Nagram/TMessagesProj/jni/opus/silk/resampler_down2.c
DrKLO 4ede311055 Update to 1.4.6
Audio notes (opus codec)
A lot of different improvements and bug fixes

Thanks to:
https://github.com/DrKLO/Telegram/issues/293
https://github.com/DrKLO/Telegram/issues/256

FOSS configuration not ready yet

I will move main dev branch to github in next couple commits
2014-03-23 02:31:55 +04:00

75 lines
3.4 KiB
C

/***********************************************************************
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AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "SigProc_FIX.h"
#include "resampler_rom.h"
/* Downsample by a factor 2 */
void silk_resampler_down2(
opus_int32 *S, /* I/O State vector [ 2 ] */
opus_int16 *out, /* O Output signal [ floor(len/2) ] */
const opus_int16 *in, /* I Input signal [ len ] */
opus_int32 inLen /* I Number of input samples */
)
{
opus_int32 k, len2 = silk_RSHIFT32( inLen, 1 );
opus_int32 in32, out32, Y, X;
silk_assert( silk_resampler_down2_0 > 0 );
silk_assert( silk_resampler_down2_1 < 0 );
/* Internal variables and state are in Q10 format */
for( k = 0; k < len2; k++ ) {
/* Convert to Q10 */
in32 = silk_LSHIFT( (opus_int32)in[ 2 * k ], 10 );
/* All-pass section for even input sample */
Y = silk_SUB32( in32, S[ 0 ] );
X = silk_SMLAWB( Y, Y, silk_resampler_down2_1 );
out32 = silk_ADD32( S[ 0 ], X );
S[ 0 ] = silk_ADD32( in32, X );
/* Convert to Q10 */
in32 = silk_LSHIFT( (opus_int32)in[ 2 * k + 1 ], 10 );
/* All-pass section for odd input sample, and add to output of previous section */
Y = silk_SUB32( in32, S[ 1 ] );
X = silk_SMULWB( Y, silk_resampler_down2_0 );
out32 = silk_ADD32( out32, S[ 1 ] );
out32 = silk_ADD32( out32, X );
S[ 1 ] = silk_ADD32( in32, X );
/* Add, convert back to int16 and store to output */
out[ k ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( out32, 11 ) );
}
}