mirror of
https://github.com/DrBeef/JKXR.git
synced 2024-11-23 20:43:09 +00:00
4597b03873
Opens in Android Studio but haven't even tried to build it yet (it won't.. I know that much!)
392 lines
8.9 KiB
C++
392 lines
8.9 KiB
C++
/*
|
|
===========================================================================
|
|
Copyright (C) 1999 - 2005, Id Software, Inc.
|
|
Copyright (C) 2000 - 2013, Raven Software, Inc.
|
|
Copyright (C) 2001 - 2013, Activision, Inc.
|
|
Copyright (C) 2013 - 2015, OpenJK contributors
|
|
|
|
This file is part of the OpenJK source code.
|
|
|
|
OpenJK is free software; you can redistribute it and/or modify it
|
|
under the terms of the GNU General Public License version 2 as
|
|
published by the Free Software Foundation.
|
|
|
|
This program 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 this program; if not, see <http://www.gnu.org/licenses/>.
|
|
===========================================================================
|
|
*/
|
|
|
|
// snd_mix.c -- portable code to mix sounds for snd_dma.c
|
|
|
|
#include "../server/exe_headers.h"
|
|
|
|
#include "snd_local.h"
|
|
|
|
portable_samplepair_t paintbuffer[PAINTBUFFER_SIZE];
|
|
int *snd_p, snd_linear_count, snd_vol;
|
|
short *snd_out;
|
|
|
|
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;
|
|
}
|
|
}
|
|
|
|
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 = (int)(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 < (short)0x8000)
|
|
val = (short)0x8000;
|
|
out[out_idx] = (short)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 < (short)0x8000)
|
|
val = (short)0x8000;
|
|
out[out_idx] = (short)((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;
|
|
int normal_vol,voice_vol;
|
|
|
|
snd_vol = normal_vol = s_volume->value*256.0f;
|
|
voice_vol = (s_volumeVoice->value*256.0f);
|
|
|
|
//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;
|
|
}
|
|
|
|
if ( ch->entchannel == CHAN_VOICE || ch->entchannel == CHAN_VOICE_ATTEN || ch->entchannel == CHAN_VOICE_GLOBAL )
|
|
snd_vol = voice_vol;
|
|
else
|
|
snd_vol = normal_vol;
|
|
|
|
ltime = s_paintedtime;
|
|
sc = ch->thesfx;
|
|
|
|
// we might have to make 2 passes if it is
|
|
// a looping sound effect and the end of
|
|
// the sameple is hit...
|
|
//
|
|
do
|
|
{
|
|
if (ch->loopSound) {
|
|
sampleOffset = ltime % sc->iSoundLengthInSamples;
|
|
} else {
|
|
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);
|
|
ltime += count;
|
|
}
|
|
} while ( ltime < end && ch->loopSound );
|
|
}
|
|
/* temprem
|
|
// 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 )
|
|
{
|
|
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;
|
|
}
|
|
}
|