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lilium-voyager/code/client/snd_mix.c

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/*
===========================================================================
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 Quake III Arena source code; 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;
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;
// 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[2], fdiv, fleftvol, frightvol;
if (sc->soundChannels <= 0) {
return;
}
samp = &paintbuffer[ bufferOffset ];
if (ch->doppler) {
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;
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;
if ( sc->soundChannels == 2 ) {
data = samples[sampleOffset++];
}
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 * sc->soundChannels;
boff = ooff;
fdata[0] = fdata[1] = 0;
for (j=aoff; j<boff; j += sc->soundChannels) {
if (j == SND_CHUNK_SIZE) {
chunk = chunk->next;
if (!chunk) {
chunk = sc->soundData;
}
samples = chunk->sndChunk;
ooff -= SND_CHUNK_SIZE;
}
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) / sc->soundChannels;
samp[i].left += (fdata[0] * fleftvol)/fdiv;
samp[i].right += (fdata[1] * 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[2], fdiv, fleftvol, frightvol;
if (sc->soundChannels <= 0) {
return;
}
samp = &paintbuffer[ bufferOffset ];
if (ch->doppler) {
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;
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;
if ( sc->soundChannels == 2 ) {
data = samples[sampleOffset++];
}
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 * sc->soundChannels;
boff = ooff;
fdata[0] = fdata[1] = 0;
for (j=aoff; j<boff; j += sc->soundChannels) {
if (j == SND_CHUNK_SIZE) {
chunk = chunk->next;
if (!chunk) {
chunk = sc->soundData;
}
samples = chunk->sndChunk;
ooff -= SND_CHUNK_SIZE;
}
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) / sc->soundChannels;
samp[i].left += (fdata[0] * fleftvol)/fdiv;
samp[i].right += (fdata[1] * 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 (chunk != NULL && 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;
int stream;
channel_t *ch;
sfx_t *sc;
int ltime, count;
int sampleOffset;
if(s_muted->integer)
snd_vol = 0;
else
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 and mix any raw samples...
Com_Memset(paintbuffer, 0, sizeof (paintbuffer));
for (stream = 0; stream < MAX_RAW_STREAMS; stream++) {
if ( s_rawend[stream] >= s_paintedtime ) {
// copy from the streaming sound source
const portable_samplepair_t *rawsamples = s_rawsamples[stream];
const int stop = (end < s_rawend[stream]) ? end : s_rawend[stream];
for ( i = s_paintedtime ; i < stop ; i++ ) {
const int s = i&(MAX_RAW_SAMPLES-1);
paintbuffer[i-s_paintedtime].left += rawsamples[s].left;
paintbuffer[i-s_paintedtime].right += rawsamples[s].right;
}
}
}
// 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;
if (sc->soundData==NULL || sc->soundLength==0) {
continue;
}
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;
}
}
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