAULT_SUBDIVSHIFT; + devc->rport.byte_count = 0; + devc->rport.frag_count = 0; + } + if (file->f_mode & FMODE_WRITE) { + devc->wport.swstate = SW_INITIAL; + devc->wport.flags = 0; + devc->wport.sw_channels = 1; + devc->wport.sw_samplefmt = sw_samplefmt; + devc->wport.sw_framerate = 8000; + devc->wport.sw_fragshift = DEFAULT_FRAGSHIFT; + devc->wport.sw_fragcount = DEFAULT_FRAGCOUNT; + devc->wport.sw_subdivshift = DEFAULT_SUBDIVSHIFT; + devc->wport.byte_count = 0; + devc->wport.frag_count = 0; + } + } + mutex_unlock(&devc->io_mutex); + + file->private_data = devc; + DBGRV(); + mutex_unlock(&vwsnd_mutex); + return 0; +} + +/* + * Release (close) the audio device. + */ + +static int vwsnd_audio_release(struct inode *inode, struct file *file) +{ + vwsnd_dev_t *devc = (vwsnd_dev_t *) file->private_data; + vwsnd_port_t *wport = NULL, *rport = NULL; + int err = 0; + + mutex_lock(&vwsnd_mutex); + mutex_lock(&devc->io_mutex); + { + DBGEV("(inode=0x%p, file=0x%p)\n", inode, file); + + if (file->f_mode & FMODE_READ) + rport = &devc->rport; + if (file->f_mode & FMODE_WRITE) { + wport = &devc->wport; + pcm_flush_frag(devc); + pcm_write_sync(devc); + } + pcm_shutdown(devc, rport, wport); + if (rport) + rport->swstate = SW_OFF; + if (wport) + wport->swstate = SW_OFF; + } + mutex_unlock(&devc->io_mutex); + + mutex_lock(&devc->open_mutex); + { + devc->open_mode &= ~file->f_mode; + } + mutex_unlock(&devc->open_mutex); + wake_up(&devc->open_wait); + DEC_USE_COUNT; + DBGR(); + mutex_unlock(&vwsnd_mutex); + return err; +} + +static const struct file_operations vwsnd_audio_fops = { + .owner = THIS_MODULE, + .llseek = no_llseek, + .read = vwsnd_audio_read, + .write = vwsnd_audio_write, + .poll = vwsnd_audio_poll, + .unlocked_ioctl = vwsnd_audio_ioctl, + .mmap = vwsnd_audio_mmap, + .open = vwsnd_audio_open, + .release = vwsnd_audio_release, +}; + +/*****************************************************************************/ +/* mixer driver */ + +/* open the mixer device. */ + +static int vwsnd_mixer_open(struct inode *inode, struct file *file) +{ + vwsnd_dev_t *devc; + + DBGEV("(inode=0x%p, file=0x%p)\n", inode, file); + + INC_USE_COUNT; + mutex_lock(&vwsnd_mutex); + for (devc = vwsnd_dev_list; devc; devc = devc->next_dev) + if (devc->mixer_minor == iminor(inode)) + break; + + if (devc == NULL) { + DEC_USE_COUNT; + mutex_unlock(&vwsnd_mutex); + return -ENODEV; + } + file->private_data = devc; + mutex_unlock(&vwsnd_mutex); + return 0; +} + +/* release (close) the mixer device. */ + +static int vwsnd_mixer_release(struct inode *inode, struct file *file) +{ + DBGEV("(inode=0x%p, file=0x%p)\n", inode, file); + DEC_USE_COUNT; + return 0; +} + +/* mixer_read_ioctl handles all read ioctls on the mixer device. */ + +static int mixer_read_ioctl(vwsnd_dev_t *devc, unsigned int nr, void __user *arg) +{ + int val = -1; + + DBGEV("(devc=0x%p, nr=0x%x, arg=0x%p)\n", devc, nr, arg); + + switch (nr) { + case SOUND_MIXER_CAPS: + val = SOUND_CAP_EXCL_INPUT; + break; + + case SOUND_MIXER_DEVMASK: + val = (SOUND_MASK_PCM | SOUND_MASK_LINE | + SOUND_MASK_MIC | SOUND_MASK_CD | SOUND_MASK_RECLEV); + break; + + case SOUND_MIXER_STEREODEVS: + val = (SOUND_MASK_PCM | SOUND_MASK_LINE | + SOUND_MASK_MIC | SOUND_MASK_CD | SOUND_MASK_RECLEV); + break; + + case SOUND_MIXER_OUTMASK: + val = (SOUND_MASK_PCM | SOUND_MASK_LINE | + SOUND_MASK_MIC | SOUND_MASK_CD); + break; + + case SOUND_MIXER_RECMASK: + val = (SOUND_MASK_PCM | SOUND_MASK_LINE | + SOUND_MASK_MIC | SOUND_MASK_CD); + break; + + case SOUND_MIXER_PCM: + val = ad1843_get_gain(&devc->lith, &ad1843_gain_PCM); + break; + + case SOUND_MIXER_LINE: + val = ad1843_get_gain(&devc->lith, &ad1843_gain_LINE); + break; + + case SOUND_MIXER_MIC: + val = ad1843_get_gain(&devc->lith, &ad1843_gain_MIC); + break; + + case SOUND_MIXER_CD: + val = ad1843_get_gain(&devc->lith, &ad1843_gain_CD); + break; + + case SOUND_MIXER_RECLEV: + val = ad1843_get_gain(&devc->lith, &ad1843_gain_RECLEV); + break; + + case SOUND_MIXER_RECSRC: + val = ad1843_get_recsrc(&devc->lith); + break; + + case SOUND_MIXER_OUTSRC: + val = ad1843_get_outsrc(&devc->lith); + break; + + default: + return -EINVAL; + } + return put_user(val, (int __user *) arg); +} + +/* mixer_write_ioctl handles all write ioctls on the mixer device. */ + +static int mixer_write_ioctl(vwsnd_dev_t *devc, unsigned int nr, void __user *arg) +{ + int val; + int err; + + DBGEV("(devc=0x%p, nr=0x%x, arg=0x%p)\n", devc, nr, arg); + + err = get_user(val, (int __user *) arg); + if (err) + return -EFAULT; + switch (nr) { + case SOUND_MIXER_PCM: + val = ad1843_set_gain(&devc->lith, &ad1843_gain_PCM, val); + break; + + case SOUND_MIXER_LINE: + val = ad1843_set_gain(&devc->lith, &ad1843_gain_LINE, val); + break; + + case SOUND_MIXER_MIC: + val = ad1843_set_gain(&devc->lith, &ad1843_gain_MIC, val); + break; + + case SOUND_MIXER_CD: + val = ad1843_set_gain(&devc->lith, &ad1843_gain_CD, val); + break; + + case SOUND_MIXER_RECLEV: + val = ad1843_set_gain(&devc->lith, &ad1843_gain_RECLEV, val); + break; + + case SOUND_MIXER_RECSRC: + if (devc->rport.swbuf || devc->wport.swbuf) + return -EBUSY; /* can't change recsrc while running */ + val = ad1843_set_recsrc(&devc->lith, val); + break; + + case SOUND_MIXER_OUTSRC: + val = ad1843_set_outsrc(&devc->lith, val); + break; + + default: + return -EINVAL; + } + if (val < 0) + return val; + return put_user(val, (int __user *) arg); +} + +/* This is the ioctl entry to the mixer driver. */ + +static long vwsnd_mixer_ioctl(struct file *file, + unsigned int cmd, + unsigned long arg) +{ + vwsnd_dev_t *devc = (vwsnd_dev_t *) file->private_data; + const unsigned int nrmask = _IOC_NRMASK << _IOC_NRSHIFT; + const unsigned int nr = (cmd & nrmask) >> _IOC_NRSHIFT; + int retval; + + DBGEV("(devc=0x%p, cmd=0x%x, arg=0x%lx)\n", devc, cmd, arg); + + mutex_lock(&vwsnd_mutex); + mutex_lock(&devc->mix_mutex); + { + if ((cmd & ~nrmask) == MIXER_READ(0)) + retval = mixer_read_ioctl(devc, nr, (void __user *) arg); + else if ((cmd & ~nrmask) == MIXER_WRITE(0)) + retval = mixer_write_ioctl(devc, nr, (void __user *) arg); + else + retval = -EINVAL; + } + mutex_unlock(&devc->mix_mutex); + mutex_unlock(&vwsnd_mutex); + return retval; +} + +static const struct file_operations vwsnd_mixer_fops = { + .owner = THIS_MODULE, + .llseek = no_llseek, + .unlocked_ioctl = vwsnd_mixer_ioctl, + .open = vwsnd_mixer_open, + .release = vwsnd_mixer_release, +}; + +/*****************************************************************************/ +/* probe/attach/unload */ + +/* driver probe routine. Return nonzero if hardware is found. */ + +static int __init probe_vwsnd(struct address_info *hw_config) +{ + lithium_t lith; + int w; + unsigned long later; + + DBGEV("(hw_config=0x%p)\n", hw_config); + + /* XXX verify lithium present (to prevent crash on non-vw) */ + + if (li_create(&lith, hw_config->io_base) != 0) { + printk(KERN_WARNING "probe_vwsnd: can't map lithium\n"); + return 0; + } + later = jiffies + 2; + li_writel(&lith, LI_HOST_CONTROLLER, LI_HC_LINK_ENABLE); + do { + w = li_readl(&lith, LI_HOST_CONTROLLER); + } while (w == LI_HC_LINK_ENABLE && time_before(jiffies, later)); + + li_destroy(&lith); + + DBGPV("HC = 0x%04x\n", w); + + if ((w == LI_HC_LINK_ENABLE) || (w & LI_HC_LINK_CODEC)) { + + /* This may indicate a beta machine with no audio, + * or a future machine with different audio. + * On beta-release 320 w/ no audio, HC == 0x4000 */ + + printk(KERN_WARNING "probe_vwsnd: audio codec not found\n"); + return 0; + } + + if (w & LI_HC_LINK_FAILURE) { + printk(KERN_WARNING "probe_vwsnd: can't init audio codec\n"); + return 0; + } + + printk(KERN_INFO "vwsnd: lithium audio at mmio %#x irq %d\n", + hw_config->io_base, hw_config->irq); + + return 1; +} + +/* + * driver attach routine. Initialize driver data structures and + * initialize hardware. A new vwsnd_dev_t is allocated and put + * onto the global list, vwsnd_dev_list. + * + * Return +minor_dev on success, -errno on failure. + */ + +static int __init attach_vwsnd(struct address_info *hw_config) +{ + vwsnd_dev_t *devc = NULL; + int err = -ENOMEM; + + DBGEV("(hw_config=0x%p)\n", hw_config); + + devc = kmalloc(sizeof (vwsnd_dev_t), GFP_KERNEL); + if (devc == NULL) + goto fail0; + + err = li_create(&devc->lith, hw_config->io_base); + if (err) + goto fail1; + + init_waitqueue_head(&devc->open_wait); + + devc->rport.hwbuf_size = HWBUF_SIZE; + devc->rport.hwbuf_vaddr = __get_free_pages(GFP_KERNEL, HWBUF_ORDER); + if (!devc->rport.hwbuf_vaddr) + goto fail2; + devc->rport.hwbuf = (void *) devc->rport.hwbuf_vaddr; + devc->rport.hwbuf_paddr = virt_to_phys(devc->rport.hwbuf); + + /* + * Quote from the NT driver: + * + * // WARNING!!! HACK to setup output dma!!! + * // This is required because even on output there is some data + * // trickling into the input DMA channel. This is a bug in the + * // Lithium microcode. + * // --sde + * + * We set the input side's DMA base address here. It will remain + * valid until the driver is unloaded. + */ + + li_writel(&devc->lith, LI_COMM1_BASE, + devc->rport.hwbuf_paddr >> 8 | 1 << (37 - 8)); + + devc->wport.hwbuf_size = HWBUF_SIZE; + devc->wport.hwbuf_vaddr = __get_free_pages(GFP_KERNEL, HWBUF_ORDER); + if (!devc->wport.hwbuf_vaddr) + goto fail3; + devc->wport.hwbuf = (void *) devc->wport.hwbuf_vaddr; + devc->wport.hwbuf_paddr = virt_to_phys(devc->wport.hwbuf); + DBGP("wport hwbuf = 0x%p\n", devc->wport.hwbuf); + + DBGDO(shut_up++); + err = ad1843_init(&devc->lith); + DBGDO(shut_up--); + if (err) + goto fail4; + + /* install interrupt handler */ + + err = request_irq(hw_config->irq, vwsnd_audio_intr, 0, "vwsnd", devc); + if (err) + goto fail5; + + /* register this device's drivers. */ + + devc->audio_minor = register_sound_dsp(&vwsnd_audio_fops, -1); + if ((err = devc->audio_minor) < 0) { + DBGDO(printk(KERN_WARNING + "attach_vwsnd: register_sound_dsp error %d\n", + err)); + goto fail6; + } + devc->mixer_minor = register_sound_mixer(&vwsnd_mixer_fops, + devc->audio_minor >> 4); + if ((err = devc->mixer_minor) < 0) { + DBGDO(printk(KERN_WARNING + "attach_vwsnd: register_sound_mixer error %d\n", + err)); + goto fail7; + } + + /* Squirrel away device indices for unload routine. */ + + hw_config->slots[0] = devc->audio_minor; + + /* Initialize as much of *devc as possible */ + + mutex_init(&devc->open_mutex); + mutex_init(&devc->io_mutex); + mutex_init(&devc->mix_mutex); + devc->open_mode = 0; + spin_lock_init(&devc->rport.lock); + init_waitqueue_head(&devc->rport.queue); + devc->rport.swstate = SW_OFF; + devc->rport.hwstate = HW_STOPPED; + devc->rport.flags = 0; + devc->rport.swbuf = NULL; + spin_lock_init(&devc->wport.lock); + init_waitqueue_head(&devc->wport.queue); + devc->wport.swstate = SW_OFF; + devc->wport.hwstate = HW_STOPPED; + devc->wport.flags = 0; + devc->wport.swbuf = NULL; + + /* Success. Link us onto the local device list. */ + + devc->next_dev = vwsnd_dev_list; + vwsnd_dev_list = devc; + return devc->audio_minor; + + /* So many ways to fail. Undo what we did. */ + + fail7: + unregister_sound_dsp(devc->audio_minor); + fail6: + free_irq(hw_config->irq, devc); + fail5: + fail4: + free_pages(devc->wport.hwbuf_vaddr, HWBUF_ORDER); + fail3: + free_pages(devc->rport.hwbuf_vaddr, HWBUF_ORDER); + fail2: + li_destroy(&devc->lith); + fail1: + kfree(devc); + fail0: + return err; +} + +static int __exit unload_vwsnd(struct address_info *hw_config) +{ + vwsnd_dev_t *devc, **devcp; + + DBGE("()\n"); + + devcp = &vwsnd_dev_list; + while ((devc = *devcp)) { + if (devc->audio_minor == hw_config->slots[0]) { + *devcp = devc->next_dev; + break; + } + devcp = &devc->next_dev; + } + + if (!devc) + return -ENODEV; + + unregister_sound_mixer(devc->mixer_minor); + unregister_sound_dsp(devc->audio_minor); + free_irq(hw_config->irq, devc); + free_pages(devc->wport.hwbuf_vaddr, HWBUF_ORDER); + free_pages(devc->rport.hwbuf_vaddr, HWBUF_ORDER); + li_destroy(&devc->lith); + kfree(devc); + + return 0; +} + +/*****************************************************************************/ +/* initialization and loadable kernel module interface */ + +static struct address_info the_hw_config = { + 0xFF001000, /* lithium phys addr */ + CO_IRQ(CO_APIC_LI_AUDIO) /* irq */ +}; + +MODULE_DESCRIPTION("SGI Visual Workstation sound module"); +MODULE_AUTHOR("Bob Miller <kbob@sgi.com>"); +MODULE_LICENSE("GPL"); + +static int __init init_vwsnd(void) +{ + int err; + + DBGXV("\n"); + DBGXV("sound::vwsnd::init_module()\n"); + + if (!probe_vwsnd(&the_hw_config)) + return -ENODEV; + + err = attach_vwsnd(&the_hw_config); + if (err < 0) + return err; + return 0; +} + +static void __exit cleanup_vwsnd(void) +{ + DBGX("sound::vwsnd::cleanup_module()\n"); + + unload_vwsnd(&the_hw_config); +} + +module_init(init_vwsnd); +module_exit(cleanup_vwsnd); diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c new file mode 100644 index 00000000..52468742 --- /dev/null +++ b/sound/oss/waveartist.c @@ -0,0 +1,2025 @@ +/* + * linux/sound/oss/waveartist.c + * + * The low level driver for the RWA010 Rockwell Wave Artist + * codec chip used in the Rebel.com NetWinder. + * + * Cleaned up and integrated into 2.1 by Russell King (rmk@arm.linux.org.uk) + * and Pat Beirne (patb@corel.ca) + * + * + * Copyright (C) by Rebel.com 1998-1999 + * + * RWA010 specs received under NDA from Rockwell + * + * Copyright (C) by Hannu Savolainen 1993-1997 + * + * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) + * Version 2 (June 1991). See the "COPYING" file distributed with this software + * for more info. + * + * Changes: + * 11-10-2000 Bartlomiej Zolnierkiewicz <bkz@linux-ide.org> + * Added __init to waveartist_init() + */ + +/* Debugging */ +#define DEBUG_CMD 1 +#define DEBUG_OUT 2 +#define DEBUG_IN 4 +#define DEBUG_INTR 8 +#define DEBUG_MIXER 16 +#define DEBUG_TRIGGER 32 + +#define debug_flg (0) + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/sched.h> +#include <linux/interrupt.h> +#include <linux/delay.h> +#include <linux/spinlock.h> +#include <linux/bitops.h> + +#include <asm/system.h> + +#include "sound_config.h" +#include "waveartist.h" + +#ifdef CONFIG_ARM +#include <mach/hardware.h> +#include <asm/mach-types.h> +#endif + +#ifndef NO_DMA +#define NO_DMA 255 +#endif + +#define SUPPORTED_MIXER_DEVICES (SOUND_MASK_SYNTH |\ + SOUND_MASK_PCM |\ + SOUND_MASK_LINE |\ + SOUND_MASK_MIC |\ + SOUND_MASK_LINE1 |\ + SOUND_MASK_RECLEV |\ + SOUND_MASK_VOLUME |\ + SOUND_MASK_IMIX) + +static unsigned short levels[SOUND_MIXER_NRDEVICES] = { + 0x5555, /* Master Volume */ + 0x0000, /* Bass */ + 0x0000, /* Treble */ + 0x2323, /* Synth (FM) */ + 0x4b4b, /* PCM */ + 0x6464, /* PC Speaker */ + 0x0000, /* Ext Line */ + 0x0000, /* Mic */ + 0x0000, /* CD */ + 0x6464, /* Recording monitor */ + 0x0000, /* SB PCM (ALT PCM) */ + 0x0000, /* Recording level */ + 0x6464, /* Input gain */ + 0x6464, /* Output gain */ + 0x0000, /* Line1 (Aux1) */ + 0x0000, /* Line2 (Aux2) */ + 0x0000, /* Line3 (Aux3) */ + 0x0000, /* Digital1 */ + 0x0000, /* Digital2 */ + 0x0000, /* Digital3 */ + 0x0000, /* Phone In */ + 0x6464, /* Phone Out */ + 0x0000, /* Video */ + 0x0000, /* Radio */ + 0x0000 /* Monitor */ +}; + +typedef struct { + struct address_info hw; /* hardware */ + char *chip_name; + + int xfer_count; + int audio_mode; + int open_mode; + int audio_flags; + int record_dev; + int playback_dev; + int dev_no; + + /* Mixer parameters */ + const struct waveartist_mixer_info *mix; + + unsigned short *levels; /* cache of volume settings */ + int recmask; /* currently enabled recording device! */ + +#ifdef CONFIG_ARCH_NETWINDER + signed int slider_vol; /* hardware slider volume */ + unsigned int handset_detect :1; + unsigned int telephone_detect:1; + unsigned int no_autoselect :1;/* handset/telephone autoselects a path */ + unsigned int spkr_mute_state :1;/* set by ioctl or autoselect */ + unsigned int line_mute_state :1;/* set by ioctl or autoselect */ + unsigned int use_slider :1;/* use slider setting for o/p vol */ +#endif +} wavnc_info; + +/* + * This is the implementation specific mixer information. + */ +struct waveartist_mixer_info { + unsigned int supported_devs; /* Supported devices */ + unsigned int recording_devs; /* Recordable devies */ + unsigned int stereo_devs; /* Stereo devices */ + + unsigned int (*select_input)(wavnc_info *, unsigned int, + unsigned char *, unsigned char *); + int (*decode_mixer)(wavnc_info *, int, + unsigned char, unsigned char); + int (*get_mixer)(wavnc_info *, int); +}; + +typedef struct wavnc_port_info { + int open_mode; + int speed; + int channels; + int audio_format; +} wavnc_port_info; + +static int nr_waveartist_devs; +static wavnc_info adev_info[MAX_AUDIO_DEV]; +static DEFINE_SPINLOCK(waveartist_lock); + +#ifndef CONFIG_ARCH_NETWINDER +#define machine_is_netwinder() 0 +#else +static struct timer_list vnc_timer; +static void vnc_configure_mixer(wavnc_info *devc, unsigned int input_mask); +static int vnc_private_ioctl(int dev, unsigned int cmd, int __user *arg); +static void vnc_slider_tick(unsigned long data); +#endif + +static inline void +waveartist_set_ctlr(struct address_info *hw, unsigned char clear, unsigned char set) +{ + unsigned int ctlr_port = hw->io_base + CTLR; + + clear = ~clear & inb(ctlr_port); + + outb(clear | set, ctlr_port); +} + +/* Toggle IRQ acknowledge line + */ +static inline void +waveartist_iack(wavnc_info *devc) +{ + unsigned int ctlr_port = devc->hw.io_base + CTLR; + int old_ctlr; + + old_ctlr = inb(ctlr_port) & ~IRQ_ACK; + + outb(old_ctlr | IRQ_ACK, ctlr_port); + outb(old_ctlr, ctlr_port); +} + +static inline int +waveartist_sleep(int timeout_ms) +{ + unsigned int timeout = msecs_to_jiffies(timeout_ms*100); + return schedule_timeout_interruptible(timeout); +} + +static int +waveartist_reset(wavnc_info *devc) +{ + struct address_info *hw = &devc->hw; + unsigned int timeout, res = -1; + + waveartist_set_ctlr(hw, -1, RESET); + waveartist_sleep(2); + waveartist_set_ctlr(hw, RESET, 0); + + timeout = 500; + do { + mdelay(2); + + if (inb(hw->io_base + STATR) & CMD_RF) { + res = inw(hw->io_base + CMDR); + if (res == 0x55aa) + break; + } + } while (--timeout); + + if (timeout == 0) { + printk(KERN_WARNING "WaveArtist: reset timeout "); + if (res != (unsigned int)-1) + printk("(res=%04X)", res); + printk("\n"); + return 1; + } + return 0; +} + +/* Helper function to send and receive words + * from WaveArtist. It handles all the handshaking + * and can send or receive multiple words. + */ +static int +waveartist_cmd(wavnc_info *devc, + int nr_cmd, unsigned int *cmd, + int nr_resp, unsigned int *resp) +{ + unsigned int io_base = devc->hw.io_base; + unsigned int timed_out = 0; + unsigned int i; + + if (debug_flg & DEBUG_CMD) { + printk("waveartist_cmd: cmd="); + + for (i = 0; i < nr_cmd; i++) + printk("%04X ", cmd[i]); + + printk("\n"); + } + + if (inb(io_base + STATR) & CMD_RF) { + int old_data; + + /* flush the port + */ + + old_data = inw(io_base + CMDR); + + if (debug_flg & DEBUG_CMD) + printk("flushed %04X...", old_data); + + udelay(10); + } + + for (i = 0; !timed_out && i < nr_cmd; i++) { + int count; + + for (count = 5000; count; count--) + if (inb(io_base + STATR) & CMD_WE) + break; + + if (!count) + timed_out = 1; + else + outw(cmd[i], io_base + CMDR); + } + + for (i = 0; !timed_out && i < nr_resp; i++) { + int count; + + for (count = 5000; count; count--) + if (inb(io_base + STATR) & CMD_RF) + break; + + if (!count) + timed_out = 1; + else + resp[i] = inw(io_base + CMDR); + } + + if (debug_flg & DEBUG_CMD) { + if (!timed_out) { + printk("waveartist_cmd: resp="); + + for (i = 0; i < nr_resp; i++) + printk("%04X ", resp[i]); + + printk("\n"); + } else + printk("waveartist_cmd: timed out\n"); + } + + return timed_out ? 1 : 0; +} + +/* + * Send one command word + */ +static inline int +waveartist_cmd1(wavnc_info *devc, unsigned int cmd) +{ + return waveartist_cmd(devc, 1, &cmd, 0, NULL); +} + +/* + * Send one command, receive one word + */ +static inline unsigned int +waveartist_cmd1_r(wavnc_info *devc, unsigned int cmd) +{ + unsigned int ret; + + waveartist_cmd(devc, 1, &cmd, 1, &ret); + + return ret; +} + +/* + * Send a double command, receive one + * word (and throw it away) + */ +static inline int +waveartist_cmd2(wavnc_info *devc, unsigned int cmd, unsigned int arg) +{ + unsigned int vals[2]; + + vals[0] = cmd; + vals[1] = arg; + + return waveartist_cmd(devc, 2, vals, 1, vals); +} + +/* + * Send a triple command + */ +static inline int +waveartist_cmd3(wavnc_info *devc, unsigned int cmd, + unsigned int arg1, unsigned int arg2) +{ + unsigned int vals[3]; + + vals[0] = cmd; + vals[1] = arg1; + vals[2] = arg2; + + return waveartist_cmd(devc, 3, vals, 0, NULL); +} + +static int +waveartist_getrev(wavnc_info *devc, char *rev) +{ + unsigned int temp[2]; + unsigned int cmd = WACMD_GETREV; + + waveartist_cmd(devc, 1, &cmd, 2, temp); + + rev[0] = temp[0] >> 8; + rev[1] = temp[0] & 255; + rev[2] = '\0'; + + return temp[0]; +} + +static void waveartist_halt_output(int dev); +static void waveartist_halt_input(int dev); +static void waveartist_halt(int dev); +static void waveartist_trigger(int dev, int state); + +static int +waveartist_open(int dev, int mode) +{ + wavnc_info *devc; + wavnc_port_info *portc; + unsigned long flags; + + if (dev < 0 || dev >= num_audiodevs) + return -ENXIO; + + devc = (wavnc_info *) audio_devs[dev]->devc; + portc = (wavnc_port_info *) audio_devs[dev]->portc; + + spin_lock_irqsave(&waveartist_lock, flags); + if (portc->open_mode || (devc->open_mode & mode)) { + spin_unlock_irqrestore(&waveartist_lock, flags); + return -EBUSY; + } + + devc->audio_mode = 0; + devc->open_mode |= mode; + portc->open_mode = mode; + waveartist_trigger(dev, 0); + + if (mode & OPEN_READ) + devc->record_dev = dev; + if (mode & OPEN_WRITE) + devc->playback_dev = dev; + spin_unlock_irqrestore(&waveartist_lock, flags); + + return 0; +} + +static void +waveartist_close(int dev) +{ + wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc; + wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc; + unsigned long flags; + + spin_lock_irqsave(&waveartist_lock, flags); + + waveartist_halt(dev); + + devc->audio_mode = 0; + devc->open_mode &= ~portc->open_mode; + portc->open_mode = 0; + + spin_unlock_irqrestore(&waveartist_lock, flags); +} + +static void +waveartist_output_block(int dev, unsigned long buf, int __count, int intrflag) +{ + wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc; + wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc; + unsigned long flags; + unsigned int count = __count; + + if (debug_flg & DEBUG_OUT) + printk("waveartist: output block, buf=0x%lx, count=0x%x...\n", + buf, count); + /* + * 16 bit data + */ + if (portc->audio_format & (AFMT_S16_LE | AFMT_S16_BE)) + count >>= 1; + + if (portc->channels > 1) + count >>= 1; + + count -= 1; + + if (devc->audio_mode & PCM_ENABLE_OUTPUT && + audio_devs[dev]->flags & DMA_AUTOMODE && + intrflag && + count == devc->xfer_count) { + devc->audio_mode |= PCM_ENABLE_OUTPUT; + return; /* + * Auto DMA mode on. No need to react + */ + } + + spin_lock_irqsave(&waveartist_lock, flags); + + /* + * set sample count + */ + waveartist_cmd2(devc, WACMD_OUTPUTSIZE, count); + + devc->xfer_count = count; + devc->audio_mode |= PCM_ENABLE_OUTPUT; + + spin_unlock_irqrestore(&waveartist_lock, flags); +} + +static void +waveartist_start_input(int dev, unsigned long buf, int __count, int intrflag) +{ + wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc; + wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc; + unsigned long flags; + unsigned int count = __count; + + if (debug_flg & DEBUG_IN) + printk("waveartist: start input, buf=0x%lx, count=0x%x...\n", + buf, count); + + if (portc->audio_format & (AFMT_S16_LE | AFMT_S16_BE)) /* 16 bit data */ + count >>= 1; + + if (portc->channels > 1) + count >>= 1; + + count -= 1; + + if (devc->audio_mode & PCM_ENABLE_INPUT && + audio_devs[dev]->flags & DMA_AUTOMODE && + intrflag && + count == devc->xfer_count) { + devc->audio_mode |= PCM_ENABLE_INPUT; + return; /* + * Auto DMA mode on. No need to react + */ + } + + spin_lock_irqsave(&waveartist_lock, flags); + + /* + * set sample count + */ + waveartist_cmd2(devc, WACMD_INPUTSIZE, count); + + devc->xfer_count = count; + devc->audio_mode |= PCM_ENABLE_INPUT; + + spin_unlock_irqrestore(&waveartist_lock, flags); +} + +static int +waveartist_ioctl(int dev, unsigned int cmd, void __user * arg) +{ + return -EINVAL; +} + +static unsigned int +waveartist_get_speed(wavnc_port_info *portc) +{ + unsigned int speed; + + /* + * program the speed, channels, bits + */ + if (portc->speed == 8000) + speed = 0x2E71; + else if (portc->speed == 11025) + speed = 0x4000; + else if (portc->speed == 22050) + speed = 0x8000; + else if (portc->speed == 44100) + speed = 0x0; + else { + /* + * non-standard - just calculate + */ + speed = portc->speed << 16; + + speed = (speed / 44100) & 65535; + } + + return speed; +} + +static unsigned int +waveartist_get_bits(wavnc_port_info *portc) +{ + unsigned int bits; + + if (portc->audio_format == AFMT_S16_LE) + bits = 1; + else if (portc->audio_format == AFMT_S8) + bits = 0; + else + bits = 2; //default AFMT_U8 + + return bits; +} + +static int +waveartist_prepare_for_input(int dev, int bsize, int bcount) +{ + unsigned long flags; + wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc; + wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc; + unsigned int speed, bits; + + if (devc->audio_mode) + return 0; + + speed = waveartist_get_speed(portc); + bits = waveartist_get_bits(portc); + + spin_lock_irqsave(&waveartist_lock, flags); + + if (waveartist_cmd2(devc, WACMD_INPUTFORMAT, bits)) + printk(KERN_WARNING "waveartist: error setting the " + "record format to %d\n", portc->audio_format); + + if (waveartist_cmd2(devc, WACMD_INPUTCHANNELS, portc->channels)) + printk(KERN_WARNING "waveartist: error setting record " + "to %d channels\n", portc->channels); + + /* + * write cmd SetSampleSpeedTimeConstant + */ + if (waveartist_cmd2(devc, WACMD_INPUTSPEED, speed)) + printk(KERN_WARNING "waveartist: error setting the record " + "speed to %dHz.\n", portc->speed); + + if (waveartist_cmd2(devc, WACMD_INPUTDMA, 1)) + printk(KERN_WARNING "waveartist: error setting the record " + "data path to 0x%X\n", 1); + + if (waveartist_cmd2(devc, WACMD_INPUTFORMAT, bits)) + printk(KERN_WARNING "waveartist: error setting the record " + "format to %d\n", portc->audio_format); + + devc->xfer_count = 0; + spin_unlock_irqrestore(&waveartist_lock, flags); + waveartist_halt_input(dev); + + if (debug_flg & DEBUG_INTR) { + printk("WA CTLR reg: 0x%02X.\n", + inb(devc->hw.io_base + CTLR)); + printk("WA STAT reg: 0x%02X.\n", + inb(devc->hw.io_base + STATR)); + printk("WA IRQS reg: 0x%02X.\n", + inb(devc->hw.io_base + IRQSTAT)); + } + + return 0; +} + +static int +waveartist_prepare_for_output(int dev, int bsize, int bcount) +{ + unsigned long flags; + wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc; + wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc; + unsigned int speed, bits; + + /* + * program the speed, channels, bits + */ + speed = waveartist_get_speed(portc); + bits = waveartist_get_bits(portc); + + spin_lock_irqsave(&waveartist_lock, flags); + + if (waveartist_cmd2(devc, WACMD_OUTPUTSPEED, speed) && + waveartist_cmd2(devc, WACMD_OUTPUTSPEED, speed)) + printk(KERN_WARNING "waveartist: error setting the playback " + "speed to %dHz.\n", portc->speed); + + if (waveartist_cmd2(devc, WACMD_OUTPUTCHANNELS, portc->channels)) + printk(KERN_WARNING "waveartist: error setting the playback " + "to %d channels\n", portc->channels); + + if (waveartist_cmd2(devc, WACMD_OUTPUTDMA, 0)) + printk(KERN_WARNING "waveartist: error setting the playback " + "data path to 0x%X\n", 0); + + if (waveartist_cmd2(devc, WACMD_OUTPUTFORMAT, bits)) + printk(KERN_WARNING "waveartist: error setting the playback " + "format to %d\n", portc->audio_format); + + devc->xfer_count = 0; + spin_unlock_irqrestore(&waveartist_lock, flags); + waveartist_halt_output(dev); + + if (debug_flg & DEBUG_INTR) { + printk("WA CTLR reg: 0x%02X.\n",inb(devc->hw.io_base + CTLR)); + printk("WA STAT reg: 0x%02X.\n",inb(devc->hw.io_base + STATR)); + printk("WA IRQS reg: 0x%02X.\n",inb(devc->hw.io_base + IRQSTAT)); + } + + return 0; +} + +static void +waveartist_halt(int dev) +{ + wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc; + wavnc_info *devc; + + if (portc->open_mode & OPEN_WRITE) + waveartist_halt_output(dev); + + if (portc->open_mode & OPEN_READ) + waveartist_halt_input(dev); + + devc = (wavnc_info *) audio_devs[dev]->devc; + devc->audio_mode = 0; +} + +static void +waveartist_halt_input(int dev) +{ + wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc; + unsigned long flags; + + spin_lock_irqsave(&waveartist_lock, flags); + + /* + * Stop capture + */ + waveartist_cmd1(devc, WACMD_INPUTSTOP); + + devc->audio_mode &= ~PCM_ENABLE_INPUT; + + /* + * Clear interrupt by toggling + * the IRQ_ACK bit in CTRL + */ + if (inb(devc->hw.io_base + STATR) & IRQ_REQ) + waveartist_iack(devc); + +// devc->audio_mode &= ~PCM_ENABLE_INPUT; + + spin_unlock_irqrestore(&waveartist_lock, flags); +} + +static void +waveartist_halt_output(int dev) +{ + wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc; + unsigned long flags; + + spin_lock_irqsave(&waveartist_lock, flags); + + waveartist_cmd1(devc, WACMD_OUTPUTSTOP); + + devc->audio_mode &= ~PCM_ENABLE_OUTPUT; + + /* + * Clear interrupt by toggling + * the IRQ_ACK bit in CTRL + */ + if (inb(devc->hw.io_base + STATR) & IRQ_REQ) + waveartist_iack(devc); + +// devc->audio_mode &= ~PCM_ENABLE_OUTPUT; + + spin_unlock_irqrestore(&waveartist_lock, flags); +} + +static void +waveartist_trigger(int dev, int state) +{ + wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc; + wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc; + unsigned long flags; + + if (debug_flg & DEBUG_TRIGGER) { + printk("wavnc: audio trigger "); + if (state & PCM_ENABLE_INPUT) + printk("in "); + if (state & PCM_ENABLE_OUTPUT) + printk("out"); + printk("\n"); + } + + spin_lock_irqsave(&waveartist_lock, flags); + + state &= devc->audio_mode; + + if (portc->open_mode & OPEN_READ && + state & PCM_ENABLE_INPUT) + /* + * enable ADC Data Transfer to PC + */ + waveartist_cmd1(devc, WACMD_INPUTSTART); + + if (portc->open_mode & OPEN_WRITE && + state & PCM_ENABLE_OUTPUT) + /* + * enable DAC data transfer from PC + */ + waveartist_cmd1(devc, WACMD_OUTPUTSTART); + + spin_unlock_irqrestore(&waveartist_lock, flags); +} + +static int +waveartist_set_speed(int dev, int arg) +{ + wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc; + + if (arg <= 0) + return portc->speed; + + if (arg < 5000) + arg = 5000; + if (arg > 44100) + arg = 44100; + + portc->speed = arg; + return portc->speed; + +} + +static short +waveartist_set_channels(int dev, short arg) +{ + wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc; + + if (arg != 1 && arg != 2) + return portc->channels; + + portc->channels = arg; + return arg; +} + +static unsigned int +waveartist_set_bits(int dev, unsigned int arg) +{ + wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc; + + if (arg == 0) + return portc->audio_format; + + if ((arg != AFMT_U8) && (arg != AFMT_S16_LE) && (arg != AFMT_S8)) + arg = AFMT_U8; + + portc->audio_format = arg; + + return arg; +} + +static struct audio_driver waveartist_audio_driver = { + .owner = THIS_MODULE, + .open = waveartist_open, + .close = waveartist_close, + .output_block = waveartist_output_block, + .start_input = waveartist_start_input, + .ioctl = waveartist_ioctl, + .prepare_for_input = waveartist_prepare_for_input, + .prepare_for_output = waveartist_prepare_for_output, + .halt_io = waveartist_halt, + .halt_input = waveartist_halt_input, + .halt_output = waveartist_halt_output, + .trigger = waveartist_trigger, + .set_speed = waveartist_set_speed, + .set_bits = waveartist_set_bits, + .set_channels = waveartist_set_channels +}; + + +static irqreturn_t +waveartist_intr(int irq, void *dev_id) +{ + wavnc_info *devc = dev_id; + int irqstatus, status; + + spin_lock(&waveartist_lock); + irqstatus = inb(devc->hw.io_base + IRQSTAT); + status = inb(devc->hw.io_base + STATR); + + if (debug_flg & DEBUG_INTR) + printk("waveartist_intr: stat=%02x, irqstat=%02x\n", + status, irqstatus); + + if (status & IRQ_REQ) /* Clear interrupt */ + waveartist_iack(devc); + else + printk(KERN_WARNING "waveartist: unexpected interrupt\n"); + + if (irqstatus & 0x01) { + int temp = 1; + + /* PCM buffer done + */ + if ((status & DMA0) && (devc->audio_mode & PCM_ENABLE_OUTPUT)) { + DMAbuf_outputintr(devc->playback_dev, 1); + temp = 0; + } + if ((status & DMA1) && (devc->audio_mode & PCM_ENABLE_INPUT)) { + DMAbuf_inputintr(devc->record_dev); + temp = 0; + } + if (temp) //default: + printk(KERN_WARNING "waveartist: Unknown interrupt\n"); + } + if (irqstatus & 0x2) + // We do not use SB mode natively... + printk(KERN_WARNING "waveartist: Unexpected SB interrupt...\n"); + spin_unlock(&waveartist_lock); + return IRQ_HANDLED; +} + +/* ------------------------------------------------------------------------- + * Mixer stuff + */ +struct mix_ent { + unsigned char reg_l; + unsigned char reg_r; + unsigned char shift; + unsigned char max; +}; + +static const struct mix_ent mix_devs[SOUND_MIXER_NRDEVICES] = { + { 2, 6, 1, 7 }, /* SOUND_MIXER_VOLUME */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_BASS */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_TREBLE */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_SYNTH */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_PCM */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_SPEAKER */ + { 0, 4, 6, 31 }, /* SOUND_MIXER_LINE */ + { 2, 6, 4, 3 }, /* SOUND_MIXER_MIC */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_CD */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_IMIX */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_ALTPCM */ +#if 0 + { 3, 7, 0, 10 }, /* SOUND_MIXER_RECLEV */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_IGAIN */ +#else + { 0, 0, 0, 0 }, /* SOUND_MIXER_RECLEV */ + { 3, 7, 0, 7 }, /* SOUND_MIXER_IGAIN */ +#endif + { 0, 0, 0, 0 }, /* SOUND_MIXER_OGAIN */ + { 0, 4, 1, 31 }, /* SOUND_MIXER_LINE1 */ + { 1, 5, 6, 31 }, /* SOUND_MIXER_LINE2 */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_LINE3 */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_DIGITAL1 */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_DIGITAL2 */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_DIGITAL3 */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_PHONEIN */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_PHONEOUT */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_VIDEO */ + { 0, 0, 0, 0 }, /* SOUND_MIXER_RADIO */ + { 0, 0, 0, 0 } /* SOUND_MIXER_MONITOR */ +}; + +static void +waveartist_mixer_update(wavnc_info *devc, int whichDev) +{ + unsigned int lev_left, lev_right; + + lev_left = devc->levels[whichDev] & 0xff; + lev_right = devc->levels[whichDev] >> 8; + + if (lev_left > 100) + lev_left = 100; + if (lev_right > 100) + lev_right = 100; + +#define SCALE(lev,max) ((lev) * (max) / 100) + + if (machine_is_netwinder() && whichDev == SOUND_MIXER_PHONEOUT) + whichDev = SOUND_MIXER_VOLUME; + + if (mix_devs[whichDev].reg_l || mix_devs[whichDev].reg_r) { + const struct mix_ent *mix = mix_devs + whichDev; + unsigned int mask, left, right; + + mask = mix->max << mix->shift; + lev_left = SCALE(lev_left, mix->max) << mix->shift; + lev_right = SCALE(lev_right, mix->max) << mix->shift; + + /* read left setting */ + left = waveartist_cmd1_r(devc, WACMD_GET_LEVEL | + mix->reg_l << 8); + + /* read right setting */ + right = waveartist_cmd1_r(devc, WACMD_GET_LEVEL | + mix->reg_r << 8); + + left = (left & ~mask) | (lev_left & mask); + right = (right & ~mask) | (lev_right & mask); + + /* write left,right back */ + waveartist_cmd3(devc, WACMD_SET_MIXER, left, right); + } else { + switch(whichDev) { + case SOUND_MIXER_PCM: + waveartist_cmd3(devc, WACMD_SET_LEVEL, + SCALE(lev_left, 32767), + SCALE(lev_right, 32767)); + break; + + case SOUND_MIXER_SYNTH: + waveartist_cmd3(devc, 0x0100 | WACMD_SET_LEVEL, + SCALE(lev_left, 32767), + SCALE(lev_right, 32767)); + break; + } + } +} + +/* + * Set the ADC MUX to the specified values. We do NOT do any + * checking of the values passed, since we assume that the + * relevant *_select_input function has done that for us. + */ +static void +waveartist_set_adc_mux(wavnc_info *devc, char left_dev, char right_dev) +{ + unsigned int reg_08, reg_09; + + reg_08 = waveartist_cmd1_r(devc, WACMD_GET_LEVEL | 0x0800); + reg_09 = waveartist_cmd1_r(devc, WACMD_GET_LEVEL | 0x0900); + + reg_08 = (reg_08 & ~0x3f) | right_dev << 3 | left_dev; + + waveartist_cmd3(devc, WACMD_SET_MIXER, reg_08, reg_09); +} + +/* + * Decode a recording mask into a mixer selection as follows: + * + * OSS Source WA Source Actual source + * SOUND_MASK_IMIX Mixer Mixer output (same as AD1848) + * SOUND_MASK_LINE Line Line in + * SOUND_MASK_LINE1 Aux 1 Aux 1 in + * SOUND_MASK_LINE2 Aux 2 Aux 2 in + * SOUND_MASK_MIC Mic Microphone + */ +static unsigned int +waveartist_select_input(wavnc_info *devc, unsigned int recmask, + unsigned char *dev_l, unsigned char *dev_r) +{ + unsigned int recdev = ADC_MUX_NONE; + + if (recmask & SOUND_MASK_IMIX) { + recmask = SOUND_MASK_IMIX; + recdev = ADC_MUX_MIXER; + } else if (recmask & SOUND_MASK_LINE2) { + recmask = SOUND_MASK_LINE2; + recdev = ADC_MUX_AUX2; + } else if (recmask & SOUND_MASK_LINE1) { + recmask = SOUND_MASK_LINE1; + recdev = ADC_MUX_AUX1; + } else if (recmask & SOUND_MASK_LINE) { + recmask = SOUND_MASK_LINE; + recdev = ADC_MUX_LINE; + } else if (recmask & SOUND_MASK_MIC) { + recmask = SOUND_MASK_MIC; + recdev = ADC_MUX_MIC; + } + + *dev_l = *dev_r = recdev; + + return recmask; +} + +static int +waveartist_decode_mixer(wavnc_info *devc, int dev, unsigned char lev_l, + unsigned char lev_r) +{ + switch (dev) { + case SOUND_MIXER_VOLUME: + case SOUND_MIXER_SYNTH: + case SOUND_MIXER_PCM: + case SOUND_MIXER_LINE: + case SOUND_MIXER_MIC: + case SOUND_MIXER_IGAIN: + case SOUND_MIXER_LINE1: + case SOUND_MIXER_LINE2: + devc->levels[dev] = lev_l | lev_r << 8; + break; + + case SOUND_MIXER_IMIX: + break; + + default: + dev = -EINVAL; + break; + } + + return dev; +} + +static int waveartist_get_mixer(wavnc_info *devc, int dev) +{ + return devc->levels[dev]; +} + +static const struct waveartist_mixer_info waveartist_mixer = { + .supported_devs = SUPPORTED_MIXER_DEVICES | SOUND_MASK_IGAIN, + .recording_devs = SOUND_MASK_LINE | SOUND_MASK_MIC | + SOUND_MASK_LINE1 | SOUND_MASK_LINE2 | + SOUND_MASK_IMIX, + .stereo_devs = (SUPPORTED_MIXER_DEVICES | SOUND_MASK_IGAIN) & ~ + (SOUND_MASK_SPEAKER | SOUND_MASK_IMIX), + .select_input = waveartist_select_input, + .decode_mixer = waveartist_decode_mixer, + .get_mixer = waveartist_get_mixer, +}; + +static void +waveartist_set_recmask(wavnc_info *devc, unsigned int recmask) +{ + unsigned char dev_l, dev_r; + + recmask &= devc->mix->recording_devs; + + /* + * If more than one recording device selected, + * disable the device that is currently in use. + */ + if (hweight32(recmask) > 1) + recmask &= ~devc->recmask; + + /* + * Translate the recording device mask into + * the ADC multiplexer settings. + */ + devc->recmask = devc->mix->select_input(devc, recmask, + &dev_l, &dev_r); + + waveartist_set_adc_mux(devc, dev_l, dev_r); +} + +static int +waveartist_set_mixer(wavnc_info *devc, int dev, unsigned int level) +{ + unsigned int lev_left = level & 0x00ff; + unsigned int lev_right = (level & 0xff00) >> 8; + + if (lev_left > 100) + lev_left = 100; + if (lev_right > 100) + lev_right = 100; + + /* + * Mono devices have their right volume forced to their + * left volume. (from ALSA driver OSS emulation). + */ + if (!(devc->mix->stereo_devs & (1 << dev))) + lev_right = lev_left; + + dev = devc->mix->decode_mixer(devc, dev, lev_left, lev_right); + + if (dev >= 0) + waveartist_mixer_update(devc, dev); + + return dev < 0 ? dev : 0; +} + +static int +waveartist_mixer_ioctl(int dev, unsigned int cmd, void __user * arg) +{ + wavnc_info *devc = (wavnc_info *)audio_devs[dev]->devc; + int ret = 0, val, nr; + + /* + * All SOUND_MIXER_* ioctls use type 'M' + */ + if (((cmd >> 8) & 255) != 'M') + return -ENOIOCTLCMD; + +#ifdef CONFIG_ARCH_NETWINDER + if (machine_is_netwinder()) { + ret = vnc_private_ioctl(dev, cmd, arg); + if (ret != -ENOIOCTLCMD) + return ret; + else + ret = 0; + } +#endif + + nr = cmd & 0xff; + + if (_SIOC_DIR(cmd) & _SIOC_WRITE) { + if (get_user(val, (int __user *)arg)) + return -EFAULT; + + switch (nr) { + case SOUND_MIXER_RECSRC: + waveartist_set_recmask(devc, val); + break; + + default: + ret = -EINVAL; + if (nr < SOUND_MIXER_NRDEVICES && + devc->mix->supported_devs & (1 << nr)) + ret = waveartist_set_mixer(devc, nr, val); + } + } + + if (ret == 0 && _SIOC_DIR(cmd) & _SIOC_READ) { + ret = -EINVAL; + + switch (nr) { + case SOUND_MIXER_RECSRC: + ret = devc->recmask; + break; + + case SOUND_MIXER_DEVMASK: + ret = devc->mix->supported_devs; + break; + + case SOUND_MIXER_STEREODEVS: + ret = devc->mix->stereo_devs; + break; + + case SOUND_MIXER_RECMASK: + ret = devc->mix->recording_devs; + break; + + case SOUND_MIXER_CAPS: + ret = SOUND_CAP_EXCL_INPUT; + break; + + default: + if (nr < SOUND_MIXER_NRDEVICES) + ret = devc->mix->get_mixer(devc, nr); + break; + } + + if (ret >= 0) + ret = put_user(ret, (int __user *)arg) ? -EFAULT : 0; + } + + return ret; +} + +static struct mixer_operations waveartist_mixer_operations = +{ + .owner = THIS_MODULE, + .id = "WaveArtist", + .name = "WaveArtist", + .ioctl = waveartist_mixer_ioctl +}; + +static void +waveartist_mixer_reset(wavnc_info *devc) +{ + int i; + + if (debug_flg & DEBUG_MIXER) + printk("%s: mixer_reset\n", devc->hw.name); + + /* + * reset mixer cmd + */ + waveartist_cmd1(devc, WACMD_RST_MIXER); + + /* + * set input for ADC to come from 'quiet' + * turn on default modes + */ + waveartist_cmd3(devc, WACMD_SET_MIXER, 0x9800, 0xa836); + + /* + * set mixer input select to none, RX filter gains 0 dB + */ + waveartist_cmd3(devc, WACMD_SET_MIXER, 0x4c00, 0x8c00); + + /* + * set bit 0 reg 2 to 1 - unmute MonoOut + */ + waveartist_cmd3(devc, WACMD_SET_MIXER, 0x2801, 0x6800); + + /* set default input device = internal mic + * current recording device = none + */ + waveartist_set_recmask(devc, 0); + + for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) + waveartist_mixer_update(devc, i); +} + +static int __init waveartist_init(wavnc_info *devc) +{ + wavnc_port_info *portc; + char rev[3], dev_name[64]; + int my_dev; + + if (waveartist_reset(devc)) + return -ENODEV; + + sprintf(dev_name, "%s (%s", devc->hw.name, devc->chip_name); + + if (waveartist_getrev(devc, rev)) { + strcat(dev_name, " rev. "); + strcat(dev_name, rev); + } + strcat(dev_name, ")"); + + conf_printf2(dev_name, devc->hw.io_base, devc->hw.irq, + devc->hw.dma, devc->hw.dma2); + + portc = kzalloc(sizeof(wavnc_port_info), GFP_KERNEL); + if (portc == NULL) + goto nomem; + + my_dev = sound_install_audiodrv(AUDIO_DRIVER_VERSION, dev_name, + &waveartist_audio_driver, sizeof(struct audio_driver), + devc->audio_flags, AFMT_U8 | AFMT_S16_LE | AFMT_S8, + devc, devc->hw.dma, devc->hw.dma2); + + if (my_dev < 0) + goto free; + + audio_devs[my_dev]->portc = portc; + + waveartist_mixer_reset(devc); + + /* + * clear any pending interrupt + */ + waveartist_iack(devc); + + if (request_irq(devc->hw.irq, waveartist_intr, 0, devc->hw.name, devc) < 0) { + printk(KERN_ERR "%s: IRQ %d in use\n", + devc->hw.name, devc->hw.irq); + goto uninstall; + } + + if (sound_alloc_dma(devc->hw.dma, devc->hw.name)) { + printk(KERN_ERR "%s: Can't allocate DMA%d\n", + devc->hw.name, devc->hw.dma); + goto uninstall_irq; + } + + if (devc->hw.dma != devc->hw.dma2 && devc->hw.dma2 != NO_DMA) + if (sound_alloc_dma(devc->hw.dma2, devc->hw.name)) { + printk(KERN_ERR "%s: can't allocate DMA%d\n", + devc->hw.name, devc->hw.dma2); + goto uninstall_dma; + } + + waveartist_set_ctlr(&devc->hw, 0, DMA1_IE | DMA0_IE); + + audio_devs[my_dev]->mixer_dev = + sound_install_mixer(MIXER_DRIVER_VERSION, + dev_name, + &waveartist_mixer_operations, + sizeof(struct mixer_operations), + devc); + + return my_dev; + +uninstall_dma: + sound_free_dma(devc->hw.dma); + +uninstall_irq: + free_irq(devc->hw.irq, devc); + +uninstall: + sound_unload_audiodev(my_dev); + +free: + kfree(portc); + +nomem: + return -1; +} + +static int __init probe_waveartist(struct address_info *hw_config) +{ + wavnc_info *devc = &adev_info[nr_waveartist_devs]; + + if (nr_waveartist_devs >= MAX_AUDIO_DEV) { + printk(KERN_WARNING "waveartist: too many audio devices\n"); + return 0; + } + + if (!request_region(hw_config->io_base, 15, hw_config->name)) { + printk(KERN_WARNING "WaveArtist: I/O port conflict\n"); + return 0; + } + + if (hw_config->irq > 15 || hw_config->irq < 0) { + release_region(hw_config->io_base, 15); + printk(KERN_WARNING "WaveArtist: Bad IRQ %d\n", + hw_config->irq); + return 0; + } + + if (hw_config->dma != 3) { + release_region(hw_config->io_base, 15); + printk(KERN_WARNING "WaveArtist: Bad DMA %d\n", + hw_config->dma); + return 0; + } + + hw_config->name = "WaveArtist"; + devc->hw = *hw_config; + devc->open_mode = 0; + devc->chip_name = "RWA-010"; + + return 1; +} + +static void __init +attach_waveartist(struct address_info *hw, const struct waveartist_mixer_info *mix) +{ + wavnc_info *devc = &adev_info[nr_waveartist_devs]; + + /* + * NOTE! If irq < 0, there is another driver which has allocated the + * IRQ so that this driver doesn't need to allocate/deallocate it. + * The actually used IRQ is ABS(irq). + */ + devc->hw = *hw; + devc->hw.irq = (hw->irq > 0) ? hw->irq : 0; + devc->open_mode = 0; + devc->playback_dev = 0; + devc->record_dev = 0; + devc->audio_flags = DMA_AUTOMODE; + devc->levels = levels; + + if (hw->dma != hw->dma2 && hw->dma2 != NO_DMA) + devc->audio_flags |= DMA_DUPLEX; + + devc->mix = mix; + devc->dev_no = waveartist_init(devc); + + if (devc->dev_no < 0) + release_region(hw->io_base, 15); + else { +#ifdef CONFIG_ARCH_NETWINDER + if (machine_is_netwinder()) { + init_timer(&vnc_timer); + vnc_timer.function = vnc_slider_tick; + vnc_timer.expires = jiffies; + vnc_timer.data = nr_waveartist_devs; + add_timer(&vnc_timer); + + vnc_configure_mixer(devc, 0); + + devc->no_autoselect = 1; + } +#endif + nr_waveartist_devs += 1; + } +} + +static void __exit unload_waveartist(struct address_info *hw) +{ + wavnc_info *devc = NULL; + int i; + + for (i = 0; i < nr_waveartist_devs; i++) + if (hw->io_base == adev_info[i].hw.io_base) { + devc = adev_info + i; + break; + } + + if (devc != NULL) { + int mixer; + +#ifdef CONFIG_ARCH_NETWINDER + if (machine_is_netwinder()) + del_timer(&vnc_timer); +#endif + + release_region(devc->hw.io_base, 15); + + waveartist_set_ctlr(&devc->hw, DMA1_IE|DMA0_IE, 0); + + if (devc->hw.irq >= 0) + free_irq(devc->hw.irq, devc); + + sound_free_dma(devc->hw.dma); + + if (devc->hw.dma != devc->hw.dma2 && + devc->hw.dma2 != NO_DMA) + sound_free_dma(devc->hw.dma2); + + mixer = audio_devs[devc->dev_no]->mixer_dev; + + if (mixer >= 0) + sound_unload_mixerdev(mixer); + + if (devc->dev_no >= 0) + sound_unload_audiodev(devc->dev_no); + + nr_waveartist_devs -= 1; + + for (; i < nr_waveartist_devs; i++) + adev_info[i] = adev_info[i + 1]; + } else + printk(KERN_WARNING "waveartist: can't find device " + "to unload\n"); +} + +#ifdef CONFIG_ARCH_NETWINDER + +/* + * Rebel.com Netwinder specifics... + */ + +#include <asm/hardware/dec21285.h> + +#define VNC_TIMER_PERIOD (HZ/4) //check slider 4 times/sec + +#define MIXER_PRIVATE3_RESET 0x53570000 +#define MIXER_PRIVATE3_READ 0x53570001 +#define MIXER_PRIVATE3_WRITE 0x53570002 + +#define VNC_MUTE_INTERNAL_SPKR 0x01 //the sw mute on/off control bit +#define VNC_MUTE_LINE_OUT 0x10 +#define VNC_PHONE_DETECT 0x20 +#define VNC_HANDSET_DETECT 0x40 +#define VNC_DISABLE_AUTOSWITCH 0x80 + +static inline void +vnc_mute_spkr(wavnc_info *devc) +{ + unsigned long flags; + + spin_lock_irqsave(&nw_gpio_lock, flags); + nw_cpld_modify(CPLD_UNMUTE, devc->spkr_mute_state ? 0 : CPLD_UNMUTE); + spin_unlock_irqrestore(&nw_gpio_lock, flags); +} + +static void +vnc_mute_lout(wavnc_info *devc) +{ + unsigned int left, right; + + left = waveartist_cmd1_r(devc, WACMD_GET_LEVEL); + right = waveartist_cmd1_r(devc, WACMD_GET_LEVEL | 0x400); + + if (devc->line_mute_state) { + left &= ~1; + right &= ~1; + } else { + left |= 1; + right |= 1; + } + waveartist_cmd3(devc, WACMD_SET_MIXER, left, right); + +} + +static int +vnc_volume_slider(wavnc_info *devc) +{ + static signed int old_slider_volume; + unsigned long flags; + signed int volume = 255; + + *CSR_TIMER1_LOAD = 0x00ffffff; + + spin_lock_irqsave(&waveartist_lock, flags); + + outb(0xFF, 0x201); + *CSR_TIMER1_CNTL = TIMER_CNTL_ENABLE | TIMER_CNTL_DIV1; + + while (volume && (inb(0x201) & 0x01)) + volume--; + + *CSR_TIMER1_CNTL = 0; + + spin_unlock_irqrestore(&waveartist_lock,flags); + + volume = 0x00ffffff - *CSR_TIMER1_VALUE; + + +#ifndef REVERSE + volume = 150 - (volume >> 5); +#else + volume = (volume >> 6) - 25; +#endif + + if (volume < 0) + volume = 0; + + if (volume > 100) + volume = 100; + + /* + * slider quite often reads +-8, so debounce this random noise + */ + if (abs(volume - old_slider_volume) > 7) { + old_slider_volume = volume; + + if (debug_flg & DEBUG_MIXER) + printk(KERN_DEBUG "Slider volume: %d.\n", volume); + } + + return old_slider_volume; +} + +/* + * Decode a recording mask into a mixer selection on the NetWinder + * as follows: + * + * OSS Source WA Source Actual source + * SOUND_MASK_IMIX Mixer Mixer output (same as AD1848) + * SOUND_MASK_LINE Line Line in + * SOUND_MASK_LINE1 Left Mic Handset + * SOUND_MASK_PHONEIN Left Aux Telephone microphone + * SOUND_MASK_MIC Right Mic Builtin microphone + */ +static unsigned int +netwinder_select_input(wavnc_info *devc, unsigned int recmask, + unsigned char *dev_l, unsigned char *dev_r) +{ + unsigned int recdev_l = ADC_MUX_NONE, recdev_r = ADC_MUX_NONE; + + if (recmask & SOUND_MASK_IMIX) { + recmask = SOUND_MASK_IMIX; + recdev_l = ADC_MUX_MIXER; + recdev_r = ADC_MUX_MIXER; + } else if (recmask & SOUND_MASK_LINE) { + recmask = SOUND_MASK_LINE; + recdev_l = ADC_MUX_LINE; + recdev_r = ADC_MUX_LINE; + } else if (recmask & SOUND_MASK_LINE1) { + recmask = SOUND_MASK_LINE1; + waveartist_cmd1(devc, WACMD_SET_MONO); /* left */ + recdev_l = ADC_MUX_MIC; + recdev_r = ADC_MUX_NONE; + } else if (recmask & SOUND_MASK_PHONEIN) { + recmask = SOUND_MASK_PHONEIN; + waveartist_cmd1(devc, WACMD_SET_MONO); /* left */ + recdev_l = ADC_MUX_AUX1; + recdev_r = ADC_MUX_NONE; + } else if (recmask & SOUND_MASK_MIC) { + recmask = SOUND_MASK_MIC; + waveartist_cmd1(devc, WACMD_SET_MONO | 0x100); /* right */ + recdev_l = ADC_MUX_NONE; + recdev_r = ADC_MUX_MIC; + } + + *dev_l = recdev_l; + *dev_r = recdev_r; + + return recmask; +} + +static int +netwinder_decode_mixer(wavnc_info *devc, int dev, unsigned char lev_l, + unsigned char lev_r) +{ + switch (dev) { + case SOUND_MIXER_VOLUME: + case SOUND_MIXER_SYNTH: + case SOUND_MIXER_PCM: + case SOUND_MIXER_LINE: + case SOUND_MIXER_IGAIN: + devc->levels[dev] = lev_l | lev_r << 8; + break; + + case SOUND_MIXER_MIC: /* right mic only */ + devc->levels[SOUND_MIXER_MIC] &= 0xff; + devc->levels[SOUND_MIXER_MIC] |= lev_l << 8; + break; + + case SOUND_MIXER_LINE1: /* left mic only */ + devc->levels[SOUND_MIXER_MIC] &= 0xff00; + devc->levels[SOUND_MIXER_MIC] |= lev_l; + dev = SOUND_MIXER_MIC; + break; + + case SOUND_MIXER_PHONEIN: /* left aux only */ + devc->levels[SOUND_MIXER_LINE1] = lev_l; + dev = SOUND_MIXER_LINE1; + break; + + case SOUND_MIXER_IMIX: + case SOUND_MIXER_PHONEOUT: + break; + + default: + dev = -EINVAL; + break; + } + return dev; +} + +static int netwinder_get_mixer(wavnc_info *devc, int dev) +{ + int levels; + + switch (dev) { + case SOUND_MIXER_VOLUME: + case SOUND_MIXER_SYNTH: + case SOUND_MIXER_PCM: + case SOUND_MIXER_LINE: + case SOUND_MIXER_IGAIN: + levels = devc->levels[dev]; + break; + + case SOUND_MIXER_MIC: /* builtin mic: right mic only */ + levels = devc->levels[SOUND_MIXER_MIC] >> 8; + levels |= levels << 8; + break; + + case SOUND_MIXER_LINE1: /* handset mic: left mic only */ + levels = devc->levels[SOUND_MIXER_MIC] & 0xff; + levels |= levels << 8; + break; + + case SOUND_MIXER_PHONEIN: /* phone mic: left aux1 only */ + levels = devc->levels[SOUND_MIXER_LINE1] & 0xff; + levels |= levels << 8; + break; + + default: + levels = 0; + } + + return levels; +} + +/* + * Waveartist specific mixer information. + */ +static const struct waveartist_mixer_info netwinder_mixer = { + .supported_devs = SOUND_MASK_VOLUME | SOUND_MASK_SYNTH | + SOUND_MASK_PCM | SOUND_MASK_SPEAKER | + SOUND_MASK_LINE | SOUND_MASK_MIC | + SOUND_MASK_IMIX | SOUND_MASK_LINE1 | + SOUND_MASK_PHONEIN | SOUND_MASK_PHONEOUT| + SOUND_MASK_IGAIN, + + .recording_devs = SOUND_MASK_LINE | SOUND_MASK_MIC | + SOUND_MASK_IMIX | SOUND_MASK_LINE1 | + SOUND_MASK_PHONEIN, + + .stereo_devs = SOUND_MASK_VOLUME | SOUND_MASK_SYNTH | + SOUND_MASK_PCM | SOUND_MASK_LINE | + SOUND_MASK_IMIX | SOUND_MASK_IGAIN, + + .select_input = netwinder_select_input, + .decode_mixer = netwinder_decode_mixer, + .get_mixer = netwinder_get_mixer, +}; + +static void +vnc_configure_mixer(wavnc_info *devc, unsigned int recmask) +{ + if (!devc->no_autoselect) { + if (devc->handset_detect) { + recmask = SOUND_MASK_LINE1; + devc->spkr_mute_state = devc->line_mute_state = 1; + } else if (devc->telephone_detect) { + recmask = SOUND_MASK_PHONEIN; + devc->spkr_mute_state = devc->line_mute_state = 1; + } else { + /* unless someone has asked for LINE-IN, + * we default to MIC + */ + if ((devc->recmask & SOUND_MASK_LINE) == 0) + devc->recmask = SOUND_MASK_MIC; + devc->spkr_mute_state = devc->line_mute_state = 0; + } + vnc_mute_spkr(devc); + vnc_mute_lout(devc); + + if (recmask != devc->recmask) + waveartist_set_recmask(devc, recmask); + } +} + +static int +vnc_slider(wavnc_info *devc) +{ + signed int slider_volume; + unsigned int temp, old_hs, old_td; + + /* + * read the "buttons" state. + * Bit 4 = 0 means handset present + * Bit 5 = 1 means phone offhook + */ + temp = inb(0x201); + + old_hs = devc->handset_detect; + old_td = devc->telephone_detect; + + devc->handset_detect = !(temp & 0x10); + devc->telephone_detect = !!(temp & 0x20); + + if (!devc->no_autoselect && + (old_hs != devc->handset_detect || + old_td != devc->telephone_detect)) + vnc_configure_mixer(devc, devc->recmask); + + slider_volume = vnc_volume_slider(devc); + + /* + * If we're using software controlled volume, and + * the slider moves by more than 20%, then we + * switch back to slider controlled volume. + */ + if (abs(devc->slider_vol - slider_volume) > 20) + devc->use_slider = 1; + + /* + * use only left channel + */ + temp = levels[SOUND_MIXER_VOLUME] & 0xFF; + + if (slider_volume != temp && devc->use_slider) { + devc->slider_vol = slider_volume; + + waveartist_set_mixer(devc, SOUND_MIXER_VOLUME, + slider_volume | slider_volume << 8); + + return 1; + } + + return 0; +} + +static void +vnc_slider_tick(unsigned long data) +{ + int next_timeout; + + if (vnc_slider(adev_info + data)) + next_timeout = 5; // mixer reported change + else + next_timeout = VNC_TIMER_PERIOD; + + mod_timer(&vnc_timer, jiffies + next_timeout); +} + +static int +vnc_private_ioctl(int dev, unsigned int cmd, int __user * arg) +{ + wavnc_info *devc = (wavnc_info *)audio_devs[dev]->devc; + int val; + + switch (cmd) { + case SOUND_MIXER_PRIVATE1: + { + u_int prev_spkr_mute, prev_line_mute, prev_auto_state; + int val; + + if (get_user(val, arg)) + return -EFAULT; + + /* check if parameter is logical */ + if (val & ~(VNC_MUTE_INTERNAL_SPKR | + VNC_MUTE_LINE_OUT | + VNC_DISABLE_AUTOSWITCH)) + return -EINVAL; + + prev_auto_state = devc->no_autoselect; + prev_spkr_mute = devc->spkr_mute_state; + prev_line_mute = devc->line_mute_state; + + devc->no_autoselect = (val & VNC_DISABLE_AUTOSWITCH) ? 1 : 0; + devc->spkr_mute_state = (val & VNC_MUTE_INTERNAL_SPKR) ? 1 : 0; + devc->line_mute_state = (val & VNC_MUTE_LINE_OUT) ? 1 : 0; + + if (prev_spkr_mute != devc->spkr_mute_state) + vnc_mute_spkr(devc); + + if (prev_line_mute != devc->line_mute_state) + vnc_mute_lout(devc); + + if (prev_auto_state != devc->no_autoselect) + vnc_configure_mixer(devc, devc->recmask); + + return 0; + } + + case SOUND_MIXER_PRIVATE2: + if (get_user(val, arg)) + return -EFAULT; + + switch (val) { +#define VNC_SOUND_PAUSE 0x53 //to pause the DSP +#define VNC_SOUND_RESUME 0x57 //to unpause the DSP + case VNC_SOUND_PAUSE: + waveartist_cmd1(devc, 0x16); + break; + + case VNC_SOUND_RESUME: + waveartist_cmd1(devc, 0x18); + break; + + default: + return -EINVAL; + } + return 0; + + /* private ioctl to allow bulk access to waveartist */ + case SOUND_MIXER_PRIVATE3: + { + unsigned long flags; + int mixer_reg[15], i, val; + + if (get_user(val, arg)) + return -EFAULT; + if (copy_from_user(mixer_reg, (void *)val, sizeof(mixer_reg))) + return -EFAULT; + + switch (mixer_reg[14]) { + case MIXER_PRIVATE3_RESET: + waveartist_mixer_reset(devc); + break; + + case MIXER_PRIVATE3_WRITE: + waveartist_cmd3(devc, WACMD_SET_MIXER, mixer_reg[0], mixer_reg[4]); + waveartist_cmd3(devc, WACMD_SET_MIXER, mixer_reg[1], mixer_reg[5]); + waveartist_cmd3(devc, WACMD_SET_MIXER, mixer_reg[2], mixer_reg[6]); + waveartist_cmd3(devc, WACMD_SET_MIXER, mixer_reg[3], mixer_reg[7]); + waveartist_cmd3(devc, WACMD_SET_MIXER, mixer_reg[8], mixer_reg[9]); + + waveartist_cmd3(devc, WACMD_SET_LEVEL, mixer_reg[10], mixer_reg[11]); + waveartist_cmd3(devc, WACMD_SET_LEVEL, mixer_reg[12], mixer_reg[13]); + break; + + case MIXER_PRIVATE3_READ: + spin_lock_irqsave(&waveartist_lock, flags); + + for (i = 0x30; i < 14 << 8; i += 1 << 8) + waveartist_cmd(devc, 1, &i, 1, mixer_reg + (i >> 8)); + + spin_unlock_irqrestore(&waveartist_lock, flags); + + if (copy_to_user((void *)val, mixer_reg, sizeof(mixer_reg))) + return -EFAULT; + break; + + default: + return -EINVAL; + } + return 0; + } + + /* read back the state from PRIVATE1 */ + case SOUND_MIXER_PRIVATE4: + val = (devc->spkr_mute_state ? VNC_MUTE_INTERNAL_SPKR : 0) | + (devc->line_mute_state ? VNC_MUTE_LINE_OUT : 0) | + (devc->handset_detect ? VNC_HANDSET_DETECT : 0) | + (devc->telephone_detect ? VNC_PHONE_DETECT : 0) | + (devc->no_autoselect ? VNC_DISABLE_AUTOSWITCH : 0); + + return put_user(val, arg) ? -EFAULT : 0; + } + + if (_SIOC_DIR(cmd) & _SIOC_WRITE) { + /* + * special case for master volume: if we + * received this call - switch from hw + * volume control to a software volume + * control, till the hw volume is modified + * to signal that user wants to be back in + * hardware... + */ + if ((cmd & 0xff) == SOUND_MIXER_VOLUME) + devc->use_slider = 0; + + /* speaker output */ + if ((cmd & 0xff) == SOUND_MIXER_SPEAKER) { + unsigned int val, l, r; + + if (get_user(val, arg)) + return -EFAULT; + + l = val & 0x7f; + r = (val & 0x7f00) >> 8; + val = (l + r) / 2; + devc->levels[SOUND_MIXER_SPEAKER] = val | (val << 8); + devc->spkr_mute_state = (val <= 50); + vnc_mute_spkr(devc); + return 0; + } + } + + return -ENOIOCTLCMD; +} + +#endif + +static struct address_info cfg; + +static int attached; + +static int __initdata io = 0; +static int __initdata irq = 0; +static int __initdata dma = 0; +static int __initdata dma2 = 0; + + +static int __init init_waveartist(void) +{ + const struct waveartist_mixer_info *mix; + + if (!io && machine_is_netwinder()) { + /* + * The NetWinder WaveArtist is at a fixed address. + * If the user does not supply an address, use the + * well-known parameters. + */ + io = 0x250; + irq = 12; + dma = 3; + dma2 = 7; + } + + mix = &waveartist_mixer; +#ifdef CONFIG_ARCH_NETWINDER + if (machine_is_netwinder()) + mix = &netwinder_mixer; +#endif + + cfg.io_base = io; + cfg.irq = irq; + cfg.dma = dma; + cfg.dma2 = dma2; + + if (!probe_waveartist(&cfg)) + return -ENODEV; + + attach_waveartist(&cfg, mix); + attached = 1; + + return 0; +} + +static void __exit cleanup_waveartist(void) +{ + if (attached) + unload_waveartist(&cfg); +} + +module_init(init_waveartist); +module_exit(cleanup_waveartist); + +#ifndef MODULE +static int __init setup_waveartist(char *str) +{ + /* io, irq, dma, dma2 */ + int ints[5]; + + str = get_options(str, ARRAY_SIZE(ints), ints); + + io = ints[1]; + irq = ints[2]; + dma = ints[3]; + dma2 = ints[4]; + + return 1; +} +__setup("waveartist=", setup_waveartist); +#endif + +MODULE_DESCRIPTION("Rockwell WaveArtist RWA-010 sound driver"); +module_param(io, int, 0); /* IO base */ +module_param(irq, int, 0); /* IRQ */ +module_param(dma, int, 0); /* DMA */ +module_param(dma2, int, 0); /* DMA2 */ +MODULE_LICENSE("GPL"); diff --git a/sound/oss/waveartist.h b/sound/oss/waveartist.h new file mode 100644 index 00000000..dac4ca91 --- /dev/null +++ b/sound/oss/waveartist.h @@ -0,0 +1,92 @@ +/* + * linux/sound/oss/waveartist.h + * + * def file for Rockwell RWA010 chip set, as installed in Rebel.com NetWinder + */ + +//registers +#define CMDR 0 +#define DATR 2 +#define CTLR 4 +#define STATR 5 +#define IRQSTAT 12 + +//bit defs +//reg STATR +#define CMD_WE 0x80 +#define CMD_RF 0x40 +#define DAT_WE 0x20 +#define DAT_RF 0x10 + +#define IRQ_REQ 0x08 +#define DMA1 0x04 +#define DMA0 0x02 + +//bit defs +//reg CTLR +#define CMD_WEIE 0x80 +#define CMD_RFIE 0x40 +#define DAT_WEIE 0x20 +#define DAT_RFIE 0x10 + +#define RESET 0x08 +#define DMA1_IE 0x04 +#define DMA0_IE 0x02 +#define IRQ_ACK 0x01 + +//commands + +#define WACMD_SYSTEMID 0x00 +#define WACMD_GETREV 0x00 +#define WACMD_INPUTFORMAT 0x10 //0-8S, 1-16S, 2-8U +#define WACMD_INPUTCHANNELS 0x11 //1-Mono, 2-Stereo +#define WACMD_INPUTSPEED 0x12 //sampling rate +#define WACMD_INPUTDMA 0x13 //0-8bit, 1-16bit, 2-PIO +#define WACMD_INPUTSIZE 0x14 //samples to interrupt +#define WACMD_INPUTSTART 0x15 //start ADC +#define WACMD_INPUTPAUSE 0x16 //pause ADC +#define WACMD_INPUTSTOP 0x17 //stop ADC +#define WACMD_INPUTRESUME 0x18 //resume ADC +#define WACMD_INPUTPIO 0x19 //PIO ADC + +#define WACMD_OUTPUTFORMAT 0x20 //0-8S, 1-16S, 2-8U +#define WACMD_OUTPUTCHANNELS 0x21 //1-Mono, 2-Stereo +#define WACMD_OUTPUTSPEED 0x22 //sampling rate +#define WACMD_OUTPUTDMA 0x23 //0-8bit, 1-16bit, 2-PIO +#define WACMD_OUTPUTSIZE 0x24 //samples to interrupt +#define WACMD_OUTPUTSTART 0x25 //start ADC +#define WACMD_OUTPUTPAUSE 0x26 //pause ADC +#define WACMD_OUTPUTSTOP 0x27 //stop ADC +#define WACMD_OUTPUTRESUME 0x28 //resume ADC +#define WACMD_OUTPUTPIO 0x29 //PIO ADC + +#define WACMD_GET_LEVEL 0x30 +#define WACMD_SET_LEVEL 0x31 +#define WACMD_SET_MIXER 0x32 +#define WACMD_RST_MIXER 0x33 +#define WACMD_SET_MONO 0x34 + +/* + * Definitions for left/right recording input mux + */ +#define ADC_MUX_NONE 0 +#define ADC_MUX_MIXER 1 +#define ADC_MUX_LINE 2 +#define ADC_MUX_AUX2 3 +#define ADC_MUX_AUX1 4 +#define ADC_MUX_MIC 5 + +/* + * Definitions for mixer gain settings + */ +#define MIX_GAIN_LINE 0 /* line in */ +#define MIX_GAIN_AUX1 1 /* aux1 */ +#define MIX_GAIN_AUX2 2 /* aux2 */ +#define MIX_GAIN_XMIC 3 /* crossover mic */ +#define MIX_GAIN_MIC 4 /* normal mic */ +#define MIX_GAIN_PREMIC 5 /* preamp mic */ +#define MIX_GAIN_OUT 6 /* output */ +#define MIX_GAIN_MONO 7 /* mono in */ + +int wa_sendcmd(unsigned int cmd); +int wa_writecmd(unsigned int cmd, unsigned int arg); |