engine: soundlib: implement linear interpolation in Sound_ResampleInternal
* split resampling into three functions * added resampling using SDL_AudioCVT, it's slow, so isn't enabled by default
This commit is contained in:
Alibek Omarov
parent
e19aa001b2
commit
55fcbc8880
1 changed files with 352 additions and 104 deletions
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@ -14,6 +14,9 @@ GNU General Public License for more details.
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*/
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#include "soundlib.h"
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#if XASH_SDL
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#include <SDL_audio.h>
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#endif // XASH_SDL
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/*
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=============================================================================
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@ -130,6 +133,309 @@ uint GAME_EXPORT Sound_GetApproxWavePlayLen( const char *filepath )
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return msecs;
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}
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static qboolean Sound_ConvertNoResample( wavdata_t *sc, int inwidth, int outwidth, int outcount )
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{
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size_t i;
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if( inwidth == 1 && outwidth == 2 ) // S8 to S16
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{
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for( i = 0; i < outcount * sc->channels; i++ )
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((int16_t*)sound.tempbuffer)[i] = ((int8_t *)sc->buffer)[i] * 256;
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return true;
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}
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if( inwidth == 2 && outwidth == 1 ) // S16 to S8
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{
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for( i = 0; i < outcount * sc->channels; i++ )
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((int8_t*)sound.tempbuffer)[i] = ((int16_t *)sc->buffer)[i] / 256;
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return true;
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}
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return false;
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}
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static qboolean Sound_ConvertDownsample( wavdata_t *sc, int inwidth, int outwidth, int outcount, double stepscale )
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{
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size_t i;
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double j;
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if( inwidth == 1 && outwidth == 1 )
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{
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int8_t *data = (int8_t *)sc->buffer;
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if( outwidth == 1 )
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{
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int8_t *outdata = (int8_t *)sound.tempbuffer;
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if( sc->channels == 2 )
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i*2+0] = data[((int)j)*2+0];
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outdata[i*2+1] = data[((int)j)*2+1];
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}
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}
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else
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i] = data[(int)j];
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}
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}
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return true;
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}
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if( outwidth == 2 )
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{
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int16_t *outdata = (int16_t *)sound.tempbuffer;
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if( sc->channels == 2 )
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i*2+0] = data[((int)j)*2+0] * 256;
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outdata[i*2+1] = data[((int)j)*2+1] * 256;
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}
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}
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else
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i] = data[(int)j] * 256;
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}
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}
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return true;
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}
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}
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if( inwidth == 2 )
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{
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int16_t *data = (int16_t *)sc->buffer;
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if( outwidth == 1 )
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{
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int8_t *outdata = (int8_t *)sound.tempbuffer;
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if( sc->channels == 2 )
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i*2+0] = data[((int)j)*2+0] / 256;
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outdata[i*2+1] = data[((int)j)*2+1] / 256;
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}
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}
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else
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i] = data[(int)j] / 256;
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}
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}
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return true;
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}
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if( outwidth == 2 )
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{
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int16_t *outdata = (int16_t *)sound.tempbuffer;
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if( sc->channels == 2 )
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i*2+0] = data[((int)j)*2+0];
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outdata[i*2+1] = data[((int)j)*2+1];
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}
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}
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else
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i] = data[(int)j];
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}
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}
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return true;
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}
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}
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return false;
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}
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static qboolean Sound_ConvertUpsample( wavdata_t *sc, int inwidth, int outwidth, int outcount, double stepscale )
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{
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const int incount = ( outcount * stepscale ) - 1;
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size_t i;
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double j;
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double frac;
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if( inwidth == 1 )
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{
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int8_t *data = (int8_t *)sc->buffer;
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if( outwidth == 1 )
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{
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int8_t *outdata = (int8_t *)sound.tempbuffer;
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if( sc->channels == 2 )
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i*2+0] = data[((int)j)*2+0];
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outdata[i*2+1] = data[((int)j)*2+1];
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if( j != (int)j && j < incount )
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{
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frac = j - (int)j;
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outdata[i*2+0] += (data[((int)j+1)*2+0] - data[((int)j)*2+0]) * frac;
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outdata[i*2+1] += (data[((int)j+1)*2+1] - data[((int)j)*2+1]) * frac;
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}
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}
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}
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else
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i] = data[(int)j];
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if( j != (int)j && j < incount )
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{
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frac = j - (int)j;
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outdata[i] += (data[(int)j+1] - data[(int)j]) * frac;
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}
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}
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}
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return true;
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}
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if( outwidth == 2 )
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{
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int16_t *outdata = (int16_t *)sound.tempbuffer;
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if( sc->channels == 2 )
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i*2+0] = data[((int)j)*2+0] * 256;
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outdata[i*2+1] = data[((int)j)*2+1] * 256;
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if( j != (int)j && j < incount )
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{
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frac = ( j - (int)j ) * 256;
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outdata[i*2+0] += (data[((int)j+1)*2+0] - data[((int)j)*2+0]) * frac;
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outdata[i*2+1] += (data[((int)j+1)*2+1] - data[((int)j)*2+1]) * frac;
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}
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}
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}
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else
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i] = data[(int)j] * 256;
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if( j != (int)j && j < incount )
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{
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frac = ( j - (int)j ) * 256;
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outdata[i] += (data[(int)j+1] - data[(int)j]) * frac;
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}
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}
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}
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return true;
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}
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}
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if( inwidth == 2 )
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{
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int16_t *data = (int16_t *)sc->buffer;
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if( outwidth == 1 )
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{
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int8_t *outdata = (int8_t *)sound.tempbuffer;
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if( sc->channels == 2 )
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i*2+0] = data[((int)j)*2+0] / 256;
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outdata[i*2+1] = data[((int)j)*2+1] / 256;
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if( j != (int)j && j < incount )
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{
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frac = ( j - (int)j ) / 256;
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outdata[i*2+0] += (data[((int)j+1)*2+0] - data[((int)j)*2+0]) * frac;
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outdata[i*2+1] += (data[((int)j+1)*2+1] - data[((int)j)*2+1]) * frac;
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}
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}
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}
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else
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i] = data[(int)j] / 256;
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if( j != (int)j && j < incount )
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{
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frac = ( j - (int)j ) / 256;
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outdata[i] += (data[(int)j+1] - data[(int)j]) * frac;
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}
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}
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}
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return true;
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}
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if( outwidth == 2 )
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{
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int16_t *outdata = (int16_t *)sound.tempbuffer;
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if( sc->channels == 2 )
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i*2+0] = data[((int)j)*2+0];
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outdata[i*2+1] = data[((int)j)*2+1];
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if( j != (int)j && j < incount )
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{
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frac = j - (int)j;
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outdata[i*2+0] += (data[((int)j+1)*2+0] - data[((int)j)*2+0]) * frac;
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outdata[i*2+1] += (data[((int)j+1)*2+1] - data[((int)j)*2+1]) * frac;
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}
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}
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}
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else
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{
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for( i = 0; i < outcount; i++ )
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{
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j = stepscale * i;
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outdata[i] = data[(int)j];
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if( j != (int)j && j < incount )
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{
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frac = j - (int)j;
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outdata[i] += (data[(int)j+1] - data[(int)j]) * frac;
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}
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}
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}
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return true;
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}
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}
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return false;
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}
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/*
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================
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Sound_ResampleInternal
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@ -139,125 +445,70 @@ We need convert sound to signed even if nothing to resample
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*/
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static qboolean Sound_ResampleInternal( wavdata_t *sc, int inrate, int inwidth, int outrate, int outwidth )
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{
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double stepscale, j;
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int outcount;
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int i;
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const size_t oldsize = sc->size;
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qboolean handled = false;
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double stepscale;
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double t1, t2;
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int outcount;
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if( inrate == outrate && inwidth == outwidth )
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return false;
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t1 = Sys_DoubleTime();
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stepscale = (double)inrate / outrate; // this is usually 0.5, 1, or 2
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outcount = sc->samples / stepscale;
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sc->size = outcount * outwidth * sc->channels;
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sound.tempbuffer = (byte *)Mem_Realloc( host.soundpool, sound.tempbuffer, sc->size );
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sc->samples = outcount;
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if( FBitSet( sc->flags, SOUND_LOOPED ))
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sc->loopStart = sc->loopStart / stepscale;
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if( inrate == outrate )
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#if 0 && XASH_SDL // slow but somewhat accurate
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{
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if( inwidth == 1 && outwidth == 2 ) // S8 to S16
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{
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for( i = 0; i < outcount * sc->channels; i++ )
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((int16_t*)sound.tempbuffer)[i] = ((int8_t *)sc->buffer)[i] * 256;
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handled = true;
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}
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else if( inwidth == 2 && outwidth == 1 ) // S16 to S8
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{
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for( i = 0; i < outcount * sc->channels; i++ )
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((int8_t*)sound.tempbuffer)[i] = ((int16_t *)sc->buffer)[i] / 256;
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handled = true;
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}
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}
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else // resample case
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{
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if( inwidth == 1 )
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{
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int8_t *data = (int8_t *)sc->buffer;
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const SDL_AudioFormat infmt = inwidth == 1 ? AUDIO_S8 : AUDIO_S16;
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const SDL_AudioFormat outfmt = outwidth == 1 ? AUDIO_S8 : AUDIO_S16;
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SDL_AudioCVT cvt;
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if( outwidth == 1 )
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// SDL_AudioCVT does conversion in place, original buffer is used for it
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if( SDL_BuildAudioCVT( &cvt, infmt, sc->channels, inrate, outfmt, sc->channels, outrate ) > 0 && cvt.needed )
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{
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if( sc->channels == 2 )
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{
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for( i = 0, j = 0; i < outcount; i++, j += stepscale )
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{
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((int8_t*)sound.tempbuffer)[i*2+0] = data[((int)j)*2+0];
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((int8_t*)sound.tempbuffer)[i*2+1] = data[((int)j)*2+1];
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}
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}
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else
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{
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for( i = 0, j = 0; i < outcount; i++, j += stepscale )
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((int8_t*)sound.tempbuffer)[i] = data[(int)j];
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}
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handled = true;
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}
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else if( outwidth == 2 )
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{
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if( sc->channels == 2 )
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{
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for( i = 0, j = 0; i < outcount; i++, j += stepscale )
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{
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((int16_t*)sound.tempbuffer)[i*2+0] = data[((int)j)*2+0] * 256;
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((int16_t*)sound.tempbuffer)[i*2+1] = data[((int)j)*2+1] * 256;
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}
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}
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else
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{
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for( i = 0, j = 0; i < outcount; i++, j += stepscale )
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((int16_t*)sound.tempbuffer)[i] = data[(int)j] * 256;
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}
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handled = true;
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}
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}
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else if( inwidth == 2 )
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{
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int16_t *data = (int16_t *)sc->buffer;
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sc->buffer = (byte *)Mem_Realloc( host.soundpool, sc->buffer, oldsize * cvt.len_mult );
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cvt.len = oldsize;
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cvt.buf = sc->buffer;
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if( outwidth == 1 )
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if( !SDL_ConvertAudio( &cvt ))
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{
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if( sc->channels == 2 )
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{
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for( i = 0, j = 0; i < outcount; i++, j += stepscale )
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{
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((int8_t*)sound.tempbuffer)[i*2+0] = data[((int)j)*2+0] / 256;
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((int8_t*)sound.tempbuffer)[i*2+1] = data[((int)j)*2+1] / 256;
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}
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}
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else
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{
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for( i = 0, j = 0; i < outcount; i++, j += stepscale )
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((int8_t*)sound.tempbuffer)[i] = data[(int)j] / 256;
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}
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handled = true;
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}
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else if( outwidth == 2 )
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{
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if( sc->channels == 2 )
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{
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for( i = 0, j = 0; i < outcount; i++, j += stepscale )
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{
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((int16_t*)sound.tempbuffer)[i*2+0] = data[((int)j)*2+0];
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((int16_t*)sound.tempbuffer)[i*2+1] = data[((int)j)*2+1];
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}
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}
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else
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{
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for( i = 0, j = 0; i < outcount; i++, j += stepscale )
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((int16_t*)sound.tempbuffer)[i] = data[(int)j];
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}
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handled = true;
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t2 = Sys_DoubleTime();
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Con_Reportf( "Sound_Resample: from [%d bit %d Hz] to [%d bit %d Hz] (took %.3fs through SDL)\n", inwidth * 8, inrate, outwidth * 8, outrate, t2 - t1 );
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sc->rate = outrate;
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sc->width = outwidth;
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return false; // HACKHACK: return false so Sound_Process won't reallocate buffer
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}
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}
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}
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#endif
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sound.tempbuffer = (byte *)Mem_Realloc( host.soundpool, sound.tempbuffer, sc->size );
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if( inrate == outrate ) // no resampling, just copy data
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handled = Sound_ConvertNoResample( sc, inwidth, outwidth, outcount );
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else if( inrate > outrate ) // fast case, usually downsample but is also ok for upsampling
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handled = Sound_ConvertDownsample( sc, inwidth, outwidth, outcount, stepscale );
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else // upsample case, w/ interpolation
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handled = Sound_ConvertUpsample( sc, inwidth, outwidth, outcount, stepscale );
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t2 = Sys_DoubleTime();
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if( handled )
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Con_Reportf( "Sound_Resample: from [%d bit %d Hz] to [%d bit %d Hz]\n", inwidth * 8, inrate, outwidth * 8, outrate );
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{
|
||||
if( t2 - t1 > 0.01f ) // critical, report to mod developer
|
||||
Con_Printf( S_WARN "Sound_Resample: from [%d bit %d Hz] to [%d bit %d Hz] (took %.3fs)\n", inwidth * 8, inrate, outwidth * 8, outrate, t2 - t1 );
|
||||
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 );
|
||||
Con_Reportf( "Sound_Resample: from [%d bit %d Hz] to [%d bit %d Hz] (took %.3fs)\n", inwidth * 8, inrate, outwidth * 8, outrate, t2 - t1 );
|
||||
}
|
||||
else
|
||||
Con_Printf( 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;
|
||||
|
|
@ -274,23 +525,20 @@ qboolean Sound_Process( wavdata_t **wav, int rate, int width, uint flags )
|
|||
if( !snd || !snd->buffer )
|
||||
return false;
|
||||
|
||||
if(( flags & SOUND_RESAMPLE ) && ( width > 0 || rate > 0 ))
|
||||
if( FBitSet( flags, SOUND_RESAMPLE ) && ( width > 0 || rate > 0 ))
|
||||
{
|
||||
if( Sound_ResampleInternal( snd, snd->rate, snd->width, rate, width ))
|
||||
result = Sound_ResampleInternal( snd, snd->rate, snd->width, rate, width );
|
||||
|
||||
if( result )
|
||||
{
|
||||
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;
|
||||
snd->buffer = Sound_Copy( snd->size ); // unzone buffer (don't touch sound.tempbuffer)
|
||||
}
|
||||
}
|
||||
|
||||
*wav = snd;
|
||||
|
||||
return false;
|
||||
return result;
|
||||
}
|
||||
|
||||
qboolean Sound_SupportedFileFormat( const char *fileext )
|
||||
|
|
|
|||
Loading…
Add table
Reference in a new issue