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lilium-voyager/code/client/snd_mix.c
Simon McVittie 5909b9a1cf Isolate the Altivec code so non-Altivec PPC targets can use the same binary. 
Moved all the code using Altivec intrinsics to separate files. This 
means we can optionally use GCC's -maltivec on just these files, which
are chosen at runtime if the CPU supports Altivec, and compile the rest
without it, making a single binary that has Altivec optimizations but
can still work on G3.

Unlike SSE and similar extensions on x86, there does not seem to be
a way to enable conditional, targeted use of Altivec based on runtime
detection (which is what ioquake3 wants to do) without also giving the
compiler permission to use Altivec in code generation; so to not crash
on CPUs that do not implement Altivec, we'll have to turn it off
altogether, except in translation units that are only entered when
runtime Altivec detection is successful.

This has been tested on Linux PPC (on an Altivec-enabled CPU),
but we may need further work after testing trickles out to other
PowerPC devices and ancient Mac OS X builds.

I did a little work on this patch, but the majority of the effort belongs 
to Simon McVittie (thanks!).
2018-05-12 14:14:47 -04:00

609 lines
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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"
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
===============================================================================
*/
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 translation unit or G3 systems will crash.
S_PaintChannelFrom16_altivec( paintbuffer, snd_vol, 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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