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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
===========================================================================
*/
/* This file is only compiled for PowerPC builds with Altivec support.
Altivec intrinsics need to be in a separate file, so GCC's -maltivec
command line can enable them, but give us the option to _not_ use that
on other files, where the compiler might then generate Altivec
instructions for normal floating point, crashing on G3 (etc) processors. */
#include "client.h"
#include "snd_local.h"
#if idppc_altivec
#if !defined(__APPLE__)
#include <altivec.h>
#endif
void S_PaintChannelFrom16_altivec( portable_samplepair_t paintbuffer[PAINTBUFFER_SIZE], int snd_vol, 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
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