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ioq3quest/code/client/snd_mix.c
Ryan C. Gordon 4f8c7c2f2f Support SDL audio devices that require float32 samples. 
Fixes missing audio when playing on Windows with SDL 2.0.7, which started
using WASAPI, which demands floating point audio.
2018-04-13 14:05:12 -04:00

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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(__APPLE__)
#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.isfloat) && (dma.samplebits == 32))
{
float *out = (float *) pbuf;
while (count--)
{
val = *p >> 8;
p+= step;
if (val > 0x7fff)
val = 0x7fff;
else if (val < -32767) /* clamp to one less than max to make division max out at -1.0f. */
val = -32767;
out[out_idx] = ((float) val) / 32767.0f;
out_idx = (out_idx + 1) & out_mask;
}
}
else 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 separate 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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