jedioutcast/CODE-mp/client/snd_mix.cpp

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2013-04-04 14:52:42 +00:00
// snd_mix.c -- portable code to mix sounds for snd_dma.c
#include "snd_local.h"
#include "snd_mp3.h"
static portable_samplepair_t paintbuffer[PAINTBUFFER_SIZE];
static int snd_vol;
// bk001119 - these not static, required by unix/snd_mixa.s
int* snd_p;
int snd_linear_count;
short* snd_out;
#if !( (defined __linux__ || __FreeBSD__ ) && (defined __i386__) ) // rb010123
#if !id386
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 < (short)0x8000)
snd_out[i] = (short)0x8000;
else
snd_out[i] = val;
val = snd_p[i+1]>>8;
if (val > 0x7fff)
snd_out[i+1] = 0x7fff;
else if (val < (short)0x8000)
snd_out[i+1] = (short)0x8000;
else
snd_out[i+1] = val;
}
}
#else
unsigned int uiMMXAvailable = 0; // leave as 32 bit
__declspec( naked ) void S_WriteLinearBlastStereo16 (void)
{
__asm {
push edi
push ebx
mov ecx,ds:dword ptr[snd_linear_count] // snd_linear_count is always even at this point, but not nec. mult of 4
mov ebx,ds:dword ptr[snd_p]
mov edi,ds:dword ptr[snd_out]
cmp [uiMMXAvailable], dword ptr 0
je NoMMX
// writes 8 items (128 bits) per loop pass...
//
cmp ecx,8
jb NoMMX
LWLBLoopTop_MMX:
movq mm1,[-8+ebx+ecx*4]
movq mm0,[-16+ebx+ecx*4]
movq mm3,[-24+ebx+ecx*4]
movq mm2,[-32+ebx+ecx*4]
psrad mm0,8
psrad mm1,8
psrad mm2,8
psrad mm3,8
packssdw mm0,mm1
packssdw mm2,mm3
movq [-8+edi+ecx*2],mm0
movq [-16+edi+ecx*2],mm2
sub ecx,8
cmp ecx,8
jae LWLBLoopTop_MMX
emms
// now deal with any remaining count...
//
jecxz LExit
NoMMX:
// writes 2 items (32 bits) per loop pass...
//
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
LExit:
pop ebx
pop edi
ret
}
}
#endif
#else
void S_WriteLinearBlastStereo16 (void);
#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);
}
}
/*
===================
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;
int count;
// 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
===============================================================================
*/
static void S_PaintChannelFrom16( channel_t *ch, const sfx_t *sfx, int count, int sampleOffset, int bufferOffset )
{
portable_samplepair_t *pSamplesDest;
int iData;
int iLeftVol = ch->leftvol * snd_vol;
int iRightVol = ch->rightvol * snd_vol;
pSamplesDest = &paintbuffer[ bufferOffset ];
for ( int i=0 ; i<count ; i++ )
{
iData = sfx->pSoundData[ sampleOffset++ ];
pSamplesDest[i].left += (iData * iLeftVol )>>8;
pSamplesDest[i].right += (iData * iRightVol)>>8;
}
}
void S_PaintChannelFromMP3( channel_t *ch, const sfx_t *sc, int count, int sampleOffset, int bufferOffset )
{
int data;
int leftvol, rightvol;
signed short *sfx;
int i;
portable_samplepair_t *samp;
static short tempMP3Buffer[PAINTBUFFER_SIZE];
MP3Stream_GetSamples( ch, sampleOffset, count, tempMP3Buffer, qfalse ); // qfalse = not stereo
leftvol = ch->leftvol*snd_vol;
rightvol = ch->rightvol*snd_vol;
sfx = tempMP3Buffer;
samp = &paintbuffer[ bufferOffset ];
while ( count & 3 ) {
data = *sfx;
samp->left += (data * leftvol)>>8;
samp->right += (data * rightvol)>>8;
sfx++;
samp++;
count--;
}
for ( i=0 ; i<count ; i += 4 ) {
data = sfx[i];
samp[i].left += (data * leftvol)>>8;
samp[i].right += (data * rightvol)>>8;
data = sfx[i+1];
samp[i+1].left += (data * leftvol)>>8;
samp[i+1].right += (data * rightvol)>>8;
data = sfx[i+2];
samp[i+2].left += (data * leftvol)>>8;
samp[i+2].right += (data * rightvol)>>8;
data = sfx[i+3];
samp[i+3].left += (data * leftvol)>>8;
samp[i+3].right += (data * rightvol)>>8;
}
}
// subroutinised to save code dup (called twice) -ste
//
void ChannelPaint(channel_t *ch, sfx_t *sc, int count, int sampleOffset, int bufferOffset)
{
switch (sc->eSoundCompressionMethod)
{
case ct_16:
S_PaintChannelFrom16 (ch, sc, count, sampleOffset, bufferOffset);
break;
case ct_MP3:
S_PaintChannelFromMP3 (ch, sc, count, sampleOffset, bufferOffset);
break;
default:
assert(0); // debug aid, ignored in release. FIXME: Should we ERR_DROP here for badness-catch?
break;
}
}
void S_PaintChannels( int endtime ) {
int i;
int end;
channel_t *ch;
sfx_t *sc;
int ltime, count;
int sampleOffset;
snd_vol = s_volume->value*256;
//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 to either music or zeros
if ( s_rawend < s_paintedtime ) {
if ( s_rawend ) {
//Com_DPrintf ("background sound underrun\n");
}
memset(paintbuffer, 0, (end - s_paintedtime) * sizeof(portable_samplepair_t));
} else {
// copy from the streaming sound source
int s;
int stop;
stop = (end < s_rawend) ? end : s_rawend;
for ( i = s_paintedtime ; i < stop ; i++ ) {
s = i&(MAX_RAW_SAMPLES-1);
paintbuffer[i-s_paintedtime] = s_rawsamples[s];
}
// if (i != end)
// Com_Printf ("partial stream\n");
// else
// Com_Printf ("full stream\n");
for ( ; i < end ; i++ ) {
paintbuffer[i-s_paintedtime].left =
paintbuffer[i-s_paintedtime].right = 0;
}
}
// 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->iSoundLengthInSamples ) {
count = sc->iSoundLengthInSamples - sampleOffset;
}
if ( count > 0 ) {
ChannelPaint(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->pSoundData == NULL || sc->iSoundLengthInSamples == 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->iSoundLengthInSamples);
count = end - ltime;
if ( sampleOffset + count > sc->iSoundLengthInSamples) {
count = sc->iSoundLengthInSamples - sampleOffset;
}
if ( count > 0 ) {
ChannelPaint(ch, sc, count, sampleOffset, ltime - s_paintedtime);
ltime += count;
}
} while ( ltime < end);
}
// transfer out according to DMA format
S_TransferPaintBuffer( end );
s_paintedtime = end;
}
}