mirror of
https://github.com/w23/xash3d-fwgs
synced 2024-11-15 22:46:23 +01:00
272 lines
6.3 KiB
C
272 lines
6.3 KiB
C
/*
|
|
snd_utils.c - sound common tools
|
|
Copyright (C) 2010 Uncle Mike
|
|
|
|
This program 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 3 of the License, or
|
|
(at your option) any later version.
|
|
|
|
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.
|
|
*/
|
|
|
|
#include "soundlib.h"
|
|
|
|
/*
|
|
=============================================================================
|
|
|
|
XASH3D LOAD SOUND FORMATS
|
|
|
|
=============================================================================
|
|
*/
|
|
// stub
|
|
static const loadwavfmt_t load_null[] =
|
|
{
|
|
{ NULL, NULL, NULL }
|
|
};
|
|
|
|
static const loadwavfmt_t load_game[] =
|
|
{
|
|
{ DEFAULT_SOUNDPATH "%s%s.%s", "wav", Sound_LoadWAV },
|
|
{ "%s%s.%s", "wav", Sound_LoadWAV },
|
|
{ DEFAULT_SOUNDPATH "%s%s.%s", "mp3", Sound_LoadMPG },
|
|
{ "%s%s.%s", "mp3", Sound_LoadMPG },
|
|
{ NULL, NULL, NULL }
|
|
};
|
|
|
|
/*
|
|
=============================================================================
|
|
|
|
XASH3D PROCESS STREAM FORMATS
|
|
|
|
=============================================================================
|
|
*/
|
|
// stub
|
|
static const streamfmt_t stream_null[] =
|
|
{
|
|
{ NULL, NULL, NULL, NULL, NULL, NULL, NULL }
|
|
};
|
|
|
|
static const streamfmt_t stream_game[] =
|
|
{
|
|
{ "%s%s.%s", "mp3", Stream_OpenMPG, Stream_ReadMPG, Stream_SetPosMPG, Stream_GetPosMPG, Stream_FreeMPG },
|
|
{ "%s%s.%s", "wav", Stream_OpenWAV, Stream_ReadWAV, Stream_SetPosWAV, Stream_GetPosWAV, Stream_FreeWAV },
|
|
{ NULL, NULL, NULL, NULL, NULL, NULL, NULL }
|
|
};
|
|
|
|
void Sound_Init( void )
|
|
{
|
|
// init pools
|
|
host.soundpool = Mem_AllocPool( "SoundLib Pool" );
|
|
|
|
// install image formats (can be re-install later by Sound_Setup)
|
|
switch( host.type )
|
|
{
|
|
case HOST_NORMAL:
|
|
sound.loadformats = load_game;
|
|
sound.streamformat = stream_game;
|
|
break;
|
|
default: // all other instances not using soundlib or will be reinstalling later
|
|
sound.loadformats = load_null;
|
|
sound.streamformat = stream_null;
|
|
break;
|
|
}
|
|
sound.tempbuffer = NULL;
|
|
}
|
|
|
|
void Sound_Shutdown( void )
|
|
{
|
|
Mem_Check(); // check for leaks
|
|
Mem_FreePool( &host.soundpool );
|
|
}
|
|
|
|
byte *Sound_Copy( size_t size )
|
|
{
|
|
byte *out;
|
|
|
|
out = Mem_Malloc( host.soundpool, size );
|
|
memcpy( out, sound.tempbuffer, size );
|
|
|
|
return out;
|
|
}
|
|
|
|
uint GAME_EXPORT Sound_GetApproxWavePlayLen( const char *filepath )
|
|
{
|
|
file_t *f;
|
|
wavehdr_t wav;
|
|
size_t filesize;
|
|
uint msecs;
|
|
|
|
f = FS_Open( filepath, "rb", false );
|
|
if( !f )
|
|
return 0;
|
|
|
|
if( FS_Read( f, &wav, sizeof( wav )) != sizeof( wav ))
|
|
{
|
|
FS_Close( f );
|
|
return 0;
|
|
}
|
|
|
|
filesize = FS_FileLength( f );
|
|
filesize -= 128; // magic number from GoldSrc, seems to be header size
|
|
|
|
FS_Close( f );
|
|
|
|
// is real wav file ?
|
|
if( wav.riff_id != RIFFHEADER || wav.wave_id != WAVEHEADER || wav.fmt_id != FORMHEADER )
|
|
return 0;
|
|
|
|
if( wav.nAvgBytesPerSec >= 1000 )
|
|
msecs = (uint)((float)filesize / ((float)wav.nAvgBytesPerSec / 1000.0f));
|
|
else msecs = (uint)(((float)filesize / (float)wav.nAvgBytesPerSec) * 1000.0f);
|
|
|
|
return msecs;
|
|
}
|
|
|
|
/*
|
|
================
|
|
Sound_ConvertToSigned
|
|
|
|
Convert unsigned data to signed
|
|
================
|
|
*/
|
|
void Sound_ConvertToSigned( const byte *data, int channels, int samples )
|
|
{
|
|
int i;
|
|
|
|
if( channels == 2 )
|
|
{
|
|
for( i = 0; i < samples; i++ )
|
|
{
|
|
((signed char *)sound.tempbuffer)[i*2+0] = (int)((byte)(data[i*2+0]) - 128);
|
|
((signed char *)sound.tempbuffer)[i*2+1] = (int)((byte)(data[i*2+1]) - 128);
|
|
}
|
|
}
|
|
else
|
|
{
|
|
for( i = 0; i < samples; i++ )
|
|
((signed char *)sound.tempbuffer)[i] = (int)((unsigned char)(data[i]) - 128);
|
|
}
|
|
}
|
|
|
|
/*
|
|
================
|
|
Sound_ResampleInternal
|
|
|
|
We need convert sound to signed even if nothing to resample
|
|
================
|
|
*/
|
|
qboolean Sound_ResampleInternal( wavdata_t *sc, int inrate, int inwidth, int outrate, int outwidth )
|
|
{
|
|
float stepscale;
|
|
int outcount, srcsample;
|
|
int i, sample, sample2, samplefrac, fracstep;
|
|
byte *data;
|
|
|
|
data = sc->buffer;
|
|
stepscale = (float)inrate / outrate; // this is usually 0.5, 1, or 2
|
|
outcount = sc->samples / stepscale;
|
|
sc->size = outcount * outwidth * sc->channels;
|
|
|
|
sound.tempbuffer = (byte *)Mem_Realloc( host.soundpool, sound.tempbuffer, sc->size );
|
|
|
|
sc->samples = outcount;
|
|
if( sc->loopStart != -1 )
|
|
sc->loopStart = sc->loopStart / stepscale;
|
|
|
|
// resample / decimate to the current source rate
|
|
if( stepscale == 1.0f && inwidth == 1 && outwidth == 1 )
|
|
{
|
|
Sound_ConvertToSigned( data, sc->channels, outcount );
|
|
}
|
|
else
|
|
{
|
|
// general case
|
|
samplefrac = 0;
|
|
fracstep = stepscale * 256;
|
|
|
|
if( sc->channels == 2 )
|
|
{
|
|
for( i = 0; i < outcount; i++ )
|
|
{
|
|
srcsample = samplefrac >> 8;
|
|
samplefrac += fracstep;
|
|
|
|
if( inwidth == 2 )
|
|
{
|
|
sample = ((short *)data)[srcsample*2+0];
|
|
sample2 = ((short *)data)[srcsample*2+1];
|
|
}
|
|
else
|
|
{
|
|
sample = (int)((char)(data[srcsample*2+0])) << 8;
|
|
sample2 = (int)((char)(data[srcsample*2+1])) << 8;
|
|
}
|
|
|
|
if( outwidth == 2 )
|
|
{
|
|
((short *)sound.tempbuffer)[i*2+0] = sample;
|
|
((short *)sound.tempbuffer)[i*2+1] = sample2;
|
|
}
|
|
else
|
|
{
|
|
((signed char *)sound.tempbuffer)[i*2+0] = sample >> 8;
|
|
((signed char *)sound.tempbuffer)[i*2+1] = sample2 >> 8;
|
|
}
|
|
}
|
|
}
|
|
else
|
|
{
|
|
for( i = 0; i < outcount; i++ )
|
|
{
|
|
srcsample = samplefrac >> 8;
|
|
samplefrac += fracstep;
|
|
|
|
if( inwidth == 2 ) sample = ((short *)data)[srcsample];
|
|
else sample = (int)( (char)(data[srcsample])) << 8;
|
|
|
|
if( outwidth == 2 ) ((short *)sound.tempbuffer)[i] = sample;
|
|
else ((signed char *)sound.tempbuffer)[i] = sample >> 8;
|
|
}
|
|
}
|
|
|
|
Con_Reportf( "Sound_Resample: from[%d bit %d kHz] to [%d bit %d kHz]\n", inwidth * 8, inrate, outwidth * 8, outrate );
|
|
}
|
|
|
|
sc->rate = outrate;
|
|
sc->width = outwidth;
|
|
|
|
return true;
|
|
}
|
|
|
|
qboolean Sound_Process( wavdata_t **wav, int rate, int width, uint flags )
|
|
{
|
|
wavdata_t *snd = *wav;
|
|
qboolean result = true;
|
|
|
|
// check for buffers
|
|
if( !snd || !snd->buffer )
|
|
return false;
|
|
|
|
if(( flags & SOUND_RESAMPLE ) && ( width > 0 || rate > 0 ))
|
|
{
|
|
if( Sound_ResampleInternal( snd, snd->rate, snd->width, rate, width ))
|
|
{
|
|
Mem_Free( snd->buffer ); // free original image buffer
|
|
snd->buffer = Sound_Copy( snd->size ); // unzone buffer (don't touch image.tempbuffer)
|
|
}
|
|
else
|
|
{
|
|
// not resampled
|
|
result = false;
|
|
}
|
|
}
|
|
|
|
*wav = snd;
|
|
|
|
return false;
|
|
}
|