Funko sources as released by Rotacak.HEADmaster

1 files changed, 752 insertions, 0 deletions

diff --git a/src/client/snd_mix.c b/src/client/snd_mix.c
new file mode 100644
index 0000000..4d9e716
--- /dev/null
+++ b/src/client/snd_mix.c
@@ -0,0 +1,752 @@
+/*
+===========================================================================
+Copyright (C) 1999-2005 Id Software, Inc.
+Copyright (C) 2000-2006 Tim Angus
+
+This file is part of Tremulous.
+
+Tremulous is free software; you can redistribute it
+and/or modify it under the terms of the GNU General Public License as
+published by the Free Software Foundation; either version 2 of the License,
+or (at your option) any later version.
+
+Tremulous is distributed in the hope that it will be
+useful, but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+GNU General Public License for more details.
+
+You should have received a copy of the GNU General Public License
+along with Tremulous; if not, write to the Free Software
+Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+===========================================================================
+*/
+// snd_mix.c -- portable code to mix sounds for snd_dma.c
+
+#include "client.h"
+#include "snd_local.h"
+#if idppc_altivec && !defined(MACOS_X)
+#include <altivec.h>
+#endif
+
+static portable_samplepair_t paintbuffer[PAINTBUFFER_SIZE];
+static int snd_vol;
+
+// bk001119 - these not static, required by unix/snd_mixa.s
+int*     snd_p;  
+int      snd_linear_count;
+short*   snd_out;
+
+#if	!id386                                        // if configured not to use asm
+
+void S_WriteLinearBlastStereo16 (void)
+{
+	int		i;
+	int		val;
+
+	for (i=0 ; i<snd_linear_count ; i+=2)
+	{
+		val = snd_p[i]>>8;
+		if (val > 0x7fff)
+			snd_out[i] = 0x7fff;
+		else if (val < -32768)
+			snd_out[i] = -32768;
+		else
+			snd_out[i] = val;
+
+		val = snd_p[i+1]>>8;
+		if (val > 0x7fff)
+			snd_out[i+1] = 0x7fff;
+		else if (val < -32768)
+			snd_out[i+1] = -32768;
+		else
+			snd_out[i+1] = val;
+	}
+}
+#elif defined(__GNUC__)
+// uses snd_mixa.s
+void S_WriteLinearBlastStereo16 (void);
+#else
+
+__declspec( naked ) void S_WriteLinearBlastStereo16 (void)
+{
+	__asm {
+
+ push edi
+ push ebx
+ mov ecx,ds:dword ptr[snd_linear_count]
+ mov ebx,ds:dword ptr[snd_p]
+ mov edi,ds:dword ptr[snd_out]
+LWLBLoopTop:
+ mov eax,ds:dword ptr[-8+ebx+ecx*4]
+ sar eax,8
+ cmp eax,07FFFh
+ jg LClampHigh
+ cmp eax,0FFFF8000h
+ jnl LClampDone
+ mov eax,0FFFF8000h
+ jmp LClampDone
+LClampHigh:
+ mov eax,07FFFh
+LClampDone:
+ mov edx,ds:dword ptr[-4+ebx+ecx*4]
+ sar edx,8
+ cmp edx,07FFFh
+ jg LClampHigh2
+ cmp edx,0FFFF8000h
+ jnl LClampDone2
+ mov edx,0FFFF8000h
+ jmp LClampDone2
+LClampHigh2:
+ mov edx,07FFFh
+LClampDone2:
+ shl edx,16
+ and eax,0FFFFh
+ or edx,eax
+ mov ds:dword ptr[-4+edi+ecx*2],edx
+ sub ecx,2
+ jnz LWLBLoopTop
+ pop ebx
+ pop edi
+ ret
+	}
+}
+
+#endif
+
+void S_TransferStereo16 (unsigned long *pbuf, int endtime)
+{
+	int		lpos;
+	int		ls_paintedtime;
+	
+	snd_p = (int *) paintbuffer;
+	ls_paintedtime = s_paintedtime;
+
+	while (ls_paintedtime < endtime)
+	{
+	// handle recirculating buffer issues
+		lpos = ls_paintedtime & ((dma.samples>>1)-1);
+
+		snd_out = (short *) pbuf + (lpos<<1);
+
+		snd_linear_count = (dma.samples>>1) - lpos;
+		if (ls_paintedtime + snd_linear_count > endtime)
+			snd_linear_count = endtime - ls_paintedtime;
+
+		snd_linear_count <<= 1;
+
+	// write a linear blast of samples
+		S_WriteLinearBlastStereo16 ();
+
+		snd_p += snd_linear_count;
+		ls_paintedtime += (snd_linear_count>>1);
+
+		if( CL_VideoRecording( ) )
+			CL_WriteAVIAudioFrame( (byte *)snd_out, snd_linear_count << 1 );
+	}
+}
+
+/*
+===================
+S_TransferPaintBuffer
+
+===================
+*/
+void S_TransferPaintBuffer(int endtime)
+{
+	int 	out_idx;
+	int 	count;
+	int 	out_mask;
+	int 	*p;
+	int 	step;
+	int		val;
+	unsigned long *pbuf;
+
+	pbuf = (unsigned long *)dma.buffer;
+
+
+	if ( s_testsound->integer ) {
+		int		i;
+		int		count;
+
+		// write a fixed sine wave
+		count = (endtime - s_paintedtime);
+		for (i=0 ; i<count ; i++)
+			paintbuffer[i].left = paintbuffer[i].right = sin((s_paintedtime+i)*0.1)*20000*256;
+	}
+
+
+	if (dma.samplebits == 16 && dma.channels == 2)
+	{	// optimized case
+		S_TransferStereo16 (pbuf, endtime);
+	}
+	else
+	{	// general case
+		p = (int *) paintbuffer;
+		count = (endtime - s_paintedtime) * dma.channels;
+		out_mask = dma.samples - 1; 
+		out_idx = s_paintedtime * dma.channels & out_mask;
+		step = 3 - dma.channels;
+
+		if (dma.samplebits == 16)
+		{
+			short *out = (short *) pbuf;
+			while (count--)
+			{
+				val = *p >> 8;
+				p+= step;
+				if (val > 0x7fff)
+					val = 0x7fff;
+				else if (val < -32768)
+					val = -32768;
+				out[out_idx] = val;
+				out_idx = (out_idx + 1) & out_mask;
+			}
+		}
+		else if (dma.samplebits == 8)
+		{
+			unsigned char *out = (unsigned char *) pbuf;
+			while (count--)
+			{
+				val = *p >> 8;
+				p+= step;
+				if (val > 0x7fff)
+					val = 0x7fff;
+				else if (val < -32768)
+					val = -32768;
+				out[out_idx] = (val>>8) + 128;
+				out_idx = (out_idx + 1) & out_mask;
+			}
+		}
+	}
+}
+
+
+/*
+===============================================================================
+
+CHANNEL MIXING
+
+===============================================================================
+*/
+
+#if idppc_altivec
+static void S_PaintChannelFrom16_altivec( channel_t *ch, const sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
+	int						data, aoff, boff;
+	int						leftvol, rightvol;
+	int						i, j;
+	portable_samplepair_t	*samp;
+	sndBuffer				*chunk;
+	short					*samples;
+	float					ooff, fdata, fdiv, fleftvol, frightvol;
+
+	samp = &paintbuffer[ bufferOffset ];
+
+	if (ch->doppler) {
+		sampleOffset = sampleOffset*ch->oldDopplerScale;
+	}
+
+	chunk = sc->soundData;
+	while (sampleOffset>=SND_CHUNK_SIZE) {
+		chunk = chunk->next;
+		sampleOffset -= SND_CHUNK_SIZE;
+		if (!chunk) {
+			chunk = sc->soundData;
+		}
+	}
+
+	if (!ch->doppler || ch->dopplerScale==1.0f) {
+		vector signed short volume_vec;
+		vector unsigned int volume_shift;
+		int vectorCount, samplesLeft, chunkSamplesLeft;
+		leftvol = ch->leftvol*snd_vol;
+		rightvol = ch->rightvol*snd_vol;
+		samples = chunk->sndChunk;
+		((short *)&volume_vec)[0] = leftvol;
+		((short *)&volume_vec)[1] = leftvol;
+		((short *)&volume_vec)[4] = leftvol;
+		((short *)&volume_vec)[5] = leftvol;
+		((short *)&volume_vec)[2] = rightvol;
+		((short *)&volume_vec)[3] = rightvol;
+		((short *)&volume_vec)[6] = rightvol;
+		((short *)&volume_vec)[7] = rightvol;
+		volume_shift = vec_splat_u32(8);
+		i = 0;
+
+		while(i < count) {
+			/* Try to align destination to 16-byte boundary */
+			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;
+				samp[i].right += (data * rightvol)>>8;
+	
+				if (sampleOffset == SND_CHUNK_SIZE) {
+					chunk = chunk->next;
+					samples = chunk->sndChunk;
+					sampleOffset = 0;
+				}
+				i++;
+			}
+			/* Destination is now aligned.  Process as many 8-sample 
+			   chunks as we can before we run out of room from the current
+			   sound chunk.  We do 8 per loop to avoid extra source data reads. */
+			samplesLeft = count - i;
+			chunkSamplesLeft = SND_CHUNK_SIZE - sampleOffset;
+			if(samplesLeft > chunkSamplesLeft)
+				samplesLeft = chunkSamplesLeft;
+			
+			vectorCount = samplesLeft / 8;
+			
+			if(vectorCount)
+			{
+				vector unsigned char tmp;
+				vector short s0, s1, sampleData0, sampleData1;
+				vector signed int merge0, merge1;
+				vector signed int d0, d1, d2, d3;				
+				vector unsigned char samplePermute0 =
+					VECCONST_UINT8(0, 1, 4, 5, 0, 1, 4, 5, 2, 3, 6, 7, 2, 3, 6, 7);
+				vector unsigned char samplePermute1 = 
+					VECCONST_UINT8(8, 9, 12, 13, 8, 9, 12, 13, 10, 11, 14, 15, 10, 11, 14, 15);
+				vector unsigned char loadPermute0, loadPermute1;
+				
+				// Rather than permute the vectors after we load them to do the sample
+				// replication and rearrangement, we permute the alignment vector so
+				// we do everything in one step below and avoid data shuffling.
+				tmp = vec_lvsl(0,&samples[sampleOffset]);								
+				loadPermute0 = vec_perm(tmp,tmp,samplePermute0);
+				loadPermute1 = vec_perm(tmp,tmp,samplePermute1);
+				
+				s0 = *(vector short *)&samples[sampleOffset];
+				while(vectorCount)
+				{
+					/* Load up source (16-bit) sample data */
+					s1 = *(vector short *)&samples[sampleOffset+7];
+					
+					/* Load up destination sample data */
+					d0 = *(vector signed int *)&samp[i];
+					d1 = *(vector signed int *)&samp[i+2];
+					d2 = *(vector signed int *)&samp[i+4];
+					d3 = *(vector signed int *)&samp[i+6];
+
+					sampleData0 = vec_perm(s0,s1,loadPermute0);
+					sampleData1 = vec_perm(s0,s1,loadPermute1);
+					
+					merge0 = vec_mule(sampleData0,volume_vec);
+					merge0 = vec_sra(merge0,volume_shift);	/* Shift down to proper range */
+					
+					merge1 = vec_mulo(sampleData0,volume_vec);
+					merge1 = vec_sra(merge1,volume_shift);
+					
+					d0 = vec_add(merge0,d0);
+					d1 = vec_add(merge1,d1);
+					
+					merge0 = vec_mule(sampleData1,volume_vec);
+					merge0 = vec_sra(merge0,volume_shift);	/* Shift down to proper range */
+					
+					merge1 = vec_mulo(sampleData1,volume_vec);
+					merge1 = vec_sra(merge1,volume_shift);					
+
+					d2 = vec_add(merge0,d2);
+					d3 = vec_add(merge1,d3);
+
+					/* Store destination sample data */
+					*(vector signed int *)&samp[i] = d0;
+					*(vector signed int *)&samp[i+2] = d1;
+					*(vector signed int *)&samp[i+4] = d2;
+					*(vector signed int *)&samp[i+6] = d3;
+
+					i += 8;
+					vectorCount--;
+					s0 = s1;
+					sampleOffset += 8;
+				}
+				if (sampleOffset == SND_CHUNK_SIZE) {
+					chunk = chunk->next;
+					samples = chunk->sndChunk;
+					sampleOffset = 0;
+				}
+			}
+		}
+	} else {
+		fleftvol = ch->leftvol*snd_vol;
+		frightvol = ch->rightvol*snd_vol;
+
+		ooff = sampleOffset;
+		samples = chunk->sndChunk;
+		
+		for ( i=0 ; i<count ; i++ ) {
+
+			aoff = ooff;
+			ooff = ooff + ch->dopplerScale;
+			boff = ooff;
+			fdata = 0;
+			for (j=aoff; j<boff; j++) {
+				if (j == SND_CHUNK_SIZE) {
+					chunk = chunk->next;
+					if (!chunk) {
+						chunk = sc->soundData;
+					}
+					samples = chunk->sndChunk;
+					ooff -= SND_CHUNK_SIZE;
+				}
+				fdata  += samples[j&(SND_CHUNK_SIZE-1)];
+			}
+			fdiv = 256 * (boff-aoff);
+			samp[i].left += (fdata * fleftvol)/fdiv;
+			samp[i].right += (fdata * frightvol)/fdiv;
+		}
+	}
+}
+#endif
+
+static void S_PaintChannelFrom16_scalar( channel_t *ch, const sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
+	int						data, aoff, boff;
+	int						leftvol, rightvol;
+	int						i, j;
+	portable_samplepair_t	*samp;
+	sndBuffer				*chunk;
+	short					*samples;
+	float					ooff, fdata, fdiv, fleftvol, frightvol;
+
+	samp = &paintbuffer[ bufferOffset ];
+
+	if (ch->doppler) {
+		sampleOffset = sampleOffset*ch->oldDopplerScale;
+	}
+
+	chunk = sc->soundData;
+	while (sampleOffset>=SND_CHUNK_SIZE) {
+		chunk = chunk->next;
+		sampleOffset -= SND_CHUNK_SIZE;
+		if (!chunk) {
+			chunk = sc->soundData;
+		}
+	}
+
+	if (!ch->doppler || ch->dopplerScale==1.0f) {
+		leftvol = ch->leftvol*snd_vol;
+		rightvol = ch->rightvol*snd_vol;
+		samples = chunk->sndChunk;
+		for ( i=0 ; i<count ; i++ ) {
+			data  = samples[sampleOffset++];
+			samp[i].left += (data * leftvol)>>8;
+			samp[i].right += (data * rightvol)>>8;
+
+			if (sampleOffset == SND_CHUNK_SIZE) {
+				chunk = chunk->next;
+				samples = chunk->sndChunk;
+				sampleOffset = 0;
+			}
+		}
+	} else {
+		fleftvol = ch->leftvol*snd_vol;
+		frightvol = ch->rightvol*snd_vol;
+
+		ooff = sampleOffset;
+		samples = chunk->sndChunk;
+		
+
+
+
+		for ( i=0 ; i<count ; i++ ) {
+
+			aoff = ooff;
+			ooff = ooff + ch->dopplerScale;
+			boff = ooff;
+			fdata = 0;
+			for (j=aoff; j<boff; j++) {
+				if (j == SND_CHUNK_SIZE) {
+					chunk = chunk->next;
+					if (!chunk) {
+						chunk = sc->soundData;
+					}
+					samples = chunk->sndChunk;
+					ooff -= SND_CHUNK_SIZE;
+				}
+				fdata  += samples[j&(SND_CHUNK_SIZE-1)];
+			}
+			fdiv = 256 * (boff-aoff);
+			samp[i].left += (fdata * fleftvol)/fdiv;
+			samp[i].right += (fdata * frightvol)/fdiv;
+		}
+	}
+}
+
+static void S_PaintChannelFrom16( channel_t *ch, const sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
+#if idppc_altivec
+	if (com_altivec->integer) {
+		// must be in a seperate function or G3 systems will crash.
+		S_PaintChannelFrom16_altivec( ch, sc, count, sampleOffset, bufferOffset );
+		return;
+	}
+#endif
+	S_PaintChannelFrom16_scalar( ch, sc, count, sampleOffset, bufferOffset );
+}
+
+void S_PaintChannelFromWavelet( channel_t *ch, sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
+	int						data;
+	int						leftvol, rightvol;
+	int						i;
+	portable_samplepair_t	*samp;
+	sndBuffer				*chunk;
+	short					*samples;
+
+	leftvol = ch->leftvol*snd_vol;
+	rightvol = ch->rightvol*snd_vol;
+
+	i = 0;
+	samp = &paintbuffer[ bufferOffset ];
+	chunk = sc->soundData;
+	while (sampleOffset>=(SND_CHUNK_SIZE_FLOAT*4)) {
+		chunk = chunk->next;
+		sampleOffset -= (SND_CHUNK_SIZE_FLOAT*4);
+		i++;
+	}
+
+	if (i!=sfxScratchIndex || sfxScratchPointer != sc) {
+		S_AdpcmGetSamples( chunk, sfxScratchBuffer );
+		sfxScratchIndex = i;
+		sfxScratchPointer = sc;
+	}
+
+	samples = sfxScratchBuffer;
+
+	for ( i=0 ; i<count ; i++ ) {
+		data  = samples[sampleOffset++];
+		samp[i].left += (data * leftvol)>>8;
+		samp[i].right += (data * rightvol)>>8;
+
+		if (sampleOffset == SND_CHUNK_SIZE*2) {
+			chunk = chunk->next;
+			decodeWavelet(chunk, sfxScratchBuffer);
+			sfxScratchIndex++;
+			sampleOffset = 0;
+		}
+	}
+}
+
+void S_PaintChannelFromADPCM( channel_t *ch, sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
+	int						data;
+	int						leftvol, rightvol;
+	int						i;
+	portable_samplepair_t	*samp;
+	sndBuffer				*chunk;
+	short					*samples;
+
+	leftvol = ch->leftvol*snd_vol;
+	rightvol = ch->rightvol*snd_vol;
+
+	i = 0;
+	samp = &paintbuffer[ bufferOffset ];
+	chunk = sc->soundData;
+
+	if (ch->doppler) {
+		sampleOffset = sampleOffset*ch->oldDopplerScale;
+	}
+
+	while (sampleOffset>=(SND_CHUNK_SIZE*4)) {
+		chunk = chunk->next;
+		sampleOffset -= (SND_CHUNK_SIZE*4);
+		i++;
+	}
+
+	if (i!=sfxScratchIndex || sfxScratchPointer != sc) {
+		S_AdpcmGetSamples( chunk, sfxScratchBuffer );
+		sfxScratchIndex = i;
+		sfxScratchPointer = sc;
+	}
+
+	samples = sfxScratchBuffer;
+
+	for ( i=0 ; i<count ; i++ ) {
+		data  = samples[sampleOffset++];
+		samp[i].left += (data * leftvol)>>8;
+		samp[i].right += (data * rightvol)>>8;
+
+		if (sampleOffset == SND_CHUNK_SIZE*4) {
+			chunk = chunk->next;
+			S_AdpcmGetSamples( chunk, sfxScratchBuffer);
+			sampleOffset = 0;
+			sfxScratchIndex++;
+		}
+	}
+}
+
+void S_PaintChannelFromMuLaw( channel_t *ch, sfx_t *sc, int count, int sampleOffset, int bufferOffset ) {
+	int						data;
+	int						leftvol, rightvol;
+	int						i;
+	portable_samplepair_t	*samp;
+	sndBuffer				*chunk;
+	byte					*samples;
+	float					ooff;
+
+	leftvol = ch->leftvol*snd_vol;
+	rightvol = ch->rightvol*snd_vol;
+
+	samp = &paintbuffer[ bufferOffset ];
+	chunk = sc->soundData;
+	while (sampleOffset>=(SND_CHUNK_SIZE*2)) {
+		chunk = chunk->next;
+		sampleOffset -= (SND_CHUNK_SIZE*2);
+		if (!chunk) {
+			chunk = sc->soundData;
+		}
+	}
+
+	if (!ch->doppler) {
+		samples = (byte *)chunk->sndChunk + sampleOffset;
+		for ( i=0 ; i<count ; i++ ) {
+			data  = mulawToShort[*samples];
+			samp[i].left += (data * leftvol)>>8;
+			samp[i].right += (data * rightvol)>>8;
+			samples++;
+			if (samples == (byte *)chunk->sndChunk+(SND_CHUNK_SIZE*2)) {
+				chunk = chunk->next;
+				samples = (byte *)chunk->sndChunk;
+			}
+		}
+	} else {
+		ooff = sampleOffset;
+		samples = (byte *)chunk->sndChunk;
+		for ( i=0 ; i<count ; i++ ) {
+			data  = mulawToShort[samples[(int)(ooff)]];
+			ooff = ooff + ch->dopplerScale;
+			samp[i].left += (data * leftvol)>>8;
+			samp[i].right += (data * rightvol)>>8;
+			if (ooff >= SND_CHUNK_SIZE*2) {
+				chunk = chunk->next;
+				if (!chunk) {
+					chunk = sc->soundData;
+				}
+				samples = (byte *)chunk->sndChunk;
+				ooff = 0.0;
+			}
+		}
+	}
+}
+
+/*
+===================
+S_PaintChannels
+===================
+*/
+void S_PaintChannels( int endtime ) {
+	int 	i;
+	int 	end;
+	channel_t *ch;
+	sfx_t	*sc;
+	int		ltime, count;
+	int		sampleOffset;
+
+
+	snd_vol = s_volume->value*255;
+
+//Com_Printf ("%i to %i\n", s_paintedtime, endtime);
+	while ( s_paintedtime < endtime ) {
+		// if paintbuffer is smaller than DMA buffer
+		// we may need to fill it multiple times
+		end = endtime;
+		if ( endtime - s_paintedtime > PAINTBUFFER_SIZE ) {
+			end = s_paintedtime + PAINTBUFFER_SIZE;
+		}
+
+		// clear the paint buffer to either music or zeros
+		if ( s_rawend < s_paintedtime ) {
+			if ( s_rawend ) {
+				//Com_DPrintf ("background sound underrun\n");
+			}
+			Com_Memset(paintbuffer, 0, (end - s_paintedtime) * sizeof(portable_samplepair_t));
+		} else {
+			// copy from the streaming sound source
+			int		s;
+			int		stop;
+
+			stop = (end < s_rawend) ? end : s_rawend;
+
+			for ( i = s_paintedtime ; i < stop ; i++ ) {
+				s = i&(MAX_RAW_SAMPLES-1);
+				paintbuffer[i-s_paintedtime] = s_rawsamples[s];
+			}
+//		if (i != end)
+//			Com_Printf ("partial stream\n");
+//		else
+//			Com_Printf ("full stream\n");
+			for ( ; i < end ; i++ ) {
+				paintbuffer[i-s_paintedtime].left =
+				paintbuffer[i-s_paintedtime].right = 0;
+			}
+		}
+
+		// paint in the channels.
+		ch = s_channels;
+		for ( i = 0; i < MAX_CHANNELS ; i++, ch++ ) {		
+			if ( !ch->thesfx || (ch->leftvol<0.25 && ch->rightvol<0.25 )) {
+				continue;
+			}
+
+			ltime = s_paintedtime;
+			sc = ch->thesfx;
+
+			sampleOffset = ltime - ch->startSample;
+			count = end - ltime;
+			if ( sampleOffset + count > sc->soundLength ) {
+				count = sc->soundLength - sampleOffset;
+			}
+
+			if ( count > 0 ) {	
+				if( sc->soundCompressionMethod == 1) {
+					S_PaintChannelFromADPCM		(ch, sc, count, sampleOffset, ltime - s_paintedtime);
+				} else if( sc->soundCompressionMethod == 2) {
+					S_PaintChannelFromWavelet	(ch, sc, count, sampleOffset, ltime - s_paintedtime);
+				} else if( sc->soundCompressionMethod == 3) {
+					S_PaintChannelFromMuLaw	(ch, sc, count, sampleOffset, ltime - s_paintedtime);
+				} else {
+					S_PaintChannelFrom16		(ch, sc, count, sampleOffset, ltime - s_paintedtime);
+				}
+			}
+		}
+
+		// paint in the looped channels.
+		ch = loop_channels;
+		for ( i = 0; i < numLoopChannels ; i++, ch++ ) {		
+			if ( !ch->thesfx || (!ch->leftvol && !ch->rightvol )) {
+				continue;
+			}
+
+			ltime = s_paintedtime;
+			sc = ch->thesfx;
+
+			if (sc->soundData==NULL || sc->soundLength==0) {
+				continue;
+			}
+			// we might have to make two passes if it
+			// is a looping sound effect and the end of
+			// the sample is hit
+			do {
+				sampleOffset = (ltime % sc->soundLength);
+
+				count = end - ltime;
+				if ( sampleOffset + count > sc->soundLength ) {
+					count = sc->soundLength - sampleOffset;
+				}
+
+				if ( count > 0 ) {	
+					if( sc->soundCompressionMethod == 1) {
+						S_PaintChannelFromADPCM		(ch, sc, count, sampleOffset, ltime - s_paintedtime);
+					} else if( sc->soundCompressionMethod == 2) {
+						S_PaintChannelFromWavelet	(ch, sc, count, sampleOffset, ltime - s_paintedtime);
+					} else if( sc->soundCompressionMethod == 3) {
+						S_PaintChannelFromMuLaw		(ch, sc, count, sampleOffset, ltime - s_paintedtime);
+					} else {
+						S_PaintChannelFrom16		(ch, sc, count, sampleOffset, ltime - s_paintedtime);
+					}
+					ltime += count;
+				}
+			} while ( ltime < end);
+		}
+
+		// transfer out according to DMA format
+		S_TransferPaintBuffer( end );
+		s_paintedtime = end;
+	}
+}