diff options
Diffstat (limited to 'code/client/snd_mix.c')
| -rwxr-xr-x | code/client/snd_mix.c | 1362 |
1 files changed, 681 insertions, 681 deletions
diff --git a/code/client/snd_mix.c b/code/client/snd_mix.c index ec983b9..02be59c 100755 --- a/code/client/snd_mix.c +++ b/code/client/snd_mix.c @@ -1,681 +1,681 @@ -/*
-===========================================================================
-Copyright (C) 1999-2005 Id Software, Inc.
-
-This file is part of Quake III Arena source code.
-
-Quake III Arena source code 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.
-
-Quake III Arena source code 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 Foobar; 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 "snd_local.h"
-
-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 !( (defined __linux__ || defined __FreeBSD__ ) && (defined __i386__) ) // rb010123
-#if !id386
-
-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;
- }
-}
-#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
-#else
-// forward declare, implementation somewhere else
-void S_WriteLinearBlastStereo16 (void);
-#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);
- }
-}
-
-/*
-===================
-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
-
-===============================================================================
-*/
-
-static void S_PaintChannelFrom16( 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) {
-#if idppc_altivec
- vector signed short volume_vec;
- vector unsigned int volume_shift;
- int vectorCount, samplesLeft, chunkSamplesLeft;
-#endif
- leftvol = ch->leftvol*snd_vol;
- rightvol = ch->rightvol*snd_vol;
- samples = chunk->sndChunk;
-#if idppc_altivec
- ((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 short samples0, samples1;
- vector signed int left0, right0;
- vector signed int merge0, merge1;
- vector signed int d0, d1, d2, d3;
- vector unsigned char samplePermute0 =
- (vector unsigned char)(0, 1, 4, 5, 0, 1, 4, 5, 2, 3, 6, 7, 2, 3, 6, 7);
- vector unsigned char samplePermute1 =
- (vector unsigned char)(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
- 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;
- }
- }
-#endif
- } 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;
- }
- }
-}
-
-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;
- }
-}
+/* +=========================================================================== +Copyright (C) 1999-2005 Id Software, Inc. + +This file is part of Quake III Arena source code. + +Quake III Arena source code 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. + +Quake III Arena source code 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 Foobar; 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 "snd_local.h" + +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 !( (defined __linux__ || defined __FreeBSD__ ) && (defined __i386__) ) // rb010123 +#if !id386 + +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; + } +} +#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 +#else +// forward declare, implementation somewhere else +void S_WriteLinearBlastStereo16 (void); +#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); + } +} + +/* +=================== +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 + +=============================================================================== +*/ + +static void S_PaintChannelFrom16( 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) { +#if idppc_altivec + vector signed short volume_vec; + vector unsigned int volume_shift; + int vectorCount, samplesLeft, chunkSamplesLeft; +#endif + leftvol = ch->leftvol*snd_vol; + rightvol = ch->rightvol*snd_vol; + samples = chunk->sndChunk; +#if idppc_altivec + ((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 short samples0, samples1; + vector signed int left0, right0; + vector signed int merge0, merge1; + vector signed int d0, d1, d2, d3; + vector unsigned char samplePermute0 = + (vector unsigned char)(0, 1, 4, 5, 0, 1, 4, 5, 2, 3, 6, 7, 2, 3, 6, 7); + vector unsigned char samplePermute1 = + (vector unsigned char)(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 + 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; + } + } +#endif + } 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; + } + } +} + +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; + } +} |