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engine: soundlib: rewrite sfx resampler, fix possible crash if sfx is too long

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- make same rate and same width resamples noop, as everything signed now
- minimize comparisons in loop body
This commit is contained in:
Alibek Omarov 2022-08-22 10:14:01 +03:00
parent cb34c23844
commit f435a81c97
1 changed files with 108 additions and 87 deletions
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195
engine/common/soundlib/snd_utils.c
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@ -126,31 +126,7 @@ uint GAME_EXPORT Sound_GetApproxWavePlayLen( const char *filepath )
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);
}
}
#define drint( v ) (int)( v + 0.5 )
/*
================
@ -161,13 +137,15 @@ 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;
double stepscale, j;
int outcount;
int i;
qboolean handled = false;
data = sc->buffer;
stepscale = (float)inrate / outrate; // this is usually 0.5, 1, or 2
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;
@ -177,69 +155,112 @@ qboolean Sound_ResampleInternal( wavdata_t *sc, int inrate, int inwidth, int out
if( sc->loopStart != -1 )
sc->loopStart = sc->loopStart / stepscale;
// resample / decimate to the current source rate
if( stepscale == 1.0f && inwidth == 1 && outwidth == 1 )
if( inrate == outrate )
{
Sound_ConvertToSigned( data, sc->channels, outcount );
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
{
// 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 Hz] to [%d bit %d Hz]\n", inwidth * 8, inrate, outwidth * 8, outrate );
}
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 true;
return handled;
}
qboolean Sound_Process( wavdata_t **wav, int rate, int width, uint flags )