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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 committed 2013-12-15 00:23:10 -06:00
1 parent 623d107f42
commit 1a86229538
3 files changed
+98 -53
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code/client/snd_mix.c
+52 -16
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@@ -234,7 +234,7 @@ static void S_PaintChannelFrom16_altivec( channel_t *ch, const sfx_t *sc, int co
portable_samplepair_t *samp;
sndBuffer *chunk;
short *samples;
float ooff, fdata, fdiv, fleftvol, frightvol;
float ooff, fdata[2], fdiv, fleftvol, frightvol;
samp = &paintbuffer[ bufferOffset ];
@@ -242,6 +242,14 @@ static void S_PaintChannelFrom16_altivec( channel_t *ch, const sfx_t *sc, int co
sampleOffset = sampleOffset*ch->oldDopplerScale;
}
if ( sc->soundChannels == 2 ) {
sampleOffset *= sc->soundChannels;
if ( sampleOffset & 1 ) {
sampleOffset &= ~1;
}
}
chunk = sc->soundData;
while (sampleOffset>=SND_CHUNK_SIZE) {
chunk = chunk->next;
@@ -274,6 +282,10 @@ static void S_PaintChannelFrom16_altivec( channel_t *ch, const sfx_t *sc, int co
while(i < count && (((unsigned long)&samp[i] & 0x1f) || ((count-i) < 8) || ((SND_CHUNK_SIZE - sampleOffset) < 8))) {
data = samples[sampleOffset++];
samp[i].left += (data * leftvol)>>8;
if ( sc->soundChannels == 2 ) {
data = samples[sampleOffset++];
}
samp[i].right += (data * rightvol)>>8;
if (sampleOffset == SND_CHUNK_SIZE) {
@@ -373,10 +385,10 @@ static void S_PaintChannelFrom16_altivec( channel_t *ch, const sfx_t *sc, int co
for ( i=0 ; i<count ; i++ ) {
aoff = ooff;
ooff = ooff + ch->dopplerScale;
ooff = ooff + ch->dopplerScale * sc->soundChannels;
boff = ooff;
fdata = 0;
for (j=aoff; j<boff; j++) {
fdata[0] = fdata[1] = 0;
for (j=aoff; j<boff; j += sc->soundChannels) {
if (j == SND_CHUNK_SIZE) {
chunk = chunk->next;
if (!chunk) {
@@ -385,11 +397,17 @@ static void S_PaintChannelFrom16_altivec( channel_t *ch, const sfx_t *sc, int co
samples = chunk->sndChunk;
ooff -= SND_CHUNK_SIZE;
}
fdata += samples[j&(SND_CHUNK_SIZE-1)];
if ( sc->soundChannels == 2 ) {
fdata[0] += samples[j&(SND_CHUNK_SIZE-1)];
fdata[1] += samples[(j+1)&(SND_CHUNK_SIZE-1)];
} else {
fdata[0] += samples[j&(SND_CHUNK_SIZE-1)];
fdata[1] += samples[j&(SND_CHUNK_SIZE-1)];
}
}
fdiv = 256 * (boff-aoff);
samp[i].left += (fdata * fleftvol)/fdiv;
samp[i].right += (fdata * frightvol)/fdiv;
fdiv = 256 * (boff-aoff) / sc->soundChannels;
samp[i].left += (fdata[0] * fleftvol)/fdiv;
samp[i].right += (fdata[1] * frightvol)/fdiv;
}
}
}
@@ -402,7 +420,7 @@ static void S_PaintChannelFrom16_scalar( channel_t *ch, const sfx_t *sc, int cou
portable_samplepair_t *samp;
sndBuffer *chunk;
short *samples;
float ooff, fdata, fdiv, fleftvol, frightvol;
float ooff, fdata[2], fdiv, fleftvol, frightvol;
samp = &paintbuffer[ bufferOffset ];
@@ -410,6 +428,14 @@ static void S_PaintChannelFrom16_scalar( channel_t *ch, const sfx_t *sc, int cou
sampleOffset = sampleOffset*ch->oldDopplerScale;
}
if ( sc->soundChannels == 2 ) {
sampleOffset *= sc->soundChannels;
if ( sampleOffset & 1 ) {
sampleOffset &= ~1;
}
}
chunk = sc->soundData;
while (sampleOffset>=SND_CHUNK_SIZE) {
chunk = chunk->next;
@@ -426,6 +452,10 @@ static void S_PaintChannelFrom16_scalar( channel_t *ch, const sfx_t *sc, int cou
for ( i=0 ; i<count ; i++ ) {
data = samples[sampleOffset++];
samp[i].left += (data * leftvol)>>8;
if ( sc->soundChannels == 2 ) {
data = samples[sampleOffset++];
}
samp[i].right += (data * rightvol)>>8;
if (sampleOffset == SND_CHUNK_SIZE) {
@@ -447,10 +477,10 @@ static void S_PaintChannelFrom16_scalar( channel_t *ch, const sfx_t *sc, int cou
for ( i=0 ; i<count ; i++ ) {
aoff = ooff;
ooff = ooff + ch->dopplerScale;
ooff = ooff + ch->dopplerScale * sc->soundChannels;
boff = ooff;
fdata = 0;
for (j=aoff; j<boff; j++) {
fdata[0] = fdata[1] = 0;
for (j=aoff; j<boff; j += sc->soundChannels) {
if (j == SND_CHUNK_SIZE) {
chunk = chunk->next;
if (!chunk) {
@@ -459,11 +489,17 @@ static void S_PaintChannelFrom16_scalar( channel_t *ch, const sfx_t *sc, int cou
samples = chunk->sndChunk;
ooff -= SND_CHUNK_SIZE;
}
fdata += samples[j&(SND_CHUNK_SIZE-1)];
if ( sc->soundChannels == 2 ) {
fdata[0] += samples[j&(SND_CHUNK_SIZE-1)];
fdata[1] += samples[(j+1)&(SND_CHUNK_SIZE-1)];
} else {
fdata[0] += samples[j&(SND_CHUNK_SIZE-1)];
fdata[1] += samples[j&(SND_CHUNK_SIZE-1)];
}
}
fdiv = 256 * (boff-aoff);
samp[i].left += (fdata * fleftvol)/fdiv;
samp[i].right += (fdata * frightvol)/fdiv;
fdiv = 256 * (boff-aoff) / sc->soundChannels;
samp[i].left += (fdata[0] * fleftvol)/fdiv;
samp[i].right += (fdata[1] * frightvol)/fdiv;
}
}
}