Fix SDL audio playback with 16-bit stereo sound

My commit last month "Fix SDL audio playback with surround sound" broke
16-bit stereo sound. S_TransferStereo16() still assumed that dma.samples
was a power of two. I also cleaned up code related to the previously
mentioned commit.

code/client/snd_dma.c

@@ -1242,9 +1242,6 @@ void S_GetSoundtime(void)
 	int		samplepos;
 	static	int		buffers;
 	static	int		oldsamplepos;
-	int		fullsamples;
-	
-	fullsamples = dma.samples / dma.channels;
 
 	if( CL_VideoRecording( ) )
 	{
@@ -1268,13 +1265,13 @@ void S_GetSoundtime(void)
 		if (s_paintedtime > 0x40000000)
 		{	// time to chop things off to avoid 32 bit limits
 			buffers = 0;
-			s_paintedtime = fullsamples;
+			s_paintedtime = dma.fullsamples;
 			S_Base_StopAllSounds ();
 		}
 	}
 	oldsamplepos = samplepos;
 
-	s_soundtime = buffers*fullsamples + samplepos/dma.channels;
+	s_soundtime = buffers*dma.fullsamples + samplepos/dma.channels;
 
 #if 0
 // check to make sure that we haven't overshot
@@ -1295,7 +1292,6 @@ void S_GetSoundtime(void)
 
 void S_Update_(void) {
 	unsigned        endtime;
-	int				samps;
 	static			float	lastTime = 0.0f;
 	float			ma, op;
 	float			thisTime, sane;
@@ -1339,9 +1335,8 @@ void S_Update_(void) {
 		& ~(dma.submission_chunk-1);
 
 	// never mix more than the complete buffer
-	samps = dma.samples / dma.channels;
-	if (endtime - s_soundtime > samps)
-		endtime = s_soundtime + samps;
+	if (endtime - s_soundtime > dma.fullsamples)
+		endtime = s_soundtime + dma.fullsamples;
 
 
 

code/client/snd_local.h

@@ -65,6 +65,7 @@ typedef struct sfx_s {
 typedef struct {
 	int			channels;
 	int			samples;				// mono samples in buffer
+	int			fullsamples;			// samples with all channels in buffer (samples divided by channels)
 	int			submission_chunk;		// don't mix less than this #
 	int			samplebits;
 	int			isfloat;

code/client/snd_mix.c

@@ -119,24 +119,24 @@ void S_TransferStereo16 (unsigned long *pbuf, int endtime)
 	while (ls_paintedtime < endtime)
 	{
 	// handle recirculating buffer issues
-		lpos = ls_paintedtime & ((dma.samples>>1)-1);
+		lpos = ls_paintedtime % dma.fullsamples;
 
-		snd_out = (short *) pbuf + (lpos<<1);
+		snd_out = (short *) pbuf + (lpos<<1); // lpos * dma.channels
 
-		snd_linear_count = (dma.samples>>1) - lpos;
+		snd_linear_count = dma.fullsamples - lpos;
 		if (ls_paintedtime + snd_linear_count > endtime)
 			snd_linear_count = endtime - ls_paintedtime;
 
-		snd_linear_count <<= 1;
+		snd_linear_count <<= 1; // snd_linear_count *= dma.channels
 
 	// write a linear blast of samples
 		S_WriteLinearBlastStereo16 ();
 
 		snd_p += snd_linear_count;
-		ls_paintedtime += (snd_linear_count>>1);
+		ls_paintedtime += (snd_linear_count>>1); // snd_linear_count / dma.channels
 
 		if( CL_VideoRecording( ) )
-			CL_WriteAVIAudioFrame( (byte *)snd_out, snd_linear_count << 1 );
+			CL_WriteAVIAudioFrame( (byte *)snd_out, snd_linear_count << 1 ); // snd_linear_count * (dma.samplebits/8)
 	}
 }
 
@@ -175,15 +175,15 @@ void S_TransferPaintBuffer(int endtime)
 	{	// general case
 		p = (int *) paintbuffer;
 		count = (endtime - s_paintedtime) * dma.channels;
-		out_idx = s_paintedtime * dma.channels % dma.samples;
-		step = 3 - MIN( dma.channels, 2 );
+		out_idx = (s_paintedtime * dma.channels) % dma.samples;
+		step = 3 - MIN(dma.channels, 2);
 
 		if ((dma.isfloat) && (dma.samplebits == 32))
 		{
 			float *out = (float *) pbuf;
 			for (i=0 ; i<count ; i++)
 			{
-				if ( i % dma.channels >= 2 )
+				if ((i % dma.channels) >= 2)
 				{
 					val = 0;
 				}
@@ -205,7 +205,7 @@ void S_TransferPaintBuffer(int endtime)
 			short *out = (short *) pbuf;
 			for (i=0 ; i<count ; i++)
 			{
-				if ( i % dma.channels >= 2 )
+				if ((i % dma.channels) >= 2)
 				{
 					val = 0;
 				}
@@ -227,7 +227,7 @@ void S_TransferPaintBuffer(int endtime)
 			unsigned char *out = (unsigned char *) pbuf;
 			for (i=0 ; i<count ; i++)
 			{
-				if ( i % dma.channels >= 2 )
+				if ((i % dma.channels) >= 2)
 				{
 					val = 0;
 				}

code/sdl/sdl_snd.c

@@ -272,6 +272,7 @@ qboolean SNDDMA_Init(void)
 	dma.isfloat = SDL_AUDIO_ISFLOAT(obtained.format);
 	dma.channels = obtained.channels;
 	dma.samples = tmp;
+	dma.fullsamples = dma.samples / dma.channels;
 	dma.submission_chunk = 1;
 	dma.speed = obtained.freq;
 	dmasize = (dma.samples * (dma.samplebits/8));