Xash3D FWGS engine.
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/*
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;
}
#define drint( v ) (int)( v + 0.5 )
/*
================
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 )
{
double stepscale, j;
int outcount;
int i;
qboolean handled = false;
if( inrate == outrate && inwidth == outwidth )
return false;
stepscale = (double)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;
if( inrate == outrate )
{
if( inwidth == 1 && outwidth == 2 ) // S8 to S16
{
for( i = 0; i < outcount * sc->channels; i++ )
((int16_t*)sound.tempbuffer)[i] = ((int8_t *)sc->buffer)[i] * 256;
handled = true;
}
else if( inwidth == 2 && outwidth == 1 ) // S16 to S8
{
for( i = 0; i < outcount * sc->channels; i++ )
((int8_t*)sound.tempbuffer)[i] = ((int16_t *)sc->buffer)[i] / 256;
handled = true;
}
}
else // resample case
{
if( inwidth == 1 )
{
int8_t *data = (int8_t *)sc->buffer;
if( outwidth == 1 )
{
if( sc->channels == 2 )
{
for( i = 0, j = 0; i < outcount; i++, j += stepscale )
{
((int8_t*)sound.tempbuffer)[i*2+0] = data[((int)j)*2+0];
((int8_t*)sound.tempbuffer)[i*2+1] = data[((int)j)*2+1];
}
}
else
{
for( i = 0, j = 0; i < outcount; i++, j += stepscale )
((int8_t*)sound.tempbuffer)[i] = data[(int)j];
}
handled = true;
}
else if( outwidth == 2 )
{
if( sc->channels == 2 )
{
for( i = 0, j = 0; i < outcount; i++, j += stepscale )
{
((int16_t*)sound.tempbuffer)[i*2+0] = data[((int)j)*2+0] * 256;
((int16_t*)sound.tempbuffer)[i*2+1] = data[((int)j)*2+1] * 256;
}
}
else
{
for( i = 0, j = 0; i < outcount; i++, j += stepscale )
((int16_t*)sound.tempbuffer)[i] = data[(int)j] * 256;
}
handled = true;
}
}
else if( inwidth == 2 )
{
int16_t *data = (int16_t *)sc->buffer;
if( outwidth == 1 )
{
if( sc->channels == 2 )
{
for( i = 0, j = 0; i < outcount; i++, j += stepscale )
{
((int8_t*)sound.tempbuffer)[i*2+0] = data[((int)j)*2+0] / 256;
((int8_t*)sound.tempbuffer)[i*2+1] = data[((int)j)*2+1] / 256;
}
}
else
{
for( i = 0, j = 0; i < outcount; i++, j += stepscale )
((int8_t*)sound.tempbuffer)[i] = data[(int)j] / 256;
}
handled = true;
}
else if( outwidth == 2 )
{
if( sc->channels == 2 )
{
for( i = 0, j = 0; i < outcount; i++, j += stepscale )
{
((int16_t*)sound.tempbuffer)[i*2+0] = data[((int)j)*2+0];
((int16_t*)sound.tempbuffer)[i*2+1] = data[((int)j)*2+1];
}
}
else
{
for( i = 0, j = 0; i < outcount; i++, j += stepscale )
((int16_t*)sound.tempbuffer)[i] = data[(int)j];
}
handled = true;
}
}
}
if( handled )
Con_Reportf( "Sound_Resample: from [%d bit %d Hz] to [%d bit %d Hz]\n", inwidth * 8, inrate, outwidth * 8, outrate );
else
Con_Reportf( S_ERROR "Sound_Resample: unsupported from [%d bit %d Hz] to [%d bit %d Hz]\n", inwidth * 8, inrate, outwidth * 8, outrate );
sc->rate = outrate;
sc->width = outwidth;
return handled;
}
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;
}