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diff --git a/media/libopus/silk/fixed/noise_shape_analysis_FIX.c b/media/libopus/silk/fixed/noise_shape_analysis_FIX.c
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+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+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.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+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 "main_FIX.h"
+#include "stack_alloc.h"
+#include "tuning_parameters.h"
+
+/* Compute gain to make warped filter coefficients have a zero mean log frequency response on a */
+/* non-warped frequency scale. (So that it can be implemented with a minimum-phase monic filter.) */
+/* Note: A monic filter is one with the first coefficient equal to 1.0. In Silk we omit the first */
+/* coefficient in an array of coefficients, for monic filters. */
+static OPUS_INLINE opus_int32 warped_gain( /* gain in Q16*/
+ const opus_int32 *coefs_Q24,
+ opus_int lambda_Q16,
+ opus_int order
+) {
+ opus_int i;
+ opus_int32 gain_Q24;
+
+ lambda_Q16 = -lambda_Q16;
+ gain_Q24 = coefs_Q24[ order - 1 ];
+ for( i = order - 2; i >= 0; i-- ) {
+ gain_Q24 = silk_SMLAWB( coefs_Q24[ i ], gain_Q24, lambda_Q16 );
+ }
+ gain_Q24 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 24 ), gain_Q24, -lambda_Q16 );
+ return silk_INVERSE32_varQ( gain_Q24, 40 );
+}
+
+/* Convert warped filter coefficients to monic pseudo-warped coefficients and limit maximum */
+/* amplitude of monic warped coefficients by using bandwidth expansion on the true coefficients */
+static OPUS_INLINE void limit_warped_coefs(
+ opus_int32 *coefs_Q24,
+ opus_int lambda_Q16,
+ opus_int32 limit_Q24,
+ opus_int order
+) {
+ opus_int i, iter, ind = 0;
+ opus_int32 tmp, maxabs_Q24, chirp_Q16, gain_Q16;
+ opus_int32 nom_Q16, den_Q24;
+ opus_int32 limit_Q20, maxabs_Q20;
+
+ /* Convert to monic coefficients */
+ lambda_Q16 = -lambda_Q16;
+ for( i = order - 1; i > 0; i-- ) {
+ coefs_Q24[ i - 1 ] = silk_SMLAWB( coefs_Q24[ i - 1 ], coefs_Q24[ i ], lambda_Q16 );
+ }
+ lambda_Q16 = -lambda_Q16;
+ nom_Q16 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 16 ), -(opus_int32)lambda_Q16, lambda_Q16 );
+ den_Q24 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 24 ), coefs_Q24[ 0 ], lambda_Q16 );
+ gain_Q16 = silk_DIV32_varQ( nom_Q16, den_Q24, 24 );
+ for( i = 0; i < order; i++ ) {
+ coefs_Q24[ i ] = silk_SMULWW( gain_Q16, coefs_Q24[ i ] );
+ }
+ limit_Q20 = silk_RSHIFT(limit_Q24, 4);
+ for( iter = 0; iter < 10; iter++ ) {
+ /* Find maximum absolute value */
+ maxabs_Q24 = -1;
+ for( i = 0; i < order; i++ ) {
+ tmp = silk_abs_int32( coefs_Q24[ i ] );
+ if( tmp > maxabs_Q24 ) {
+ maxabs_Q24 = tmp;
+ ind = i;
+ }
+ }
+ /* Use Q20 to avoid any overflow when multiplying by (ind + 1) later. */
+ maxabs_Q20 = silk_RSHIFT(maxabs_Q24, 4);
+ if( maxabs_Q20 <= limit_Q20 ) {
+ /* Coefficients are within range - done */
+ return;
+ }
+
+ /* Convert back to true warped coefficients */
+ for( i = 1; i < order; i++ ) {
+ coefs_Q24[ i - 1 ] = silk_SMLAWB( coefs_Q24[ i - 1 ], coefs_Q24[ i ], lambda_Q16 );
+ }
+ gain_Q16 = silk_INVERSE32_varQ( gain_Q16, 32 );
+ for( i = 0; i < order; i++ ) {
+ coefs_Q24[ i ] = silk_SMULWW( gain_Q16, coefs_Q24[ i ] );
+ }
+
+ /* Apply bandwidth expansion */
+ chirp_Q16 = SILK_FIX_CONST( 0.99, 16 ) - silk_DIV32_varQ(
+ silk_SMULWB( maxabs_Q20 - limit_Q20, silk_SMLABB( SILK_FIX_CONST( 0.8, 10 ), SILK_FIX_CONST( 0.1, 10 ), iter ) ),
+ silk_MUL( maxabs_Q20, ind + 1 ), 22 );
+ silk_bwexpander_32( coefs_Q24, order, chirp_Q16 );
+
+ /* Convert to monic warped coefficients */
+ lambda_Q16 = -lambda_Q16;
+ for( i = order - 1; i > 0; i-- ) {
+ coefs_Q24[ i - 1 ] = silk_SMLAWB( coefs_Q24[ i - 1 ], coefs_Q24[ i ], lambda_Q16 );
+ }
+ lambda_Q16 = -lambda_Q16;
+ nom_Q16 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 16 ), -(opus_int32)lambda_Q16, lambda_Q16 );
+ den_Q24 = silk_SMLAWB( SILK_FIX_CONST( 1.0, 24 ), coefs_Q24[ 0 ], lambda_Q16 );
+ gain_Q16 = silk_DIV32_varQ( nom_Q16, den_Q24, 24 );
+ for( i = 0; i < order; i++ ) {
+ coefs_Q24[ i ] = silk_SMULWW( gain_Q16, coefs_Q24[ i ] );
+ }
+ }
+ silk_assert( 0 );
+}
+
+/* Disable MIPS version until it's updated. */
+#if 0 && defined(MIPSr1_ASM)
+#include "mips/noise_shape_analysis_FIX_mipsr1.h"
+#endif
+
+/**************************************************************/
+/* Compute noise shaping coefficients and initial gain values */
+/**************************************************************/
+#ifndef OVERRIDE_silk_noise_shape_analysis_FIX
+void silk_noise_shape_analysis_FIX(
+ silk_encoder_state_FIX *psEnc, /* I/O Encoder state FIX */
+ silk_encoder_control_FIX *psEncCtrl, /* I/O Encoder control FIX */
+ const opus_int16 *pitch_res, /* I LPC residual from pitch analysis */
+ const opus_int16 *x, /* I Input signal [ frame_length + la_shape ] */
+ int arch /* I Run-time architecture */
+)
+{
+ silk_shape_state_FIX *psShapeSt = &psEnc->sShape;
+ opus_int k, i, nSamples, nSegs, Qnrg, b_Q14, warping_Q16, scale = 0;
+ opus_int32 SNR_adj_dB_Q7, HarmShapeGain_Q16, Tilt_Q16, tmp32;
+ opus_int32 nrg, log_energy_Q7, log_energy_prev_Q7, energy_variation_Q7;
+ opus_int32 BWExp_Q16, gain_mult_Q16, gain_add_Q16, strength_Q16, b_Q8;
+ opus_int32 auto_corr[ MAX_SHAPE_LPC_ORDER + 1 ];
+ opus_int32 refl_coef_Q16[ MAX_SHAPE_LPC_ORDER ];
+ opus_int32 AR_Q24[ MAX_SHAPE_LPC_ORDER ];
+ VARDECL( opus_int16, x_windowed );
+ const opus_int16 *x_ptr, *pitch_res_ptr;
+ SAVE_STACK;
+
+ /* Point to start of first LPC analysis block */
+ x_ptr = x - psEnc->sCmn.la_shape;
+
+ /****************/
+ /* GAIN CONTROL */
+ /****************/
+ SNR_adj_dB_Q7 = psEnc->sCmn.SNR_dB_Q7;
+
+ /* Input quality is the average of the quality in the lowest two VAD bands */
+ psEncCtrl->input_quality_Q14 = ( opus_int )silk_RSHIFT( (opus_int32)psEnc->sCmn.input_quality_bands_Q15[ 0 ]
+ + psEnc->sCmn.input_quality_bands_Q15[ 1 ], 2 );
+
+ /* Coding quality level, between 0.0_Q0 and 1.0_Q0, but in Q14 */
+ psEncCtrl->coding_quality_Q14 = silk_RSHIFT( silk_sigm_Q15( silk_RSHIFT_ROUND( SNR_adj_dB_Q7 -
+ SILK_FIX_CONST( 20.0, 7 ), 4 ) ), 1 );
+
+ /* Reduce coding SNR during low speech activity */
+ if( psEnc->sCmn.useCBR == 0 ) {
+ b_Q8 = SILK_FIX_CONST( 1.0, 8 ) - psEnc->sCmn.speech_activity_Q8;
+ b_Q8 = silk_SMULWB( silk_LSHIFT( b_Q8, 8 ), b_Q8 );
+ SNR_adj_dB_Q7 = silk_SMLAWB( SNR_adj_dB_Q7,
+ silk_SMULBB( SILK_FIX_CONST( -BG_SNR_DECR_dB, 7 ) >> ( 4 + 1 ), b_Q8 ), /* Q11*/
+ silk_SMULWB( SILK_FIX_CONST( 1.0, 14 ) + psEncCtrl->input_quality_Q14, psEncCtrl->coding_quality_Q14 ) ); /* Q12*/
+ }
+
+ if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) {
+ /* Reduce gains for periodic signals */
+ SNR_adj_dB_Q7 = silk_SMLAWB( SNR_adj_dB_Q7, SILK_FIX_CONST( HARM_SNR_INCR_dB, 8 ), psEnc->LTPCorr_Q15 );
+ } else {
+ /* For unvoiced signals and low-quality input, adjust the quality slower than SNR_dB setting */
+ SNR_adj_dB_Q7 = silk_SMLAWB( SNR_adj_dB_Q7,
+ silk_SMLAWB( SILK_FIX_CONST( 6.0, 9 ), -SILK_FIX_CONST( 0.4, 18 ), psEnc->sCmn.SNR_dB_Q7 ),
+ SILK_FIX_CONST( 1.0, 14 ) - psEncCtrl->input_quality_Q14 );
+ }
+
+ /*************************/
+ /* SPARSENESS PROCESSING */
+ /*************************/
+ /* Set quantizer offset */
+ if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) {
+ /* Initially set to 0; may be overruled in process_gains(..) */
+ psEnc->sCmn.indices.quantOffsetType = 0;
+ } else {
+ /* Sparseness measure, based on relative fluctuations of energy per 2 milliseconds */
+ nSamples = silk_LSHIFT( psEnc->sCmn.fs_kHz, 1 );
+ energy_variation_Q7 = 0;
+ log_energy_prev_Q7 = 0;
+ pitch_res_ptr = pitch_res;
+ nSegs = silk_SMULBB( SUB_FRAME_LENGTH_MS, psEnc->sCmn.nb_subfr ) / 2;
+ for( k = 0; k < nSegs; k++ ) {
+ silk_sum_sqr_shift( &nrg, &scale, pitch_res_ptr, nSamples );
+ nrg += silk_RSHIFT( nSamples, scale ); /* Q(-scale)*/
+
+ log_energy_Q7 = silk_lin2log( nrg );
+ if( k > 0 ) {
+ energy_variation_Q7 += silk_abs( log_energy_Q7 - log_energy_prev_Q7 );
+ }
+ log_energy_prev_Q7 = log_energy_Q7;
+ pitch_res_ptr += nSamples;
+ }
+
+ /* Set quantization offset depending on sparseness measure */
+ if( energy_variation_Q7 > SILK_FIX_CONST( ENERGY_VARIATION_THRESHOLD_QNT_OFFSET, 7 ) * (nSegs-1) ) {
+ psEnc->sCmn.indices.quantOffsetType = 0;
+ } else {
+ psEnc->sCmn.indices.quantOffsetType = 1;
+ }
+ }
+
+ /*******************************/
+ /* Control bandwidth expansion */
+ /*******************************/
+ /* More BWE for signals with high prediction gain */
+ strength_Q16 = silk_SMULWB( psEncCtrl->predGain_Q16, SILK_FIX_CONST( FIND_PITCH_WHITE_NOISE_FRACTION, 16 ) );
+ BWExp_Q16 = silk_DIV32_varQ( SILK_FIX_CONST( BANDWIDTH_EXPANSION, 16 ),
+ silk_SMLAWW( SILK_FIX_CONST( 1.0, 16 ), strength_Q16, strength_Q16 ), 16 );
+
+ if( psEnc->sCmn.warping_Q16 > 0 ) {
+ /* Slightly more warping in analysis will move quantization noise up in frequency, where it's better masked */
+ warping_Q16 = silk_SMLAWB( psEnc->sCmn.warping_Q16, (opus_int32)psEncCtrl->coding_quality_Q14, SILK_FIX_CONST( 0.01, 18 ) );
+ } else {
+ warping_Q16 = 0;
+ }
+
+ /********************************************/
+ /* Compute noise shaping AR coefs and gains */
+ /********************************************/
+ ALLOC( x_windowed, psEnc->sCmn.shapeWinLength, opus_int16 );
+ for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
+ /* Apply window: sine slope followed by flat part followed by cosine slope */
+ opus_int shift, slope_part, flat_part;
+ flat_part = psEnc->sCmn.fs_kHz * 3;
+ slope_part = silk_RSHIFT( psEnc->sCmn.shapeWinLength - flat_part, 1 );
+
+ silk_apply_sine_window( x_windowed, x_ptr, 1, slope_part );
+ shift = slope_part;
+ silk_memcpy( x_windowed + shift, x_ptr + shift, flat_part * sizeof(opus_int16) );
+ shift += flat_part;
+ silk_apply_sine_window( x_windowed + shift, x_ptr + shift, 2, slope_part );
+
+ /* Update pointer: next LPC analysis block */
+ x_ptr += psEnc->sCmn.subfr_length;
+
+ if( psEnc->sCmn.warping_Q16 > 0 ) {
+ /* Calculate warped auto correlation */
+ silk_warped_autocorrelation_FIX( auto_corr, &scale, x_windowed, warping_Q16, psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder, arch );
+ } else {
+ /* Calculate regular auto correlation */
+ silk_autocorr( auto_corr, &scale, x_windowed, psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder + 1, arch );
+ }
+
+ /* Add white noise, as a fraction of energy */
+ auto_corr[0] = silk_ADD32( auto_corr[0], silk_max_32( silk_SMULWB( silk_RSHIFT( auto_corr[ 0 ], 4 ),
+ SILK_FIX_CONST( SHAPE_WHITE_NOISE_FRACTION, 20 ) ), 1 ) );
+
+ /* Calculate the reflection coefficients using schur */
+ nrg = silk_schur64( refl_coef_Q16, auto_corr, psEnc->sCmn.shapingLPCOrder );
+ silk_assert( nrg >= 0 );
+
+ /* Convert reflection coefficients to prediction coefficients */
+ silk_k2a_Q16( AR_Q24, refl_coef_Q16, psEnc->sCmn.shapingLPCOrder );
+
+ Qnrg = -scale; /* range: -12...30*/
+ silk_assert( Qnrg >= -12 );
+ silk_assert( Qnrg <= 30 );
+
+ /* Make sure that Qnrg is an even number */
+ if( Qnrg & 1 ) {
+ Qnrg -= 1;
+ nrg >>= 1;
+ }
+
+ tmp32 = silk_SQRT_APPROX( nrg );
+ Qnrg >>= 1; /* range: -6...15*/
+
+ psEncCtrl->Gains_Q16[ k ] = silk_LSHIFT_SAT32( tmp32, 16 - Qnrg );
+
+ if( psEnc->sCmn.warping_Q16 > 0 ) {
+ /* Adjust gain for warping */
+ gain_mult_Q16 = warped_gain( AR_Q24, warping_Q16, psEnc->sCmn.shapingLPCOrder );
+ silk_assert( psEncCtrl->Gains_Q16[ k ] > 0 );
+ if( psEncCtrl->Gains_Q16[ k ] < SILK_FIX_CONST( 0.25, 16 ) ) {
+ psEncCtrl->Gains_Q16[ k ] = silk_SMULWW( psEncCtrl->Gains_Q16[ k ], gain_mult_Q16 );
+ } else {
+ psEncCtrl->Gains_Q16[ k ] = silk_SMULWW( silk_RSHIFT_ROUND( psEncCtrl->Gains_Q16[ k ], 1 ), gain_mult_Q16 );
+ if ( psEncCtrl->Gains_Q16[ k ] >= ( silk_int32_MAX >> 1 ) ) {
+ psEncCtrl->Gains_Q16[ k ] = silk_int32_MAX;
+ } else {
+ psEncCtrl->Gains_Q16[ k ] = silk_LSHIFT32( psEncCtrl->Gains_Q16[ k ], 1 );
+ }
+ }
+ silk_assert( psEncCtrl->Gains_Q16[ k ] > 0 );
+ }
+
+ /* Bandwidth expansion */
+ silk_bwexpander_32( AR_Q24, psEnc->sCmn.shapingLPCOrder, BWExp_Q16 );
+
+ if( psEnc->sCmn.warping_Q16 > 0 ) {
+ /* Convert to monic warped prediction coefficients and limit absolute values */
+ limit_warped_coefs( AR_Q24, warping_Q16, SILK_FIX_CONST( 3.999, 24 ), psEnc->sCmn.shapingLPCOrder );
+
+ /* Convert from Q24 to Q13 and store in int16 */
+ for( i = 0; i < psEnc->sCmn.shapingLPCOrder; i++ ) {
+ psEncCtrl->AR_Q13[ k * MAX_SHAPE_LPC_ORDER + i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( AR_Q24[ i ], 11 ) );
+ }
+ } else {
+ silk_LPC_fit( &psEncCtrl->AR_Q13[ k * MAX_SHAPE_LPC_ORDER ], AR_Q24, 13, 24, psEnc->sCmn.shapingLPCOrder );
+ }
+ }
+
+ /*****************/
+ /* Gain tweaking */
+ /*****************/
+ /* Increase gains during low speech activity and put lower limit on gains */
+ gain_mult_Q16 = silk_log2lin( -silk_SMLAWB( -SILK_FIX_CONST( 16.0, 7 ), SNR_adj_dB_Q7, SILK_FIX_CONST( 0.16, 16 ) ) );
+ gain_add_Q16 = silk_log2lin( silk_SMLAWB( SILK_FIX_CONST( 16.0, 7 ), SILK_FIX_CONST( MIN_QGAIN_DB, 7 ), SILK_FIX_CONST( 0.16, 16 ) ) );
+ silk_assert( gain_mult_Q16 > 0 );
+ for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
+ psEncCtrl->Gains_Q16[ k ] = silk_SMULWW( psEncCtrl->Gains_Q16[ k ], gain_mult_Q16 );
+ silk_assert( psEncCtrl->Gains_Q16[ k ] >= 0 );
+ psEncCtrl->Gains_Q16[ k ] = silk_ADD_POS_SAT32( psEncCtrl->Gains_Q16[ k ], gain_add_Q16 );
+ }
+
+
+ /************************************************/
+ /* Control low-frequency shaping and noise tilt */
+ /************************************************/
+ /* Less low frequency shaping for noisy inputs */
+ strength_Q16 = silk_MUL( SILK_FIX_CONST( LOW_FREQ_SHAPING, 4 ), silk_SMLAWB( SILK_FIX_CONST( 1.0, 12 ),
+ SILK_FIX_CONST( LOW_QUALITY_LOW_FREQ_SHAPING_DECR, 13 ), psEnc->sCmn.input_quality_bands_Q15[ 0 ] - SILK_FIX_CONST( 1.0, 15 ) ) );
+ strength_Q16 = silk_RSHIFT( silk_MUL( strength_Q16, psEnc->sCmn.speech_activity_Q8 ), 8 );
+ if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) {
+ /* Reduce low frequencies quantization noise for periodic signals, depending on pitch lag */
+ /*f = 400; freqz([1, -0.98 + 2e-4 * f], [1, -0.97 + 7e-4 * f], 2^12, Fs); axis([0, 1000, -10, 1])*/
+ opus_int fs_kHz_inv = silk_DIV32_16( SILK_FIX_CONST( 0.2, 14 ), psEnc->sCmn.fs_kHz );
+ for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
+ b_Q14 = fs_kHz_inv + silk_DIV32_16( SILK_FIX_CONST( 3.0, 14 ), psEncCtrl->pitchL[ k ] );
+ /* Pack two coefficients in one int32 */
+ psEncCtrl->LF_shp_Q14[ k ] = silk_LSHIFT( SILK_FIX_CONST( 1.0, 14 ) - b_Q14 - silk_SMULWB( strength_Q16, b_Q14 ), 16 );
+ psEncCtrl->LF_shp_Q14[ k ] |= (opus_uint16)( b_Q14 - SILK_FIX_CONST( 1.0, 14 ) );
+ }
+ silk_assert( SILK_FIX_CONST( HARM_HP_NOISE_COEF, 24 ) < SILK_FIX_CONST( 0.5, 24 ) ); /* Guarantees that second argument to SMULWB() is within range of an opus_int16*/
+ Tilt_Q16 = - SILK_FIX_CONST( HP_NOISE_COEF, 16 ) -
+ silk_SMULWB( SILK_FIX_CONST( 1.0, 16 ) - SILK_FIX_CONST( HP_NOISE_COEF, 16 ),
+ silk_SMULWB( SILK_FIX_CONST( HARM_HP_NOISE_COEF, 24 ), psEnc->sCmn.speech_activity_Q8 ) );
+ } else {
+ b_Q14 = silk_DIV32_16( 21299, psEnc->sCmn.fs_kHz ); /* 1.3_Q0 = 21299_Q14*/
+ /* Pack two coefficients in one int32 */
+ psEncCtrl->LF_shp_Q14[ 0 ] = silk_LSHIFT( SILK_FIX_CONST( 1.0, 14 ) - b_Q14 -
+ silk_SMULWB( strength_Q16, silk_SMULWB( SILK_FIX_CONST( 0.6, 16 ), b_Q14 ) ), 16 );
+ psEncCtrl->LF_shp_Q14[ 0 ] |= (opus_uint16)( b_Q14 - SILK_FIX_CONST( 1.0, 14 ) );
+ for( k = 1; k < psEnc->sCmn.nb_subfr; k++ ) {
+ psEncCtrl->LF_shp_Q14[ k ] = psEncCtrl->LF_shp_Q14[ 0 ];
+ }
+ Tilt_Q16 = -SILK_FIX_CONST( HP_NOISE_COEF, 16 );
+ }
+
+ /****************************/
+ /* HARMONIC SHAPING CONTROL */
+ /****************************/
+ if( USE_HARM_SHAPING && psEnc->sCmn.indices.signalType == TYPE_VOICED ) {
+ /* More harmonic noise shaping for high bitrates or noisy input */
+ HarmShapeGain_Q16 = silk_SMLAWB( SILK_FIX_CONST( HARMONIC_SHAPING, 16 ),
+ SILK_FIX_CONST( 1.0, 16 ) - silk_SMULWB( SILK_FIX_CONST( 1.0, 18 ) - silk_LSHIFT( psEncCtrl->coding_quality_Q14, 4 ),
+ psEncCtrl->input_quality_Q14 ), SILK_FIX_CONST( HIGH_RATE_OR_LOW_QUALITY_HARMONIC_SHAPING, 16 ) );
+
+ /* Less harmonic noise shaping for less periodic signals */
+ HarmShapeGain_Q16 = silk_SMULWB( silk_LSHIFT( HarmShapeGain_Q16, 1 ),
+ silk_SQRT_APPROX( silk_LSHIFT( psEnc->LTPCorr_Q15, 15 ) ) );
+ } else {
+ HarmShapeGain_Q16 = 0;
+ }
+
+ /*************************/
+ /* Smooth over subframes */
+ /*************************/
+ for( k = 0; k < MAX_NB_SUBFR; k++ ) {
+ psShapeSt->HarmShapeGain_smth_Q16 =
+ silk_SMLAWB( psShapeSt->HarmShapeGain_smth_Q16, HarmShapeGain_Q16 - psShapeSt->HarmShapeGain_smth_Q16, SILK_FIX_CONST( SUBFR_SMTH_COEF, 16 ) );
+ psShapeSt->Tilt_smth_Q16 =
+ silk_SMLAWB( psShapeSt->Tilt_smth_Q16, Tilt_Q16 - psShapeSt->Tilt_smth_Q16, SILK_FIX_CONST( SUBFR_SMTH_COEF, 16 ) );
+
+ psEncCtrl->HarmShapeGain_Q14[ k ] = ( opus_int )silk_RSHIFT_ROUND( psShapeSt->HarmShapeGain_smth_Q16, 2 );
+ psEncCtrl->Tilt_Q14[ k ] = ( opus_int )silk_RSHIFT_ROUND( psShapeSt->Tilt_smth_Q16, 2 );
+ }
+ RESTORE_STACK;
+}
+#endif /* OVERRIDE_silk_noise_shape_analysis_FIX */