mirror of
https://github.com/DrBeef/ioq3quest.git
synced 2024-11-23 20:42:30 +00:00
776 lines
20 KiB
C
776 lines
20 KiB
C
/*
|
|
===========================================================================
|
|
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;
|
|
|
|
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;
|
|
|
|
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;
|
|
|
|
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;
|
|
}
|
|
}
|