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https://github.com/UberGames/lilium-voyager.git
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274 lines
6.9 KiB
C
274 lines
6.9 KiB
C
/*
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===========================================================================
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Copyright (C) 1999-2005 Id Software, Inc.
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This file is part of Quake III Arena source code.
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Quake III Arena source code is free software; you can redistribute it
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and/or modify it under the terms of the GNU General Public License as
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published by the Free Software Foundation; either version 2 of the License,
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or (at your option) any later version.
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Quake III Arena source code is distributed in the hope that it will be
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useful, but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with Quake III Arena source code; if not, write to the Free Software
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Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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===========================================================================
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*/
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/*****************************************************************************
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* name: snd_mem.c
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*
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* desc: sound caching
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*
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* $Archive: /MissionPack/code/client/snd_mem.c $
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*
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*****************************************************************************/
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#include "snd_local.h"
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#include "snd_codec.h"
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#define DEF_COMSOUNDMEGS "8"
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/*
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===============================================================================
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memory management
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===============================================================================
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*/
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static sndBuffer *buffer = NULL;
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static sndBuffer *freelist = NULL;
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static int inUse = 0;
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static int totalInUse = 0;
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short *sfxScratchBuffer = NULL;
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sfx_t *sfxScratchPointer = NULL;
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int sfxScratchIndex = 0;
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void SND_free(sndBuffer *v) {
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*(sndBuffer **)v = freelist;
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freelist = (sndBuffer*)v;
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inUse += sizeof(sndBuffer);
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}
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sndBuffer* SND_malloc(void) {
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sndBuffer *v;
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redo:
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if (freelist == NULL) {
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S_FreeOldestSound();
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goto redo;
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}
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inUse -= sizeof(sndBuffer);
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totalInUse += sizeof(sndBuffer);
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v = freelist;
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freelist = *(sndBuffer **)freelist;
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v->next = NULL;
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return v;
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}
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void SND_setup(void) {
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sndBuffer *p, *q;
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cvar_t *cv;
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int scs;
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cv = Cvar_Get( "com_soundMegs", DEF_COMSOUNDMEGS, CVAR_LATCH | CVAR_ARCHIVE );
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scs = (cv->integer*1536);
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buffer = malloc(scs*sizeof(sndBuffer) );
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// allocate the stack based hunk allocator
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sfxScratchBuffer = malloc(SND_CHUNK_SIZE * sizeof(short) * 4); //Hunk_Alloc(SND_CHUNK_SIZE * sizeof(short) * 4);
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sfxScratchPointer = NULL;
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inUse = scs*sizeof(sndBuffer);
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p = buffer;;
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q = p + scs;
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while (--q > p)
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*(sndBuffer **)q = q-1;
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*(sndBuffer **)q = NULL;
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freelist = p + scs - 1;
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Com_Printf("Sound memory manager started\n");
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}
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void SND_shutdown(void)
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{
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free(sfxScratchBuffer);
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free(buffer);
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}
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/*
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================
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ResampleSfx
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resample / decimate to the current source rate
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================
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*/
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static int ResampleSfx( sfx_t *sfx, int channels, int inrate, int inwidth, int samples, byte *data, qboolean compressed ) {
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int outcount;
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int srcsample;
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float stepscale;
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int i, j;
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int sample, samplefrac, fracstep;
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int part;
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sndBuffer *chunk;
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stepscale = (float)inrate / dma.speed; // this is usually 0.5, 1, or 2
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outcount = samples / stepscale;
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samplefrac = 0;
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fracstep = stepscale * 256 * channels;
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chunk = sfx->soundData;
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for (i=0 ; i<outcount ; i++)
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{
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srcsample = samplefrac >> 8;
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samplefrac += fracstep;
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for (j=0 ; j<channels ; j++)
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{
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if( inwidth == 2 ) {
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sample = ( ((short *)data)[srcsample+j] );
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} else {
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sample = (int)( (unsigned char)(data[srcsample+j]) - 128) << 8;
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}
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part = (i*channels+j)&(SND_CHUNK_SIZE-1);
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if (part == 0) {
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sndBuffer *newchunk;
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newchunk = SND_malloc();
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if (chunk == NULL) {
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sfx->soundData = newchunk;
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} else {
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chunk->next = newchunk;
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}
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chunk = newchunk;
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}
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chunk->sndChunk[part] = sample;
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}
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}
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return outcount;
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}
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/*
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================
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ResampleSfx
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resample / decimate to the current source rate
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================
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*/
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static int ResampleSfxRaw( short *sfx, int channels, int inrate, int inwidth, int samples, byte *data ) {
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int outcount;
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int srcsample;
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float stepscale;
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int i, j;
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int sample, samplefrac, fracstep;
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stepscale = (float)inrate / dma.speed; // this is usually 0.5, 1, or 2
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outcount = samples / stepscale;
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samplefrac = 0;
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fracstep = stepscale * 256 * channels;
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for (i=0 ; i<outcount ; i++)
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{
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srcsample = samplefrac >> 8;
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samplefrac += fracstep;
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for (j=0 ; j<channels ; j++)
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{
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if( inwidth == 2 ) {
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sample = LittleShort ( ((short *)data)[srcsample+j] );
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} else {
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sample = (int)( (unsigned char)(data[srcsample+j]) - 128) << 8;
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}
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sfx[i*channels+j] = sample;
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}
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}
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return outcount;
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}
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//=============================================================================
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/*
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==============
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S_LoadSound
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The filename may be different than sfx->name in the case
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of a forced fallback of a player specific sound
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==============
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*/
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qboolean S_LoadSound( sfx_t *sfx )
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{
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byte *data;
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short *samples;
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snd_info_t info;
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// int size;
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// load it in
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data = S_CodecLoad(sfx->soundName, &info);
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if(!data)
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return qfalse;
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if ( info.width == 1 ) {
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Com_DPrintf(S_COLOR_YELLOW "WARNING: %s is a 8 bit audio file\n", sfx->soundName);
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}
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if ( info.rate != 22050 ) {
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Com_DPrintf(S_COLOR_YELLOW "WARNING: %s is not a 22kHz audio file\n", sfx->soundName);
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}
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samples = Hunk_AllocateTempMemory(info.channels * info.samples * sizeof(short) * 2);
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sfx->lastTimeUsed = Com_Milliseconds()+1;
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// each of these compression schemes works just fine
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// but the 16bit quality is much nicer and with a local
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// install assured we can rely upon the sound memory
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// manager to do the right thing for us and page
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// sound in as needed
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if( info.channels == 1 && sfx->soundCompressed == qtrue) {
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sfx->soundCompressionMethod = 1;
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sfx->soundData = NULL;
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sfx->soundLength = ResampleSfxRaw( samples, info.channels, info.rate, info.width, info.samples, data + info.dataofs );
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S_AdpcmEncodeSound(sfx, samples);
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#if 0
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} else if (info.channels == 1 && info.samples>(SND_CHUNK_SIZE*16) && info.width >1) {
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sfx->soundCompressionMethod = 3;
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sfx->soundData = NULL;
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sfx->soundLength = ResampleSfxRaw( samples, info.channels, info.rate, info.width, info.samples, (data + info.dataofs) );
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encodeMuLaw( sfx, samples);
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} else if (info.channels == 1 && info.samples>(SND_CHUNK_SIZE*6400) && info.width >1) {
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sfx->soundCompressionMethod = 2;
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sfx->soundData = NULL;
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sfx->soundLength = ResampleSfxRaw( samples, info.channels, info.rate, info.width, info.samples, (data + info.dataofs) );
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encodeWavelet( sfx, samples);
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#endif
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} else {
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sfx->soundCompressionMethod = 0;
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sfx->soundData = NULL;
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sfx->soundLength = ResampleSfx( sfx, info.channels, info.rate, info.width, info.samples, data + info.dataofs, qfalse );
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}
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sfx->soundChannels = info.channels;
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Hunk_FreeTempMemory(samples);
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Hunk_FreeTempMemory(data);
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return qtrue;
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}
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void S_DisplayFreeMemory(void) {
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Com_Printf("%d bytes free sound buffer memory, %d total used\n", inUse, totalInUse);
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}
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