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authorMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
committerMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
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Add m-esr52 at 52.6.0
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diff --git a/media/libopus/silk/x86/VAD_sse.c b/media/libopus/silk/x86/VAD_sse.c
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+/* Copyright (c) 2014, Cisco Systems, INC
+ Written by XiangMingZhu WeiZhou MinPeng YanWang
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ - Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+
+ - Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in the
+ documentation and/or other materials provided with the distribution.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+ OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <xmmintrin.h>
+#include <emmintrin.h>
+#include <smmintrin.h>
+
+#include "main.h"
+#include "stack_alloc.h"
+
+/* Weighting factors for tilt measure */
+static const opus_int32 tiltWeights[ VAD_N_BANDS ] = { 30000, 6000, -12000, -12000 };
+
+/***************************************/
+/* Get the speech activity level in Q8 */
+/***************************************/
+opus_int silk_VAD_GetSA_Q8_sse4_1( /* O Return value, 0 if success */
+ silk_encoder_state *psEncC, /* I/O Encoder state */
+ const opus_int16 pIn[] /* I PCM input */
+)
+{
+ opus_int SA_Q15, pSNR_dB_Q7, input_tilt;
+ opus_int decimated_framelength1, decimated_framelength2;
+ opus_int decimated_framelength;
+ opus_int dec_subframe_length, dec_subframe_offset, SNR_Q7, i, b, s;
+ opus_int32 sumSquared, smooth_coef_Q16;
+ opus_int16 HPstateTmp;
+ VARDECL( opus_int16, X );
+ opus_int32 Xnrg[ VAD_N_BANDS ];
+ opus_int32 NrgToNoiseRatio_Q8[ VAD_N_BANDS ];
+ opus_int32 speech_nrg, x_tmp;
+ opus_int X_offset[ VAD_N_BANDS ];
+ opus_int ret = 0;
+ silk_VAD_state *psSilk_VAD = &psEncC->sVAD;
+
+ SAVE_STACK;
+
+ /* Safety checks */
+ silk_assert( VAD_N_BANDS == 4 );
+ silk_assert( MAX_FRAME_LENGTH >= psEncC->frame_length );
+ silk_assert( psEncC->frame_length <= 512 );
+ silk_assert( psEncC->frame_length == 8 * silk_RSHIFT( psEncC->frame_length, 3 ) );
+
+ /***********************/
+ /* Filter and Decimate */
+ /***********************/
+ decimated_framelength1 = silk_RSHIFT( psEncC->frame_length, 1 );
+ decimated_framelength2 = silk_RSHIFT( psEncC->frame_length, 2 );
+ decimated_framelength = silk_RSHIFT( psEncC->frame_length, 3 );
+ /* Decimate into 4 bands:
+ 0 L 3L L 3L 5L
+ - -- - -- --
+ 8 8 2 4 4
+
+ [0-1 kHz| temp. |1-2 kHz| 2-4 kHz | 4-8 kHz |
+
+ They're arranged to allow the minimal ( frame_length / 4 ) extra
+ scratch space during the downsampling process */
+ X_offset[ 0 ] = 0;
+ X_offset[ 1 ] = decimated_framelength + decimated_framelength2;
+ X_offset[ 2 ] = X_offset[ 1 ] + decimated_framelength;
+ X_offset[ 3 ] = X_offset[ 2 ] + decimated_framelength2;
+ ALLOC( X, X_offset[ 3 ] + decimated_framelength1, opus_int16 );
+
+ /* 0-8 kHz to 0-4 kHz and 4-8 kHz */
+ silk_ana_filt_bank_1( pIn, &psSilk_VAD->AnaState[ 0 ],
+ X, &X[ X_offset[ 3 ] ], psEncC->frame_length );
+
+ /* 0-4 kHz to 0-2 kHz and 2-4 kHz */
+ silk_ana_filt_bank_1( X, &psSilk_VAD->AnaState1[ 0 ],
+ X, &X[ X_offset[ 2 ] ], decimated_framelength1 );
+
+ /* 0-2 kHz to 0-1 kHz and 1-2 kHz */
+ silk_ana_filt_bank_1( X, &psSilk_VAD->AnaState2[ 0 ],
+ X, &X[ X_offset[ 1 ] ], decimated_framelength2 );
+
+ /*********************************************/
+ /* HP filter on lowest band (differentiator) */
+ /*********************************************/
+ X[ decimated_framelength - 1 ] = silk_RSHIFT( X[ decimated_framelength - 1 ], 1 );
+ HPstateTmp = X[ decimated_framelength - 1 ];
+ for( i = decimated_framelength - 1; i > 0; i-- ) {
+ X[ i - 1 ] = silk_RSHIFT( X[ i - 1 ], 1 );
+ X[ i ] -= X[ i - 1 ];
+ }
+ X[ 0 ] -= psSilk_VAD->HPstate;
+ psSilk_VAD->HPstate = HPstateTmp;
+
+ /*************************************/
+ /* Calculate the energy in each band */
+ /*************************************/
+ for( b = 0; b < VAD_N_BANDS; b++ ) {
+ /* Find the decimated framelength in the non-uniformly divided bands */
+ decimated_framelength = silk_RSHIFT( psEncC->frame_length, silk_min_int( VAD_N_BANDS - b, VAD_N_BANDS - 1 ) );
+
+ /* Split length into subframe lengths */
+ dec_subframe_length = silk_RSHIFT( decimated_framelength, VAD_INTERNAL_SUBFRAMES_LOG2 );
+ dec_subframe_offset = 0;
+
+ /* Compute energy per sub-frame */
+ /* initialize with summed energy of last subframe */
+ Xnrg[ b ] = psSilk_VAD->XnrgSubfr[ b ];
+ for( s = 0; s < VAD_INTERNAL_SUBFRAMES; s++ ) {
+ __m128i xmm_X, xmm_acc;
+ sumSquared = 0;
+
+ xmm_acc = _mm_setzero_si128();
+
+ for( i = 0; i < dec_subframe_length - 7; i += 8 )
+ {
+ xmm_X = _mm_loadu_si128( (__m128i *)&(X[ X_offset[ b ] + i + dec_subframe_offset ] ) );
+ xmm_X = _mm_srai_epi16( xmm_X, 3 );
+ xmm_X = _mm_madd_epi16( xmm_X, xmm_X );
+ xmm_acc = _mm_add_epi32( xmm_acc, xmm_X );
+ }
+
+ xmm_acc = _mm_add_epi32( xmm_acc, _mm_unpackhi_epi64( xmm_acc, xmm_acc ) );
+ xmm_acc = _mm_add_epi32( xmm_acc, _mm_shufflelo_epi16( xmm_acc, 0x0E ) );
+
+ sumSquared += _mm_cvtsi128_si32( xmm_acc );
+
+ for( ; i < dec_subframe_length; i++ ) {
+ /* The energy will be less than dec_subframe_length * ( silk_int16_MIN / 8 ) ^ 2. */
+ /* Therefore we can accumulate with no risk of overflow (unless dec_subframe_length > 128) */
+ x_tmp = silk_RSHIFT(
+ X[ X_offset[ b ] + i + dec_subframe_offset ], 3 );
+ sumSquared = silk_SMLABB( sumSquared, x_tmp, x_tmp );
+
+ /* Safety check */
+ silk_assert( sumSquared >= 0 );
+ }
+
+ /* Add/saturate summed energy of current subframe */
+ if( s < VAD_INTERNAL_SUBFRAMES - 1 ) {
+ Xnrg[ b ] = silk_ADD_POS_SAT32( Xnrg[ b ], sumSquared );
+ } else {
+ /* Look-ahead subframe */
+ Xnrg[ b ] = silk_ADD_POS_SAT32( Xnrg[ b ], silk_RSHIFT( sumSquared, 1 ) );
+ }
+
+ dec_subframe_offset += dec_subframe_length;
+ }
+ psSilk_VAD->XnrgSubfr[ b ] = sumSquared;
+ }
+
+ /********************/
+ /* Noise estimation */
+ /********************/
+ silk_VAD_GetNoiseLevels( &Xnrg[ 0 ], psSilk_VAD );
+
+ /***********************************************/
+ /* Signal-plus-noise to noise ratio estimation */
+ /***********************************************/
+ sumSquared = 0;
+ input_tilt = 0;
+ for( b = 0; b < VAD_N_BANDS; b++ ) {
+ speech_nrg = Xnrg[ b ] - psSilk_VAD->NL[ b ];
+ if( speech_nrg > 0 ) {
+ /* Divide, with sufficient resolution */
+ if( ( Xnrg[ b ] & 0xFF800000 ) == 0 ) {
+ NrgToNoiseRatio_Q8[ b ] = silk_DIV32( silk_LSHIFT( Xnrg[ b ], 8 ), psSilk_VAD->NL[ b ] + 1 );
+ } else {
+ NrgToNoiseRatio_Q8[ b ] = silk_DIV32( Xnrg[ b ], silk_RSHIFT( psSilk_VAD->NL[ b ], 8 ) + 1 );
+ }
+
+ /* Convert to log domain */
+ SNR_Q7 = silk_lin2log( NrgToNoiseRatio_Q8[ b ] ) - 8 * 128;
+
+ /* Sum-of-squares */
+ sumSquared = silk_SMLABB( sumSquared, SNR_Q7, SNR_Q7 ); /* Q14 */
+
+ /* Tilt measure */
+ if( speech_nrg < ( (opus_int32)1 << 20 ) ) {
+ /* Scale down SNR value for small subband speech energies */
+ SNR_Q7 = silk_SMULWB( silk_LSHIFT( silk_SQRT_APPROX( speech_nrg ), 6 ), SNR_Q7 );
+ }
+ input_tilt = silk_SMLAWB( input_tilt, tiltWeights[ b ], SNR_Q7 );
+ } else {
+ NrgToNoiseRatio_Q8[ b ] = 256;
+ }
+ }
+
+ /* Mean-of-squares */
+ sumSquared = silk_DIV32_16( sumSquared, VAD_N_BANDS ); /* Q14 */
+
+ /* Root-mean-square approximation, scale to dBs, and write to output pointer */
+ pSNR_dB_Q7 = (opus_int16)( 3 * silk_SQRT_APPROX( sumSquared ) ); /* Q7 */
+
+ /*********************************/
+ /* Speech Probability Estimation */
+ /*********************************/
+ SA_Q15 = silk_sigm_Q15( silk_SMULWB( VAD_SNR_FACTOR_Q16, pSNR_dB_Q7 ) - VAD_NEGATIVE_OFFSET_Q5 );
+
+ /**************************/
+ /* Frequency Tilt Measure */
+ /**************************/
+ psEncC->input_tilt_Q15 = silk_LSHIFT( silk_sigm_Q15( input_tilt ) - 16384, 1 );
+
+ /**************************************************/
+ /* Scale the sigmoid output based on power levels */
+ /**************************************************/
+ speech_nrg = 0;
+ for( b = 0; b < VAD_N_BANDS; b++ ) {
+ /* Accumulate signal-without-noise energies, higher frequency bands have more weight */
+ speech_nrg += ( b + 1 ) * silk_RSHIFT( Xnrg[ b ] - psSilk_VAD->NL[ b ], 4 );
+ }
+
+ /* Power scaling */
+ if( speech_nrg <= 0 ) {
+ SA_Q15 = silk_RSHIFT( SA_Q15, 1 );
+ } else if( speech_nrg < 32768 ) {
+ if( psEncC->frame_length == 10 * psEncC->fs_kHz ) {
+ speech_nrg = silk_LSHIFT_SAT32( speech_nrg, 16 );
+ } else {
+ speech_nrg = silk_LSHIFT_SAT32( speech_nrg, 15 );
+ }
+
+ /* square-root */
+ speech_nrg = silk_SQRT_APPROX( speech_nrg );
+ SA_Q15 = silk_SMULWB( 32768 + speech_nrg, SA_Q15 );
+ }
+
+ /* Copy the resulting speech activity in Q8 */
+ psEncC->speech_activity_Q8 = silk_min_int( silk_RSHIFT( SA_Q15, 7 ), silk_uint8_MAX );
+
+ /***********************************/
+ /* Energy Level and SNR estimation */
+ /***********************************/
+ /* Smoothing coefficient */
+ smooth_coef_Q16 = silk_SMULWB( VAD_SNR_SMOOTH_COEF_Q18, silk_SMULWB( (opus_int32)SA_Q15, SA_Q15 ) );
+
+ if( psEncC->frame_length == 10 * psEncC->fs_kHz ) {
+ smooth_coef_Q16 >>= 1;
+ }
+
+ for( b = 0; b < VAD_N_BANDS; b++ ) {
+ /* compute smoothed energy-to-noise ratio per band */
+ psSilk_VAD->NrgRatioSmth_Q8[ b ] = silk_SMLAWB( psSilk_VAD->NrgRatioSmth_Q8[ b ],
+ NrgToNoiseRatio_Q8[ b ] - psSilk_VAD->NrgRatioSmth_Q8[ b ], smooth_coef_Q16 );
+
+ /* signal to noise ratio in dB per band */
+ SNR_Q7 = 3 * ( silk_lin2log( psSilk_VAD->NrgRatioSmth_Q8[b] ) - 8 * 128 );
+ /* quality = sigmoid( 0.25 * ( SNR_dB - 16 ) ); */
+ psEncC->input_quality_bands_Q15[ b ] = silk_sigm_Q15( silk_RSHIFT( SNR_Q7 - 16 * 128, 4 ) );
+ }
+
+ RESTORE_STACK;
+ return( ret );
+}