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Diffstat (limited to 'demos/3rdparty/doom/i_sound.c')
| -rw-r--r-- | demos/3rdparty/doom/i_sound.c | 820 |
1 files changed, 0 insertions, 820 deletions
diff --git a/demos/3rdparty/doom/i_sound.c b/demos/3rdparty/doom/i_sound.c deleted file mode 100644 index 48ee6d19..00000000 --- a/demos/3rdparty/doom/i_sound.c +++ /dev/null @@ -1,820 +0,0 @@ -// Emacs style mode select -*- C++ -*- -//----------------------------------------------------------------------------- -// -// $Id:$ -// -// Copyright (C) 1993-1996 by id Software, Inc. -// -// This source is available for distribution and/or modification -// only under the terms of the DOOM Source Code License as -// published by id Software. All rights reserved. -// -// The source is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// FITNESS FOR A PARTICULAR PURPOSE. See the DOOM Source Code License -// for more details. -// -// $Log:$ -// -// DESCRIPTION: -// System interface for sound. -// -//----------------------------------------------------------------------------- - -#include "gfx.h" - -#include "z_zone.h" - -#include "i_system.h" -#include "i_sound.h" -#include "m_argv.h" -#include "m_misc.h" -#include "w_wad.h" - -#include "doomdef.h" - -// A quick hack to establish a protocol between -// synchronous mix buffer updates and asynchronous -// audio writes. Probably redundant with gametic. -static int flag = 0; - -// The number of internal mixing channels, -// the samples calculated for each mixing step, -// the size of the 16bit, 2 hardware channel (stereo) -// mixing buffer, and the samplerate of the raw data. - - -// Needed for calling the actual sound output. -#define SAMPLECOUNT 512 -#define NUM_CHANNELS 8 -// It is 2 for 16bit, and 2 for two channels. -#define BUFMUL 4 -#define MIXBUFFERSIZE (SAMPLECOUNT*BUFMUL) - -#define SAMPLERATE 11025 // Hz -#define SAMPLESIZE 2 // 16bit - -// The actual lengths of all sound effects. -int lengths[NUMSFX]; - -// The actual output device. -int audio_fd; - -// The global mixing buffer. -// Basically, samples from all active internal channels -// are modifed and added, and stored in the buffer -// that is submitted to the audio device. -signed short mixbuffer[MIXBUFFERSIZE]; - - -// The channel step amount... -unsigned int channelstep[NUM_CHANNELS]; -// ... and a 0.16 bit remainder of last step. -unsigned int channelstepremainder[NUM_CHANNELS]; - - -// The channel data pointers, start and end. -unsigned char* channels[NUM_CHANNELS]; -unsigned char* channelsend[NUM_CHANNELS]; - - -// Time/gametic that the channel started playing, -// used to determine oldest, which automatically -// has lowest priority. -// In case number of active sounds exceeds -// available channels. -int channelstart[NUM_CHANNELS]; - -// The sound in channel handles, -// determined on registration, -// might be used to unregister/stop/modify, -// currently unused. -int channelhandles[NUM_CHANNELS]; - -// SFX id of the playing sound effect. -// Used to catch duplicates (like chainsaw). -int channelids[NUM_CHANNELS]; - -// Pitch to stepping lookup, unused. -int steptable[256]; - -// Volume lookups. -int vol_lookup[128*256]; - -// Hardware left and right channel volume lookup. -int* channelleftvol_lookup[NUM_CHANNELS]; -int* channelrightvol_lookup[NUM_CHANNELS]; - - -// -// This function loads the sound data from the WAD lump, -// for single sound. -// -void* -getsfx -( char* sfxname, - int* len ) -{ - unsigned char* sfx; - unsigned char* paddedsfx; - int i; - int size; - int paddedsize; - char name[20]; - int sfxlump; - - - // Get the sound data from the WAD, allocate lump - // in zone memory. - I_sprintf(name, "ds%s", sfxname); - - // Now, there is a severe problem with the - // sound handling, in it is not (yet/anymore) - // gamemode aware. That means, sounds from - // DOOM II will be requested even with DOOM - // shareware. - // The sound list is wired into sounds.c, - // which sets the external variable. - // I do not do runtime patches to that - // variable. Instead, we will use a - // default sound for replacement. - if ( W_CheckNumForName(name) == -1 ) - sfxlump = W_GetNumForName("dspistol"); - else - sfxlump = W_GetNumForName(name); - - size = W_LumpLength( sfxlump ); - - // Debug. - // I_DBGprintf( "." ); - //I_DBGprintf( " -loading %s (lump %d, %d bytes)\n", - // sfxname, sfxlump, size ); - - sfx = (unsigned char*)W_CacheLumpNum( sfxlump, PU_STATIC ); - - // Pads the sound effect out to the mixing buffer size. - // The original realloc would interfere with zone memory. - paddedsize = ((size-8 + (SAMPLECOUNT-1)) / SAMPLECOUNT) * SAMPLECOUNT; - - // Allocate from zone memory. - paddedsfx = (unsigned char*)Z_Malloc( paddedsize+8, PU_STATIC, 0 ); - // ddt: (unsigned char *) I_Realloc(sfx, paddedsize+8); - // This should interfere with zone memory handling, - // which does not kick in in the soundserver. - - // Now copy and pad. - memcpy( paddedsfx, sfx, size ); - for (i=size ; i<paddedsize+8 ; i++) - paddedsfx[i] = 128; - - // Remove the cached lump. - Z_Free( sfx ); - - // Preserve padded length. - *len = paddedsize; - - // Return allocated padded data. - return (void *) (paddedsfx + 8); -} - - - - - -// -// This function adds a sound to the -// list of currently active sounds, -// which is maintained as a given number -// (eight, usually) of internal channels. -// Returns a handle. -// -int -addsfx -( int sfxid, - int volume, - int step, - int seperation ) -{ - static unsigned short handlenums = 0; - - int i; - int rc = -1; - - int oldest = gametic; - int oldestnum = 0; - int slot; - - int rightvol; - int leftvol; - - // Chainsaw troubles. - // Play these sound effects only one at a time. - if ( sfxid == sfx_sawup - || sfxid == sfx_sawidl - || sfxid == sfx_sawful - || sfxid == sfx_sawhit - || sfxid == sfx_stnmov - || sfxid == sfx_pistol ) - { - // Loop all channels, check. - for (i=0 ; i<NUM_CHANNELS ; i++) - { - // Active, and using the same SFX? - if ( (channels[i]) - && (channelids[i] == sfxid) ) - { - // Reset. - channels[i] = 0; - // We are sure that iff, - // there will only be one. - break; - } - } - } - - // Loop all channels to find oldest SFX. - for (i=0; (i<NUM_CHANNELS) && (channels[i]); i++) - { - if (channelstart[i] < oldest) - { - oldestnum = i; - oldest = channelstart[i]; - } - } - - // Tales from the cryptic. - // If we found a channel, fine. - // If not, we simply overwrite the first one, 0. - // Probably only happens at startup. - if (i == NUM_CHANNELS) - slot = oldestnum; - else - slot = i; - - // Okay, in the less recent channel, - // we will handle the new SFX. - // Set pointer to raw data. - channels[slot] = (unsigned char *) S_sfx[sfxid].data; - // Set pointer to end of raw data. - channelsend[slot] = channels[slot] + lengths[sfxid]; - - // Reset current handle number, limited to 0..100. - if (!handlenums) - handlenums = 100; - - // Assign current handle number. - // Preserved so sounds could be stopped (unused). - channelhandles[slot] = rc = handlenums++; - - // Set stepping??? - // Kinda getting the impression this is never used. - channelstep[slot] = step; - // ??? - channelstepremainder[slot] = 0; - // Should be gametic, I presume. - channelstart[slot] = gametic; - - // Separation, that is, orientation/stereo. - // range is: 1 - 256 - seperation += 1; - - // Per left/right channel. - // x^2 seperation, - // adjust volume properly. - leftvol = - volume - ((volume*seperation*seperation) >> 16); ///(256*256); - seperation = seperation - 257; - rightvol = - volume - ((volume*seperation*seperation) >> 16); - - // Sanity check, clamp volume. - if (rightvol < 0 || rightvol > 127) - I_Error("rightvol out of bounds"); - - if (leftvol < 0 || leftvol > 127) - I_Error("leftvol out of bounds"); - - // Get the proper lookup table piece - // for this volume level??? - channelleftvol_lookup[slot] = &vol_lookup[leftvol*256]; - channelrightvol_lookup[slot] = &vol_lookup[rightvol*256]; - - // Preserve sound SFX id, - // e.g. for avoiding duplicates of chainsaw. - channelids[slot] = sfxid; - - // You tell me. - return rc; -} - -// -// SFX API -// Note: this was called by S_Init. -// However, whatever they did in the -// old DPMS based DOS version, this -// were simply dummies in the Linux -// version. -// See soundserver initdata(). -// -void I_SetChannels() -{ -#if 0 - // Init internal lookups (raw data, mixing buffer, channels). - // This function sets up internal lookups used during - // the mixing process. - int i; - int j; - - int* steptablemid = steptable + 128; - - // Okay, reset internal mixing channels to zero. - /*for (i=0; i<NUM_CHANNELS; i++) - { - channels[i] = 0; - }*/ - - // This table provides step widths for pitch parameters. - // I fail to see that this is currently used. - for (i=-128 ; i<128 ; i++) - steptablemid[i] = (int)(pow(2.0, (i/64.0))*65536.0); - - - // Generates volume lookup tables - // which also turn the unsigned samples - // into signed samples. - for (i=0 ; i<128 ; i++) - for (j=0 ; j<256 ; j++) - vol_lookup[i*256+j] = (i*(j-128)*256)/127; -#endif -} - - -void I_SetSfxVolume(int volume) -{ -#if 0 - // Identical to DOS. - // Basically, this should propagate - // the menu/config file setting - // to the state variable used in - // the mixing. - snd_SfxVolume = volume; -#endif -} - -// MUSIC API - dummy. Some code from DOS version. -void I_SetMusicVolume(int volume) -{ -#if 0 - // Internal state variable. - snd_MusicVolume = volume; - // Now set volume on output device. - // Whatever( snd_MusciVolume ); -#endif -} - - -// -// Retrieve the raw data lump index -// for a given SFX name. -// -int I_GetSfxLumpNum(sfxinfo_t* sfx) -{ - char namebuf[9]; - I_sprintf(namebuf, "ds%s", sfx->name); - return W_GetNumForName(namebuf); -} - -// -// Starting a sound means adding it -// to the current list of active sounds -// in the internal channels. -// As the SFX info struct contains -// e.g. a pointer to the raw data, -// it is ignored. -// As our sound handling does not handle -// priority, it is ignored. -// Pitching (that is, increased speed of playback) -// is set, but currently not used by mixing. -// -int -I_StartSound -( int id, - int vol, - int sep, - int pitch, - int priority ) -{ -#if 0 - // UNUSED - priority = 0; - -#ifdef SNDSERV - if (sndserver) - { - fprintf(sndserver, "p%2.2x%2.2x%2.2x%2.2x\n", id, pitch, vol, sep); - fflush(sndserver); - } - // warning: control reaches end of non-void function. - return id; -#else - // Debug. - //I_DBGprintf("starting sound %d", id ); - - // Returns a handle (not used). - id = addsfx( id, vol, steptable[pitch], sep ); - - // I_DBGprintf("/handle is %d\n", id ); - - return id; -#endif - -#else - return id; -#endif -} - - - -void I_StopSound (int handle) -{ - // You need the handle returned by StartSound. - // Would be looping all channels, - // tracking down the handle, - // an setting the channel to zero. - - // UNUSED. - handle = 0; -} - - -int I_SoundIsPlaying(int handle) -{ -#if 0 - // Ouch. - return gametic < handle; -#else - return 0; -#endif -} - -// -// This function loops all active (internal) sound -// channels, retrieves a given number of samples -// from the raw sound data, modifies it according -// to the current (internal) channel parameters, -// mixes the per channel samples into the global -// mixbuffer, clamping it to the allowed range, -// and sets up everything for transferring the -// contents of the mixbuffer to the (two) -// hardware channels (left and right, that is). -// -// This function currently supports only 16bit. -// -void I_UpdateSound( void ) -{ -#if 0 - -#ifdef SNDINTR - // Debug. Count buffer misses with interrupt. - static int misses = 0; -#endif - - - // Mix current sound data. - // Data, from raw sound, for right and left. - register unsigned int sample; - register int dl; - register int dr; - - // Pointers in global mixbuffer, left, right, end. - signed short* leftout; - signed short* rightout; - signed short* leftend; - // Step in mixbuffer, left and right, thus two. - int step; - - // Mixing channel index. - int chan; - - // Left and right channel - // are in global mixbuffer, alternating. - leftout = mixbuffer; - rightout = mixbuffer+1; - step = 2; - - // Determine end, for left channel only - // (right channel is implicit). - leftend = mixbuffer + SAMPLECOUNT*step; - - // Mix sounds into the mixing buffer. - // Loop over step*SAMPLECOUNT, - // that is 512 values for two channels. - while (leftout != leftend) - { - // Reset left/right value. - dl = 0; - dr = 0; - - // Love thy L2 chache - made this a loop. - // Now more channels could be set at compile time - // as well. Thus loop those channels. - for ( chan = 0; chan < NUM_CHANNELS; chan++ ) - { - // Check channel, if active. - if (channels[ chan ]) - { - // Get the raw data from the channel. - sample = *channels[ chan ]; - // Add left and right part - // for this channel (sound) - // to the current data. - // Adjust volume accordingly. - dl += channelleftvol_lookup[ chan ][sample]; - dr += channelrightvol_lookup[ chan ][sample]; - // Increment index ??? - channelstepremainder[ chan ] += channelstep[ chan ]; - // MSB is next sample??? - channels[ chan ] += channelstepremainder[ chan ] >> 16; - // Limit to LSB??? - channelstepremainder[ chan ] &= 65536-1; - - // Check whether we are done. - if (channels[ chan ] >= channelsend[ chan ]) - channels[ chan ] = 0; - } - } - - // Clamp to range. Left hardware channel. - // Has been char instead of short. - // if (dl > 127) *leftout = 127; - // else if (dl < -128) *leftout = -128; - // else *leftout = dl; - - if (dl > 0x7fff) - *leftout = 0x7fff; - else if (dl < -0x8000) - *leftout = -0x8000; - else - *leftout = dl; - - // Same for right hardware channel. - if (dr > 0x7fff) - *rightout = 0x7fff; - else if (dr < -0x8000) - *rightout = -0x8000; - else - *rightout = dr; - - // Increment current pointers in mixbuffer. - leftout += step; - rightout += step; - } - -#ifdef SNDINTR - // Debug check. - if ( flag ) - { - misses += flag; - flag = 0; - } - - if ( misses > 10 ) - { - I_DBGprintf("I_SoundUpdate: missed 10 buffer writes\n"); - misses = 0; - } - - // Increment flag for update. - flag++; -#endif - -#endif -} - - -// -// This would be used to write out the mixbuffer -// during each game loop update. -// Updates sound buffer and audio device at runtime. -// It is called during Timer interrupt with SNDINTR. -// Mixing now done synchronous, and -// only output be done asynchronous? -// -void -I_SubmitSound(void) -{ -#if 0 - // Write it to DSP device. - write(audio_fd, mixbuffer, SAMPLECOUNT*BUFMUL); -#endif -} - - - -void -I_UpdateSoundParams -( int handle, - int vol, - int sep, - int pitch) -{ -#if 0 - // I fail too see that this is used. - // Would be using the handle to identify - // on which channel the sound might be active, - // and resetting the channel parameters. - - // UNUSED. - handle = vol = sep = pitch = 0; -#endif -} - - - - -void I_ShutdownSound(void) -{ -} - -void I_InitSound() -{ -#if 0 - -#ifdef SNDSERV - char buffer[256]; - - if (getenv("DOOMWADDIR")) - I_sprintf(buffer, "%s/%s", - getenv("DOOMWADDIR"), - sndserver_filename); - else - I_sprintf(buffer, "%s", sndserver_filename); - - // start sound process - if ( !access(buffer, X_OK) ) - { - strcat(buffer, " -quiet"); - sndserver = popen(buffer, "w"); - } - else - I_DBGprintf("Could not start sound server [%s]\n", buffer); -#else - - int i; - -#ifdef SNDINTR - I_DBGprintf("I_SoundSetTimer: %d microsecs\n", SOUND_INTERVAL ); - I_SoundSetTimer( SOUND_INTERVAL ); -#endif - - // Secure and configure sound device first. - I_DBGprintf( "I_InitSound: "); - - audio_fd = open("/dev/dsp", O_WRONLY); - if (audio_fd<0) - I_DBGprintf( "Could not open /dev/dsp\n"); - - - i = 11 | (2<<16); - myioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &i); - myioctl(audio_fd, SNDCTL_DSP_RESET, 0); - - i=SAMPLERATE; - - myioctl(audio_fd, SNDCTL_DSP_SPEED, &i); - - i=1; - myioctl(audio_fd, SNDCTL_DSP_STEREO, &i); - - myioctl(audio_fd, SNDCTL_DSP_GETFMTS, &i); - - if (i&=AFMT_S16_LE) - myioctl(audio_fd, SNDCTL_DSP_SETFMT, &i); - else - I_DBGprintf("Could not play signed 16 data\n"); - - I_DBGprintf(" configured audio device\n" ); - - - // Initialize external data (all sounds) at start, keep static. - I_DBGprintf("I_InitSound: "); - - for (i=1 ; i<NUMSFX ; i++) - { - // Alias? Example is the chaingun sound linked to pistol. - if (!S_sfx[i].link) - { - // Load data from WAD file. - S_sfx[i].data = getsfx( S_sfx[i].name, &lengths[i] ); - } - else - { - // Previously loaded already? - S_sfx[i].data = S_sfx[i].link->data; - lengths[i] = lengths[(S_sfx[i].link - S_sfx)/sizeof(sfxinfo_t)]; - } - } - - I_DBGprintf(" pre-cached all sound data\n"); - - // Now initialize mixbuffer with zero. - for ( i = 0; i< MIXBUFFERSIZE; i++ ) - mixbuffer[i] = 0; - - // Finished initialization. - I_DBGprintf("I_InitSound: sound module ready\n"); - -#endif - -#endif -} - - - - -// -// MUSIC API. -// Still no music done. -// Remains. Dummies. -// -void I_InitMusic(void) { } -void I_ShutdownMusic(void) { } - -static int looping=0; -static int musicdies=-1; - -void I_PlaySong(int handle, int looping) -{ - // UNUSED. - handle = looping = 0; - musicdies = gametic + TICRATE*30; -} - -void I_PauseSong (int handle) -{ - // UNUSED. - handle = 0; -} - -void I_ResumeSong (int handle) -{ - // UNUSED. - handle = 0; -} - -void I_StopSong(int handle) -{ - // UNUSED. - handle = 0; - - looping = 0; - musicdies = 0; -} - -void I_UnRegisterSong(int handle) -{ - // UNUSED. - handle = 0; -} - -int I_RegisterSong(void* data) -{ - // UNUSED. - data = NULL; - - return 1; -} - -// Is the song playing? -int I_QrySongPlaying(int handle) -{ - // UNUSED. - handle = 0; - return looping || musicdies > gametic; -} - -// Interrupt handler. -void I_HandleSoundTimer( int ignore ) -{ -#if 0 - // Debug. - //I_DBGprintf("%c", '+' ); fflush( stderr ); - - // Feed sound device if necesary. - if ( flag ) - { - // See I_SubmitSound(). - // Write it to DSP device. - write(audio_fd, mixbuffer, SAMPLECOUNT*BUFMUL); - - // Reset flag counter. - flag = 0; - } - else - return; - - // UNUSED, but required. - ignore = 0; - return; -#endif -} |