mirror of
https://github.com/DrBeef/ioq3quest.git
synced 2024-11-14 16:40:48 +00:00
274 lines
6.9 KiB
C
274 lines
6.9 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
|
|
===========================================================================
|
|
*/
|
|
|
|
/*****************************************************************************
|
|
* name: snd_mem.c
|
|
*
|
|
* desc: sound caching
|
|
*
|
|
* $Archive: /MissionPack/code/client/snd_mem.c $
|
|
*
|
|
*****************************************************************************/
|
|
|
|
#include "snd_local.h"
|
|
#include "snd_codec.h"
|
|
|
|
#define DEF_COMSOUNDMEGS "8"
|
|
|
|
/*
|
|
===============================================================================
|
|
|
|
memory management
|
|
|
|
===============================================================================
|
|
*/
|
|
|
|
static sndBuffer *buffer = NULL;
|
|
static sndBuffer *freelist = NULL;
|
|
static int inUse = 0;
|
|
static int totalInUse = 0;
|
|
|
|
short *sfxScratchBuffer = NULL;
|
|
sfx_t *sfxScratchPointer = NULL;
|
|
int sfxScratchIndex = 0;
|
|
|
|
void SND_free(sndBuffer *v) {
|
|
*(sndBuffer **)v = freelist;
|
|
freelist = (sndBuffer*)v;
|
|
inUse += sizeof(sndBuffer);
|
|
}
|
|
|
|
sndBuffer* SND_malloc(void) {
|
|
sndBuffer *v;
|
|
redo:
|
|
if (freelist == NULL) {
|
|
S_FreeOldestSound();
|
|
goto redo;
|
|
}
|
|
|
|
inUse -= sizeof(sndBuffer);
|
|
totalInUse += sizeof(sndBuffer);
|
|
|
|
v = freelist;
|
|
freelist = *(sndBuffer **)freelist;
|
|
v->next = NULL;
|
|
return v;
|
|
}
|
|
|
|
void SND_setup(void) {
|
|
sndBuffer *p, *q;
|
|
cvar_t *cv;
|
|
int scs;
|
|
|
|
cv = Cvar_Get( "com_soundMegs", DEF_COMSOUNDMEGS, CVAR_LATCH | CVAR_ARCHIVE );
|
|
|
|
scs = (cv->integer*1536);
|
|
|
|
buffer = malloc(scs*sizeof(sndBuffer) );
|
|
// allocate the stack based hunk allocator
|
|
sfxScratchBuffer = malloc(SND_CHUNK_SIZE * sizeof(short) * 4); //Hunk_Alloc(SND_CHUNK_SIZE * sizeof(short) * 4);
|
|
sfxScratchPointer = NULL;
|
|
|
|
inUse = scs*sizeof(sndBuffer);
|
|
p = buffer;;
|
|
q = p + scs;
|
|
while (--q > p)
|
|
*(sndBuffer **)q = q-1;
|
|
|
|
*(sndBuffer **)q = NULL;
|
|
freelist = p + scs - 1;
|
|
|
|
Com_Printf("Sound memory manager started\n");
|
|
}
|
|
|
|
void SND_shutdown(void)
|
|
{
|
|
free(sfxScratchBuffer);
|
|
free(buffer);
|
|
}
|
|
|
|
/*
|
|
================
|
|
ResampleSfx
|
|
|
|
resample / decimate to the current source rate
|
|
================
|
|
*/
|
|
static int ResampleSfx( sfx_t *sfx, int channels, int inrate, int inwidth, int samples, byte *data, qboolean compressed ) {
|
|
int outcount;
|
|
int srcsample;
|
|
float stepscale;
|
|
int i, j;
|
|
int sample, samplefrac, fracstep;
|
|
int part;
|
|
sndBuffer *chunk;
|
|
|
|
stepscale = (float)inrate / dma.speed; // this is usually 0.5, 1, or 2
|
|
|
|
outcount = samples / stepscale;
|
|
|
|
samplefrac = 0;
|
|
fracstep = stepscale * 256 * channels;
|
|
chunk = sfx->soundData;
|
|
|
|
for (i=0 ; i<outcount ; i++)
|
|
{
|
|
srcsample = samplefrac >> 8;
|
|
samplefrac += fracstep;
|
|
for (j=0 ; j<channels ; j++)
|
|
{
|
|
if( inwidth == 2 ) {
|
|
sample = ( ((short *)data)[srcsample+j] );
|
|
} else {
|
|
sample = (int)( (unsigned char)(data[srcsample+j]) - 128) << 8;
|
|
}
|
|
part = (i*channels+j)&(SND_CHUNK_SIZE-1);
|
|
if (part == 0) {
|
|
sndBuffer *newchunk;
|
|
newchunk = SND_malloc();
|
|
if (chunk == NULL) {
|
|
sfx->soundData = newchunk;
|
|
} else {
|
|
chunk->next = newchunk;
|
|
}
|
|
chunk = newchunk;
|
|
}
|
|
|
|
chunk->sndChunk[part] = sample;
|
|
}
|
|
}
|
|
|
|
return outcount;
|
|
}
|
|
|
|
/*
|
|
================
|
|
ResampleSfx
|
|
|
|
resample / decimate to the current source rate
|
|
================
|
|
*/
|
|
static int ResampleSfxRaw( short *sfx, int channels, int inrate, int inwidth, int samples, byte *data ) {
|
|
int outcount;
|
|
int srcsample;
|
|
float stepscale;
|
|
int i, j;
|
|
int sample, samplefrac, fracstep;
|
|
|
|
stepscale = (float)inrate / dma.speed; // this is usually 0.5, 1, or 2
|
|
|
|
outcount = samples / stepscale;
|
|
|
|
samplefrac = 0;
|
|
fracstep = stepscale * 256 * channels;
|
|
|
|
for (i=0 ; i<outcount ; i++)
|
|
{
|
|
srcsample = samplefrac >> 8;
|
|
samplefrac += fracstep;
|
|
for (j=0 ; j<channels ; j++)
|
|
{
|
|
if( inwidth == 2 ) {
|
|
sample = LittleShort ( ((short *)data)[srcsample+j] );
|
|
} else {
|
|
sample = (int)( (unsigned char)(data[srcsample+j]) - 128) << 8;
|
|
}
|
|
sfx[i*channels+j] = sample;
|
|
}
|
|
}
|
|
return outcount;
|
|
}
|
|
|
|
//=============================================================================
|
|
|
|
/*
|
|
==============
|
|
S_LoadSound
|
|
|
|
The filename may be different than sfx->name in the case
|
|
of a forced fallback of a player specific sound
|
|
==============
|
|
*/
|
|
qboolean S_LoadSound( sfx_t *sfx )
|
|
{
|
|
byte *data;
|
|
short *samples;
|
|
snd_info_t info;
|
|
// int size;
|
|
|
|
// load it in
|
|
data = S_CodecLoad(sfx->soundName, &info);
|
|
if(!data)
|
|
return qfalse;
|
|
|
|
if ( info.width == 1 ) {
|
|
Com_DPrintf(S_COLOR_YELLOW "WARNING: %s is a 8 bit audio file\n", sfx->soundName);
|
|
}
|
|
|
|
if ( info.rate != 22050 ) {
|
|
Com_DPrintf(S_COLOR_YELLOW "WARNING: %s is not a 22kHz audio file\n", sfx->soundName);
|
|
}
|
|
|
|
samples = Hunk_AllocateTempMemory(info.channels * info.samples * sizeof(short) * 2);
|
|
|
|
sfx->lastTimeUsed = Com_Milliseconds()+1;
|
|
|
|
// each of these compression schemes works just fine
|
|
// but the 16bit quality is much nicer and with a local
|
|
// install assured we can rely upon the sound memory
|
|
// manager to do the right thing for us and page
|
|
// sound in as needed
|
|
|
|
if( info.channels == 1 && sfx->soundCompressed == qtrue) {
|
|
sfx->soundCompressionMethod = 1;
|
|
sfx->soundData = NULL;
|
|
sfx->soundLength = ResampleSfxRaw( samples, info.channels, info.rate, info.width, info.samples, data + info.dataofs );
|
|
S_AdpcmEncodeSound(sfx, samples);
|
|
#if 0
|
|
} else if (info.channels == 1 && info.samples>(SND_CHUNK_SIZE*16) && info.width >1) {
|
|
sfx->soundCompressionMethod = 3;
|
|
sfx->soundData = NULL;
|
|
sfx->soundLength = ResampleSfxRaw( samples, info.channels, info.rate, info.width, info.samples, (data + info.dataofs) );
|
|
encodeMuLaw( sfx, samples);
|
|
} else if (info.channels == 1 && info.samples>(SND_CHUNK_SIZE*6400) && info.width >1) {
|
|
sfx->soundCompressionMethod = 2;
|
|
sfx->soundData = NULL;
|
|
sfx->soundLength = ResampleSfxRaw( samples, info.channels, info.rate, info.width, info.samples, (data + info.dataofs) );
|
|
encodeWavelet( sfx, samples);
|
|
#endif
|
|
} else {
|
|
sfx->soundCompressionMethod = 0;
|
|
sfx->soundData = NULL;
|
|
sfx->soundLength = ResampleSfx( sfx, info.channels, info.rate, info.width, info.samples, data + info.dataofs, qfalse );
|
|
}
|
|
|
|
sfx->soundChannels = info.channels;
|
|
|
|
Hunk_FreeTempMemory(samples);
|
|
Hunk_FreeTempMemory(data);
|
|
|
|
return qtrue;
|
|
}
|
|
|
|
void S_DisplayFreeMemory(void) {
|
|
Com_Printf("%d bytes free sound buffer memory, %d total used\n", inUse, totalInUse);
|
|
}
|