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Fix playback of stereo sounds in Base sound system

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Already works correctly in OpenAL.
This commit is contained in:
Zack Middleton 2013-12-15 00:18:29 -06:00
parent 623d107f42
commit 1a86229538
3 changed files with 98 additions and 53 deletions
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82
code/client/snd_mem.c
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@ -113,47 +113,51 @@ ResampleSfx
resample / decimate to the current source rate
================
*/
static void ResampleSfx( sfx_t *sfx, int inrate, int inwidth, byte *data, qboolean compressed ) {
static int ResampleSfx( sfx_t *sfx, int channels, int inrate, int inwidth, int samples, byte *data, qboolean compressed ) {
int outcount;
int srcsample;
float stepscale;
int i;
int i, j;
int sample, samplefrac, fracstep;
int part;
sndBuffer *chunk;
stepscale = (float)inrate / dma.speed; // this is usually 0.5, 1, or 2
outcount = sfx->soundLength / stepscale;
sfx->soundLength = outcount;
outcount = samples / stepscale;
samplefrac = 0;
fracstep = stepscale * 256;
fracstep = stepscale * 256 * channels;
chunk = sfx->soundData;
for (i=0 ; i<outcount ; i++)
{
srcsample = samplefrac >> 8;
samplefrac += fracstep;
if( inwidth == 2 ) {
sample = ( ((short *)data)[srcsample] );
} else {
sample = (int)( (unsigned char)(data[srcsample]) - 128) << 8;
}
part = (i&(SND_CHUNK_SIZE-1));
if (part == 0) {
sndBuffer *newchunk;
newchunk = SND_malloc();
if (chunk == NULL) {
sfx->soundData = newchunk;
for (j=0 ; j<channels ; j++)
{
if( inwidth == 2 ) {
sample = ( ((short *)data)[srcsample+j] );
} else {
chunk->next = newchunk;
sample = (int)( (unsigned char)(data[srcsample+j]) - 128) << 8;
}
part = (i*channels+j)&(SND_CHUNK_SIZE-1);
if (part == 0) {
sndBuffer *newchunk;
newchunk = SND_malloc();
if (chunk == NULL) {
sfx->soundData = newchunk;
} else {
chunk->next = newchunk;
}
chunk = newchunk;
}
chunk = newchunk;
}
chunk->sndChunk[part] = sample;
chunk->sndChunk[part] = sample;
}
}
return outcount;
}
/*
@ -163,11 +167,11 @@ ResampleSfx
resample / decimate to the current source rate
================
*/
static int ResampleSfxRaw( short *sfx, int inrate, int inwidth, int samples, byte *data ) {
static int ResampleSfxRaw( short *sfx, int channels, int inrate, int inwidth, int samples, byte *data ) {
int outcount;
int srcsample;
float stepscale;
int i;
int i, j;
int sample, samplefrac, fracstep;
stepscale = (float)inrate / dma.speed; // this is usually 0.5, 1, or 2
@ -175,18 +179,21 @@ static int ResampleSfxRaw( short *sfx, int inrate, int inwidth, int samples, byt
outcount = samples / stepscale;
samplefrac = 0;
fracstep = stepscale * 256;
fracstep = stepscale * 256 * channels;
for (i=0 ; i<outcount ; i++)
{
srcsample = samplefrac >> 8;
samplefrac += fracstep;
if( inwidth == 2 ) {
sample = LittleShort ( ((short *)data)[srcsample] );
} else {
sample = (int)( (unsigned char)(data[srcsample]) - 128) << 8;
for (j=0 ; j<channels ; j++)
{
if( inwidth == 2 ) {
sample = LittleShort ( ((short *)data)[srcsample+j] );
} else {
sample = (int)( (unsigned char)(data[srcsample+j]) - 128) << 8;
}
sfx[i*channels+j] = sample;
}
sfx[i] = sample;
}
return outcount;
}
@ -221,7 +228,7 @@ qboolean S_LoadSound( sfx_t *sfx )
Com_DPrintf(S_COLOR_YELLOW "WARNING: %s is not a 22kHz audio file\n", sfx->soundName);
}
samples = Hunk_AllocateTempMemory(info.samples * sizeof(short) * 2);
samples = Hunk_AllocateTempMemory(info.channels * info.samples * sizeof(short) * 2);
sfx->lastTimeUsed = Com_Milliseconds()+1;
@ -231,29 +238,30 @@ qboolean S_LoadSound( sfx_t *sfx )
// manager to do the right thing for us and page
// sound in as needed
if( sfx->soundCompressed == qtrue) {
if( info.channels == 1 && sfx->soundCompressed == qtrue) {
sfx->soundCompressionMethod = 1;
sfx->soundData = NULL;
sfx->soundLength = ResampleSfxRaw( samples, info.rate, info.width, info.samples, data + info.dataofs );
sfx->soundLength = ResampleSfxRaw( samples, info.channels, info.rate, info.width, info.samples, data + info.dataofs );
S_AdpcmEncodeSound(sfx, samples);
#if 0
} else if (info.samples>(SND_CHUNK_SIZE*16) && info.width >1) {
} else if (info.channels == 1 && info.samples>(SND_CHUNK_SIZE*16) && info.width >1) {
sfx->soundCompressionMethod = 3;
sfx->soundData = NULL;
sfx->soundLength = ResampleSfxRaw( samples, info.rate, info.width, info.samples, (data + info.dataofs) );
sfx->soundLength = ResampleSfxRaw( samples, info.channels, info.rate, info.width, info.samples, (data + info.dataofs) );
encodeMuLaw( sfx, samples);
} else if (info.samples>(SND_CHUNK_SIZE*6400) && info.width >1) {
} else if (info.channels == 1 && info.samples>(SND_CHUNK_SIZE*6400) && info.width >1) {
sfx->soundCompressionMethod = 2;
sfx->soundData = NULL;
sfx->soundLength = ResampleSfxRaw( samples, info.rate, info.width, info.samples, (data + info.dataofs) );
sfx->soundLength = ResampleSfxRaw( samples, info.channels, info.rate, info.width, info.samples, (data + info.dataofs) );
encodeWavelet( sfx, samples);
#endif
} else {
sfx->soundCompressionMethod = 0;
sfx->soundLength = info.samples;
sfx->soundData = NULL;
ResampleSfx( sfx, info.rate, info.width, data + info.dataofs, qfalse );
sfx->soundLength = ResampleSfx( sfx, info.channels, info.rate, info.width, info.samples, data + info.dataofs, qfalse );
}
sfx->soundChannels = info.channels;
Hunk_FreeTempMemory(samples);
Hunk_FreeTempMemory(data);