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Code Issues 2 Projects Releases Wiki Activity

engine: soundlib: add support for converting stereo to mono and back, attempt to simplify conversion boilerplate

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This commit is contained in:
Alibek Omarov 2024-07-15 12:44:50 +03:00
parent 56ba2325c1
commit d3586c2143
4 changed files with 236 additions and 307 deletions
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6
engine/client/s_load.c
View File

@ -155,11 +155,11 @@ wavdata_t *S_LoadSound( sfx_t *sfx )
if( !sc ) sc = S_CreateDefaultSound();
if( sc->rate < SOUND_11k ) // some bad sounds
Sound_Process( &sc, SOUND_11k, sc->width, SOUND_RESAMPLE );
Sound_Process( &sc, SOUND_11k, sc->width, sc->channels, SOUND_RESAMPLE );
else if( sc->rate > SOUND_11k && sc->rate < SOUND_22k ) // some bad sounds
Sound_Process( &sc, SOUND_22k, sc->width, SOUND_RESAMPLE );
Sound_Process( &sc, SOUND_22k, sc->width, sc->channels, SOUND_RESAMPLE );
else if( sc->rate > SOUND_22k && sc->rate < SOUND_44k ) // some bad sounds
Sound_Process( &sc, SOUND_44k, sc->width, SOUND_RESAMPLE );
Sound_Process( &sc, SOUND_44k, sc->width, sc->channels, SOUND_RESAMPLE );
sfx->cache = sc;

2
engine/client/voice.c
View File

@ -397,7 +397,7 @@ void Voice_RecordStart( void )
if( voice.input_file )
{
Sound_Process( &voice.input_file, voice.samplerate, voice.width, SOUND_RESAMPLE );
Sound_Process( &voice.input_file, voice.samplerate, voice.width, VOICE_PCM_CHANNELS, SOUND_RESAMPLE );
voice.input_file_pos = 0;
voice.start_time = Sys_DoubleTime();

2
engine/common/common.h
View File

@ -492,7 +492,7 @@ int FS_ReadStream( stream_t *stream, int bytes, void *buffer );
int FS_SetStreamPos( stream_t *stream, int newpos );
int FS_GetStreamPos( stream_t *stream );
void FS_FreeStream( stream_t *stream );
qboolean Sound_Process( wavdata_t **wav, int rate, int width, uint flags );
qboolean Sound_Process( wavdata_t **wav, int rate, int width, int channels, uint flags );
uint Sound_GetApproxWavePlayLen( const char *filepath );
qboolean Sound_SupportedFileFormat( const char *fileext );

533
engine/common/soundlib/snd_utils.c
View File

@ -133,346 +133,275 @@ uint GAME_EXPORT Sound_GetApproxWavePlayLen( const char *filepath )
return msecs;
}
static qboolean Sound_ConvertNoResample( wavdata_t *sc, int inwidth, int outwidth, int outcount )
#define SOUND_FORMATCONVERT_BOILERPLATE( resamplemacro ) \
if( inwidth == 1 ) \
{ \
const int8_t *data = (const int8_t *)sc->buffer; \
if( outwidth == 1 ) \
{ \
int8_t *outdata = (int8_t *)sound.tempbuffer; \
resamplemacro( 1 ) \
} \
else if( outwidth == 2 ) \
{ \
int16_t *outdata = (int16_t *)sound.tempbuffer; \
resamplemacro( 256 ) \
} \
else \
return false; \
} \
else if( inwidth == 2 ) \
{ \
const int16_t *data = (const int16_t *)sc->buffer; \
if( outwidth == 1 ) \
{ \
int8_t *outdata = (int8_t *)sound.tempbuffer; \
resamplemacro( 1 / 256.0 ) \
} \
else if( outwidth == 2 ) \
{ \
int16_t *outdata = (int16_t *)sound.tempbuffer; \
resamplemacro( 1 ) \
} \
else \
return false; \
} \
else \
return false; \
#define SOUND_CONVERTNORESAMPLE_BOILERPLATE( multiplier ) \
if( inchannels == 1 ) \
{ \
if( outchannels == 1 ) \
{ \
for( i = 0; i < outcount * outchannels; i++ ) \
outdata[i] = data[i] * ( multiplier ); \
} \
else if( outchannels == 2 ) \
{ \
for( i = 0; i < outcount; i++ ) \
{ \
outdata[i * 2 + 0] = data[i] * ( multiplier ); \
outdata[i * 2 + 1] = data[i] * ( multiplier ); \
} \
} \
else \
return false; \
} \
else if( inchannels == 2 ) \
{ \
if( outchannels == 1 ) \
{ \
for( i = 0; i < outcount; i++ ) \
outdata[i] = ( data[i * 2 + 0] + data[i * 2 + 1] ) * ( multiplier ) / 2; \
} \
else if( outchannels == 2 ) \
{ \
for( i = 0; i < outcount * outchannels; i++ ) \
outdata[i] = data[i] * ( multiplier ); \
} \
else \
return false; \
} \
else \
return false; \
#define SOUND_CONVERTDOWNSAMPLE_BOILERPLATE( multiplier ) \
if( inchannels == 1 ) \
{ \
if( outchannels == 1 ) \
{ \
for( i = 0; i < outcount; i++ ) \
{ \
double j = stepscale * i; \
outdata[i] = data[(int)j] * ( multiplier ); \
} \
} \
else if( outchannels == 2 ) \
{ \
for( i = 0; i < outcount; i++ ) \
{ \
double j = stepscale * i;\
outdata[i * 2 + 0] = data[(int)j] * ( multiplier ); \
outdata[i * 2 + 1] = data[(int)j] * ( multiplier ); \
} \
} \
else \
return false; \
} \
else if( inchannels == 2 ) \
{ \
if( outchannels == 1 ) \
{ \
for( i = 0; i < outcount; i++ ) \
{ \
double j = stepscale * i; \
outdata[i] = ( data[((int)j) * 2 + 0] + data[((int)j) * 2 + 1] ) * ( multiplier ) / 2; \
} \
} \
else if( outchannels == 2 ) \
{ \
for( i = 0; i < outcount; i++ ) \
{ \
double j = stepscale * i; \
outdata[i * 2 + 0] = data[((int)j) * 2 + 0] * ( multiplier ); \
outdata[i * 2 + 1] = data[((int)j) * 2 + 1] * ( multiplier ); \
} \
} \
else \
return false; \
} \
else \
return false; \
#define SOUND_CONVERTUPSAMPLE_BOILERPLATE( multiplier ) \
if( inchannels == 1 ) \
{ \
if( outchannels == 1 ) \
{ \
for( i = 0; i < outcount; i++ ) \
{ \
double j = stepscale * i; \
outdata[i] = data[(int)j] * ( multiplier ); \
if( j != (int)j && j < incount ) \
{ \
double frac = ( j - (int)j ) * ( multiplier ); \
outdata[i] += (data[(int)j+1] - data[(int)j]) * frac; \
} \
} \
} \
else if( outchannels == 2 ) \
{ \
for( i = 0; i < outcount; i++ ) \
{ \
double j = stepscale * i; \
outdata[i * 2 + 0] = data[(int)j] * ( multiplier ); \
if( j != (int)j && j < incount ) \
{ \
double frac = ( j - (int)j ) * ( multiplier ); \
outdata[i * 2 + 0] += (data[(int)j+1] - data[(int)j]) * frac; \
} \
outdata[i * 2 + 1] = outdata[i * 2 + 0]; \
} \
} \
else \
return false; \
} \
else if( inchannels == 2 ) \
{ \
if( outchannels == 1 ) \
{ \
for( i = 0; i < outcount; i++ ) \
{ \
double j = stepscale * i; \
outdata[i] = ( data[((int)j) * 2 + 0] + data[((int)j) * 2 + 1] ) * ( multiplier ) / 2; \
if( j != (int)j && j < incount ) \
{ \
double frac = ( j - (int)j ) * ( multiplier ) / 2; \
outdata[i] += (data[((int)j + 1 ) * 2 + 0] - data[((int)j) * 2 + 0]) * frac; \
outdata[i] += (data[((int)j + 1 ) * 2 + 1] - data[((int)j) * 2 + 1]) * frac; \
} \
} \
} \
else if( outchannels == 2 ) \
{ \
for( i = 0; i < outcount; i++ ) \
{ \
double j = stepscale * i; \
outdata[i*2+0] = data[((int)j)*2+0] * ( multiplier ); \
outdata[i*2+1] = data[((int)j)*2+1] * ( multiplier ); \
if( j != (int)j && j < incount ) \
{ \
double frac = ( j - (int)j ) * ( multiplier ); \
outdata[i*2+0] += (data[((int)j+1)*2+0] - data[((int)j)*2+0]) * frac; \
outdata[i*2+1] += (data[((int)j+1)*2+1] - data[((int)j)*2+1]) * frac; \
} \
} \
} \
else \
return false; \
} \
else \
return false; \
static qboolean Sound_ConvertNoResample( wavdata_t *sc, int inwidth, int inchannels, int outwidth, int outchannels, int outcount )
{
size_t i;
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;
return true;
}
// I could write a generic case here but I also want to make it easier for compiler
// to unroll these for-loops
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;
return true;
}
SOUND_FORMATCONVERT_BOILERPLATE( SOUND_CONVERTNORESAMPLE_BOILERPLATE )
return false;
return true;
}
static qboolean Sound_ConvertDownsample( wavdata_t *sc, int inwidth, int outwidth, int outcount, double stepscale )
static qboolean Sound_ConvertDownsample( wavdata_t *sc, int inwidth, int inchannels, int outwidth, int outchannels, int outcount, double stepscale )
{
size_t i;
double j;
if( inwidth == 1 && outwidth == 1 )
{
int8_t *data = (int8_t *)sc->buffer;
SOUND_FORMATCONVERT_BOILERPLATE( SOUND_CONVERTDOWNSAMPLE_BOILERPLATE )
if( outwidth == 1 )
{
int8_t *outdata = (int8_t *)sound.tempbuffer;
if( sc->channels == 2 )
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i*2+0] = data[((int)j)*2+0];
outdata[i*2+1] = data[((int)j)*2+1];
}
}
else
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i] = data[(int)j];
}
}
return true;
}
if( outwidth == 2 )
{
int16_t *outdata = (int16_t *)sound.tempbuffer;
if( sc->channels == 2 )
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i*2+0] = data[((int)j)*2+0] * 256;
outdata[i*2+1] = data[((int)j)*2+1] * 256;
}
}
else
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i] = data[(int)j] * 256;
}
}
return true;
}
}
if( inwidth == 2 )
{
int16_t *data = (int16_t *)sc->buffer;
if( outwidth == 1 )
{
int8_t *outdata = (int8_t *)sound.tempbuffer;
if( sc->channels == 2 )
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i*2+0] = data[((int)j)*2+0] / 256;
outdata[i*2+1] = data[((int)j)*2+1] / 256;
}
}
else
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i] = data[(int)j] / 256;
}
}
return true;
}
if( outwidth == 2 )
{
int16_t *outdata = (int16_t *)sound.tempbuffer;
if( sc->channels == 2 )
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i*2+0] = data[((int)j)*2+0];
outdata[i*2+1] = data[((int)j)*2+1];
}
}
else
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i] = data[(int)j];
}
}
return true;
}
}
return false;
return true;
}
static qboolean Sound_ConvertUpsample( wavdata_t *sc, int inwidth, int outwidth, int outcount, int incount, double stepscale )
static qboolean Sound_ConvertUpsample( wavdata_t *sc, int inwidth, int inchannels, int incount, int outwidth, int outchannels, int outcount, double stepscale )
{
size_t i;
double j;
double frac;
incount--; // to not go past last sample while interpolating
if( inwidth == 1 )
{
int8_t *data = (int8_t *)sc->buffer;
SOUND_FORMATCONVERT_BOILERPLATE( SOUND_CONVERTUPSAMPLE_BOILERPLATE )
if( outwidth == 1 )
{
int8_t *outdata = (int8_t *)sound.tempbuffer;
if( sc->channels == 2 )
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i*2+0] = data[((int)j)*2+0];
outdata[i*2+1] = data[((int)j)*2+1];
if( j != (int)j && j < incount )
{
frac = j - (int)j;
outdata[i*2+0] += (data[((int)j+1)*2+0] - data[((int)j)*2+0]) * frac;
outdata[i*2+1] += (data[((int)j+1)*2+1] - data[((int)j)*2+1]) * frac;
}
}
}
else
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i] = data[(int)j];
if( j != (int)j && j < incount )
{
frac = j - (int)j;
outdata[i] += (data[(int)j+1] - data[(int)j]) * frac;
}
}
}
return true;
}
if( outwidth == 2 )
{
int16_t *outdata = (int16_t *)sound.tempbuffer;
if( sc->channels == 2 )
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i*2+0] = data[((int)j)*2+0] * 256;
outdata[i*2+1] = data[((int)j)*2+1] * 256;
if( j != (int)j && j < incount )
{
frac = ( j - (int)j ) * 256;
outdata[i*2+0] += (data[((int)j+1)*2+0] - data[((int)j)*2+0]) * frac;
outdata[i*2+1] += (data[((int)j+1)*2+1] - data[((int)j)*2+1]) * frac;
}
}
}
else
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i] = data[(int)j] * 256;
if( j != (int)j && j < incount )
{
frac = ( j - (int)j ) * 256;
outdata[i] += (data[(int)j+1] - data[(int)j]) * frac;
}
}
}
return true;
}
}
if( inwidth == 2 )
{
int16_t *data = (int16_t *)sc->buffer;
if( outwidth == 1 )
{
int8_t *outdata = (int8_t *)sound.tempbuffer;
if( sc->channels == 2 )
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i*2+0] = data[((int)j)*2+0] / 256;
outdata[i*2+1] = data[((int)j)*2+1] / 256;
if( j != (int)j && j < incount )
{
frac = ( j - (int)j ) / 256;
outdata[i*2+0] += (data[((int)j+1)*2+0] - data[((int)j)*2+0]) * frac;
outdata[i*2+1] += (data[((int)j+1)*2+1] - data[((int)j)*2+1]) * frac;
}
}
}
else
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i] = data[(int)j] / 256;
if( j != (int)j && j < incount )
{
frac = ( j - (int)j ) / 256;
outdata[i] += (data[(int)j+1] - data[(int)j]) * frac;
}
}
}
return true;
}
if( outwidth == 2 )
{
int16_t *outdata = (int16_t *)sound.tempbuffer;
if( sc->channels == 2 )
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i*2+0] = data[((int)j)*2+0];
outdata[i*2+1] = data[((int)j)*2+1];
if( j != (int)j && j < incount )
{
frac = j - (int)j;
outdata[i*2+0] += (data[((int)j+1)*2+0] - data[((int)j)*2+0]) * frac;
outdata[i*2+1] += (data[((int)j+1)*2+1] - data[((int)j)*2+1]) * frac;
}
}
}
else
{
for( i = 0; i < outcount; i++ )
{
j = stepscale * i;
outdata[i] = data[(int)j];
if( j != (int)j && j < incount )
{
frac = j - (int)j;
outdata[i] += (data[(int)j+1] - data[(int)j]) * frac;
}
}
}
return true;
}
}
return false;
return true;
}
#undef SOUND_FORMATCONVERT_BOILERPLATE
#undef SOUND_CONVERTNORESAMPLE_BOILERPLATE
#undef SOUND_CONVERTDOWNSAMPLE_BOILERPLATE
#undef SOUND_CONVERTUPSAMPLE_BOILERPLATE
/*
================
Sound_ResampleInternal
We need convert sound to signed even if nothing to resample
================
*/
static qboolean Sound_ResampleInternal( wavdata_t *sc, int inrate, int inwidth, int outrate, int outwidth )
static qboolean Sound_ResampleInternal( wavdata_t *sc, int outrate, int outwidth, int outchannels )
{
const size_t oldsize = sc->size;
const int inrate = sc->rate;
const int inwidth = sc->width;
const int inchannels = sc->channels;
const int incount = sc->samples;
qboolean handled = false;
double stepscale;
double t1, t2;
int outcount, incount = sc->samples;
int outcount;
if( inrate == outrate && inwidth == outwidth )
if( inrate == outrate && inwidth == outwidth && inchannels == outchannels )
return false;
t1 = Sys_DoubleTime();
stepscale = (double)inrate / outrate; // this is usually 0.5, 1, or 2
outcount = sc->samples / stepscale;
sc->size = outcount * outwidth * sc->channels;
sc->size = outcount * outwidth * outchannels;
sc->channels = outchannels;
sc->samples = outcount;
if( FBitSet( sc->flags, SOUND_LOOPED ))
sc->loopStart = sc->loopStart / stepscale;
#if 0 && XASH_SDL // slow but somewhat accurate
#if 0 && XASH_SDL // slow but somewhat accurate (wasn't updated to channel manipulation!!!)
{
const SDL_AudioFormat infmt = inwidth == 1 ? AUDIO_S8 : AUDIO_S16;
const SDL_AudioFormat outfmt = outwidth == 1 ? AUDIO_S8 : AUDIO_S16;
SDL_AudioCVT cvt;
// SDL_AudioCVT does conversion in place, original buffer is used for it
if( SDL_BuildAudioCVT( &cvt, infmt, sc->channels, inrate, outfmt, sc->channels, outrate ) > 0 && cvt.needed )
if( SDL_BuildAudioCVT( &cvt, infmt, inchannels, inrate, outfmt, outchannels, outrate ) > 0 && cvt.needed )
{
sc->buffer = (byte *)Mem_Realloc( host.soundpool, sc->buffer, oldsize * cvt.len_mult );
cvt.len = oldsize;
@ -493,23 +422,23 @@ static qboolean Sound_ResampleInternal( wavdata_t *sc, int inrate, int inwidth,
sound.tempbuffer = (byte *)Mem_Realloc( host.soundpool, sound.tempbuffer, sc->size );
if( inrate == outrate ) // no resampling, just copy data
handled = Sound_ConvertNoResample( sc, inwidth, outwidth, outcount );
handled = Sound_ConvertNoResample( sc, inwidth, inchannels, outwidth, outchannels, outcount );
else if( inrate > outrate ) // fast case, usually downsample but is also ok for upsampling
handled = Sound_ConvertDownsample( sc, inwidth, outwidth, outcount, stepscale );
handled = Sound_ConvertDownsample( sc, inwidth, inchannels, outwidth, outchannels, outcount, stepscale );
else // upsample case, w/ interpolation
handled = Sound_ConvertUpsample( sc, inwidth, outwidth, outcount, incount, stepscale );
handled = Sound_ConvertUpsample( sc, inwidth, inchannels, incount, outwidth, outchannels, outcount, stepscale );
t2 = Sys_DoubleTime();
if( handled )
{
if( t2 - t1 > 0.01f ) // critical, report to mod developer
Con_Printf( S_WARN "%s: from [%d bit %d Hz] to [%d bit %d Hz] (took %.3fs)\n", __func__, inwidth * 8, inrate, outwidth * 8, outrate, t2 - t1 );
Con_Printf( S_WARN "%s: from [%d bit %d Hz %dch] to [%d bit %d Hz %dch] (took %.3fs)\n", __func__, inwidth * 8, inrate, inchannels, outwidth * 8, outrate, outchannels, t2 - t1 );
else
Con_Reportf( "%s: from [%d bit %d Hz] to [%d bit %d Hz] (took %.3fs)\n", __func__, inwidth * 8, inrate, outwidth * 8, outrate, t2 - t1 );
Con_Reportf( "%s: from [%d bit %d Hz %dch] to [%d bit %d Hz %dch] (took %.3fs)\n", __func__, inwidth * 8, inrate, inchannels, outwidth * 8, outrate, outchannels, t2 - t1 );
}
else
Con_Printf( S_ERROR "%s: unsupported from [%d bit %d Hz] to [%d bit %d Hz]\n", __func__, inwidth * 8, inrate, outwidth * 8, outrate );
Con_Printf( S_ERROR "%s: unsupported from [%d bit %d Hz %dch] to [%d bit %d Hz %dch]\n", __func__, inwidth * 8, inrate, inchannels, outwidth * 8, outrate, outchannels );
sc->rate = outrate;
sc->width = outwidth;
@ -517,23 +446,23 @@ static qboolean Sound_ResampleInternal( wavdata_t *sc, int inrate, int inwidth,
return handled;
}
qboolean Sound_Process( wavdata_t **wav, int rate, int width, uint flags )
qboolean Sound_Process( wavdata_t **wav, int rate, int width, int channels, uint flags )
{
wavdata_t *snd = *wav;
qboolean result = true;
// check for buffers
if( !snd || !snd->buffer )
if( unlikely( !snd || !snd->buffer ))
return false;
if( FBitSet( flags, SOUND_RESAMPLE ) && ( width > 0 || rate > 0 ))
if( likely( FBitSet( flags, SOUND_RESAMPLE ) && ( width > 0 || rate > 0 || channels > 0 )))
{
result = Sound_ResampleInternal( snd, snd->rate, snd->width, rate, width );
result = Sound_ResampleInternal( snd, rate, width, channels );
if( result )
{
Mem_Free( snd->buffer ); // free original image buffer
snd->buffer = Sound_Copy( snd->size ); // unzone buffer (don't touch sound.tempbuffer)
snd->buffer = Sound_Copy( snd->size ); // unzone buffer
}
}