Fix playback of stereo sounds in Base sound system

Already works correctly in OpenAL.

3 files changed, 98 insertions, 53 deletions

diff --git a/src/client/snd_local.h b/src/client/snd_local.h
index 1f842d89..8f5be112 100644
--- a/src/client/snd_local.h
+++ b/src/client/snd_local.h
@@ -57,6 +57,7 @@ typedef struct sfx_s {
 	qboolean		soundCompressed;		// not in Memory
 	int				soundCompressionMethod;	
 	int 			soundLength;
+	int			soundChannels;
 	char 			soundName[MAX_QPATH];
 	int				lastTimeUsed;
 	int				duration;
diff --git a/src/client/snd_mem.c b/src/client/snd_mem.c
index ecca5292..299d9d34 100644
--- a/src/client/snd_mem.c
+++ b/src/client/snd_mem.c
@@ -114,47 +114,51 @@ 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;
 }
 
 /*
@@ -164,11 +168,11 @@ 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
@@ -176,18 +180,21 @@ 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;
 }
@@ -226,7 +233,7 @@ 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;
 
@@ -236,29 +243,30 @@ 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);
diff --git a/src/client/snd_mix.c b/src/client/snd_mix.c
index 80c892f5..3c2fd240 100644
--- a/src/client/snd_mix.c
+++ b/src/client/snd_mix.c
@@ -235,7 +235,7 @@ 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 ];
 
@@ -243,6 +243,14 @@ 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;
@@ -275,6 +283,10 @@ 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) {
@@ -374,10 +386,10 @@ 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) {
@@ -386,11 +398,17 @@ 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;
 		}
 	}
 }
@@ -403,7 +421,7 @@ 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 ];
 
@@ -411,6 +429,14 @@ 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;
@@ -427,6 +453,10 @@ 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) {
@@ -448,10 +478,10 @@ 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) {
@@ -460,11 +490,17 @@ 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;
 		}
 	}
 }