* #30460: added opus dep
diff --git a/jni/libopus/silk/float/noise_shape_analysis_FLP.c b/jni/libopus/silk/float/noise_shape_analysis_FLP.c
new file mode 100644
index 0000000..33bfd20
--- /dev/null
+++ b/jni/libopus/silk/float/noise_shape_analysis_FLP.c
@@ -0,0 +1,365 @@
+/***********************************************************************
+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_FLP.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 inline silk_float warped_gain(
+ const silk_float *coefs,
+ silk_float lambda,
+ opus_int order
+) {
+ opus_int i;
+ silk_float gain;
+
+ lambda = -lambda;
+ gain = coefs[ order - 1 ];
+ for( i = order - 2; i >= 0; i-- ) {
+ gain = lambda * gain + coefs[ i ];
+ }
+ return (silk_float)( 1.0f / ( 1.0f - lambda * gain ) );
+}
+
+/* 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 inline void warped_true2monic_coefs(
+ silk_float *coefs_syn,
+ silk_float *coefs_ana,
+ silk_float lambda,
+ silk_float limit,
+ opus_int order
+) {
+ opus_int i, iter, ind = 0;
+ silk_float tmp, maxabs, chirp, gain_syn, gain_ana;
+
+ /* Convert to monic coefficients */
+ for( i = order - 1; i > 0; i-- ) {
+ coefs_syn[ i - 1 ] -= lambda * coefs_syn[ i ];
+ coefs_ana[ i - 1 ] -= lambda * coefs_ana[ i ];
+ }
+ gain_syn = ( 1.0f - lambda * lambda ) / ( 1.0f + lambda * coefs_syn[ 0 ] );
+ gain_ana = ( 1.0f - lambda * lambda ) / ( 1.0f + lambda * coefs_ana[ 0 ] );
+ for( i = 0; i < order; i++ ) {
+ coefs_syn[ i ] *= gain_syn;
+ coefs_ana[ i ] *= gain_ana;
+ }
+
+ /* Limit */
+ for( iter = 0; iter < 10; iter++ ) {
+ /* Find maximum absolute value */
+ maxabs = -1.0f;
+ for( i = 0; i < order; i++ ) {
+ tmp = silk_max( silk_abs_float( coefs_syn[ i ] ), silk_abs_float( coefs_ana[ i ] ) );
+ if( tmp > maxabs ) {
+ maxabs = tmp;
+ ind = i;
+ }
+ }
+ if( maxabs <= limit ) {
+ /* Coefficients are within range - done */
+ return;
+ }
+
+ /* Convert back to true warped coefficients */
+ for( i = 1; i < order; i++ ) {
+ coefs_syn[ i - 1 ] += lambda * coefs_syn[ i ];
+ coefs_ana[ i - 1 ] += lambda * coefs_ana[ i ];
+ }
+ gain_syn = 1.0f / gain_syn;
+ gain_ana = 1.0f / gain_ana;
+ for( i = 0; i < order; i++ ) {
+ coefs_syn[ i ] *= gain_syn;
+ coefs_ana[ i ] *= gain_ana;
+ }
+
+ /* Apply bandwidth expansion */
+ chirp = 0.99f - ( 0.8f + 0.1f * iter ) * ( maxabs - limit ) / ( maxabs * ( ind + 1 ) );
+ silk_bwexpander_FLP( coefs_syn, order, chirp );
+ silk_bwexpander_FLP( coefs_ana, order, chirp );
+
+ /* Convert to monic warped coefficients */
+ for( i = order - 1; i > 0; i-- ) {
+ coefs_syn[ i - 1 ] -= lambda * coefs_syn[ i ];
+ coefs_ana[ i - 1 ] -= lambda * coefs_ana[ i ];
+ }
+ gain_syn = ( 1.0f - lambda * lambda ) / ( 1.0f + lambda * coefs_syn[ 0 ] );
+ gain_ana = ( 1.0f - lambda * lambda ) / ( 1.0f + lambda * coefs_ana[ 0 ] );
+ for( i = 0; i < order; i++ ) {
+ coefs_syn[ i ] *= gain_syn;
+ coefs_ana[ i ] *= gain_ana;
+ }
+ }
+ silk_assert( 0 );
+}
+
+/* Compute noise shaping coefficients and initial gain values */
+void silk_noise_shape_analysis_FLP(
+ silk_encoder_state_FLP *psEnc, /* I/O Encoder state FLP */
+ silk_encoder_control_FLP *psEncCtrl, /* I/O Encoder control FLP */
+ const silk_float *pitch_res, /* I LPC residual from pitch analysis */
+ const silk_float *x /* I Input signal [frame_length + la_shape] */
+)
+{
+ silk_shape_state_FLP *psShapeSt = &psEnc->sShape;
+ opus_int k, nSamples;
+ silk_float SNR_adj_dB, HarmBoost, HarmShapeGain, Tilt;
+ silk_float nrg, pre_nrg, log_energy, log_energy_prev, energy_variation;
+ silk_float delta, BWExp1, BWExp2, gain_mult, gain_add, strength, b, warping;
+ silk_float x_windowed[ SHAPE_LPC_WIN_MAX ];
+ silk_float auto_corr[ MAX_SHAPE_LPC_ORDER + 1 ];
+ const silk_float *x_ptr, *pitch_res_ptr;
+
+ /* Point to start of first LPC analysis block */
+ x_ptr = x - psEnc->sCmn.la_shape;
+
+ /****************/
+ /* GAIN CONTROL */
+ /****************/
+ SNR_adj_dB = psEnc->sCmn.SNR_dB_Q7 * ( 1 / 128.0f );
+
+ /* Input quality is the average of the quality in the lowest two VAD bands */
+ psEncCtrl->input_quality = 0.5f * ( psEnc->sCmn.input_quality_bands_Q15[ 0 ] + psEnc->sCmn.input_quality_bands_Q15[ 1 ] ) * ( 1.0f / 32768.0f );
+
+ /* Coding quality level, between 0.0 and 1.0 */
+ psEncCtrl->coding_quality = silk_sigmoid( 0.25f * ( SNR_adj_dB - 20.0f ) );
+
+ if( psEnc->sCmn.useCBR == 0 ) {
+ /* Reduce coding SNR during low speech activity */
+ b = 1.0f - psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f );
+ SNR_adj_dB -= BG_SNR_DECR_dB * psEncCtrl->coding_quality * ( 0.5f + 0.5f * psEncCtrl->input_quality ) * b * b;
+ }
+
+ if( psEnc->sCmn.indices.signalType == TYPE_VOICED ) {
+ /* Reduce gains for periodic signals */
+ SNR_adj_dB += HARM_SNR_INCR_dB * psEnc->LTPCorr;
+ } else {
+ /* For unvoiced signals and low-quality input, adjust the quality slower than SNR_dB setting */
+ SNR_adj_dB += ( -0.4f * psEnc->sCmn.SNR_dB_Q7 * ( 1 / 128.0f ) + 6.0f ) * ( 1.0f - psEncCtrl->input_quality );
+ }
+
+ /*************************/
+ /* 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;
+ psEncCtrl->sparseness = 0.0f;
+ } else {
+ /* Sparseness measure, based on relative fluctuations of energy per 2 milliseconds */
+ nSamples = 2 * psEnc->sCmn.fs_kHz;
+ energy_variation = 0.0f;
+ log_energy_prev = 0.0f;
+ pitch_res_ptr = pitch_res;
+ for( k = 0; k < silk_SMULBB( SUB_FRAME_LENGTH_MS, psEnc->sCmn.nb_subfr ) / 2; k++ ) {
+ nrg = ( silk_float )nSamples + ( silk_float )silk_energy_FLP( pitch_res_ptr, nSamples );
+ log_energy = silk_log2( nrg );
+ if( k > 0 ) {
+ energy_variation += silk_abs_float( log_energy - log_energy_prev );
+ }
+ log_energy_prev = log_energy;
+ pitch_res_ptr += nSamples;
+ }
+ psEncCtrl->sparseness = silk_sigmoid( 0.4f * ( energy_variation - 5.0f ) );
+
+ /* Set quantization offset depending on sparseness measure */
+ if( psEncCtrl->sparseness > SPARSENESS_THRESHOLD_QNT_OFFSET ) {
+ psEnc->sCmn.indices.quantOffsetType = 0;
+ } else {
+ psEnc->sCmn.indices.quantOffsetType = 1;
+ }
+
+ /* Increase coding SNR for sparse signals */
+ SNR_adj_dB += SPARSE_SNR_INCR_dB * ( psEncCtrl->sparseness - 0.5f );
+ }
+
+ /*******************************/
+ /* Control bandwidth expansion */
+ /*******************************/
+ /* More BWE for signals with high prediction gain */
+ strength = FIND_PITCH_WHITE_NOISE_FRACTION * psEncCtrl->predGain; /* between 0.0 and 1.0 */
+ BWExp1 = BWExp2 = BANDWIDTH_EXPANSION / ( 1.0f + strength * strength );
+ delta = LOW_RATE_BANDWIDTH_EXPANSION_DELTA * ( 1.0f - 0.75f * psEncCtrl->coding_quality );
+ BWExp1 -= delta;
+ BWExp2 += delta;
+ /* BWExp1 will be applied after BWExp2, so make it relative */
+ BWExp1 /= BWExp2;
+
+ if( psEnc->sCmn.warping_Q16 > 0 ) {
+ /* Slightly more warping in analysis will move quantization noise up in frequency, where it's better masked */
+ warping = (silk_float)psEnc->sCmn.warping_Q16 / 65536.0f + 0.01f * psEncCtrl->coding_quality;
+ } else {
+ warping = 0.0f;
+ }
+
+ /********************************************/
+ /* Compute noise shaping AR coefs and gains */
+ /********************************************/
+ 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 = ( psEnc->sCmn.shapeWinLength - flat_part ) / 2;
+
+ silk_apply_sine_window_FLP( x_windowed, x_ptr, 1, slope_part );
+ shift = slope_part;
+ silk_memcpy( x_windowed + shift, x_ptr + shift, flat_part * sizeof(silk_float) );
+ shift += flat_part;
+ silk_apply_sine_window_FLP( 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_FLP( auto_corr, x_windowed, warping,
+ psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder );
+ } else {
+ /* Calculate regular auto correlation */
+ silk_autocorrelation_FLP( auto_corr, x_windowed, psEnc->sCmn.shapeWinLength, psEnc->sCmn.shapingLPCOrder + 1 );
+ }
+
+ /* Add white noise, as a fraction of energy */
+ auto_corr[ 0 ] += auto_corr[ 0 ] * SHAPE_WHITE_NOISE_FRACTION;
+
+ /* Convert correlations to prediction coefficients, and compute residual energy */
+ nrg = silk_levinsondurbin_FLP( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], auto_corr, psEnc->sCmn.shapingLPCOrder );
+ psEncCtrl->Gains[ k ] = ( silk_float )sqrt( nrg );
+
+ if( psEnc->sCmn.warping_Q16 > 0 ) {
+ /* Adjust gain for warping */
+ psEncCtrl->Gains[ k ] *= warped_gain( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], warping, psEnc->sCmn.shapingLPCOrder );
+ }
+
+ /* Bandwidth expansion for synthesis filter shaping */
+ silk_bwexpander_FLP( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], psEnc->sCmn.shapingLPCOrder, BWExp2 );
+
+ /* Compute noise shaping filter coefficients */
+ silk_memcpy(
+ &psEncCtrl->AR1[ k * MAX_SHAPE_LPC_ORDER ],
+ &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ],
+ psEnc->sCmn.shapingLPCOrder * sizeof( silk_float ) );
+
+ /* Bandwidth expansion for analysis filter shaping */
+ silk_bwexpander_FLP( &psEncCtrl->AR1[ k * MAX_SHAPE_LPC_ORDER ], psEnc->sCmn.shapingLPCOrder, BWExp1 );
+
+ /* Ratio of prediction gains, in energy domain */
+ pre_nrg = silk_LPC_inverse_pred_gain_FLP( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], psEnc->sCmn.shapingLPCOrder );
+ nrg = silk_LPC_inverse_pred_gain_FLP( &psEncCtrl->AR1[ k * MAX_SHAPE_LPC_ORDER ], psEnc->sCmn.shapingLPCOrder );
+ psEncCtrl->GainsPre[ k ] = 1.0f - 0.7f * ( 1.0f - pre_nrg / nrg );
+
+ /* Convert to monic warped prediction coefficients and limit absolute values */
+ warped_true2monic_coefs( &psEncCtrl->AR2[ k * MAX_SHAPE_LPC_ORDER ], &psEncCtrl->AR1[ k * MAX_SHAPE_LPC_ORDER ],
+ warping, 3.999f, psEnc->sCmn.shapingLPCOrder );
+ }
+
+ /*****************/
+ /* Gain tweaking */
+ /*****************/
+ /* Increase gains during low speech activity */
+ gain_mult = (silk_float)pow( 2.0f, -0.16f * SNR_adj_dB );
+ gain_add = (silk_float)pow( 2.0f, 0.16f * MIN_QGAIN_DB );
+ for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
+ psEncCtrl->Gains[ k ] *= gain_mult;
+ psEncCtrl->Gains[ k ] += gain_add;
+ }
+
+ gain_mult = 1.0f + INPUT_TILT + psEncCtrl->coding_quality * HIGH_RATE_INPUT_TILT;
+ for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
+ psEncCtrl->GainsPre[ k ] *= gain_mult;
+ }
+
+ /************************************************/
+ /* Control low-frequency shaping and noise tilt */
+ /************************************************/
+ /* Less low frequency shaping for noisy inputs */
+ strength = LOW_FREQ_SHAPING * ( 1.0f + LOW_QUALITY_LOW_FREQ_SHAPING_DECR * ( psEnc->sCmn.input_quality_bands_Q15[ 0 ] * ( 1.0f / 32768.0f ) - 1.0f ) );
+ strength *= psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f );
+ 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])*/
+ for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
+ b = 0.2f / psEnc->sCmn.fs_kHz + 3.0f / psEncCtrl->pitchL[ k ];
+ psEncCtrl->LF_MA_shp[ k ] = -1.0f + b;
+ psEncCtrl->LF_AR_shp[ k ] = 1.0f - b - b * strength;
+ }
+ Tilt = - HP_NOISE_COEF -
+ (1 - HP_NOISE_COEF) * HARM_HP_NOISE_COEF * psEnc->sCmn.speech_activity_Q8 * ( 1.0f / 256.0f );
+ } else {
+ b = 1.3f / psEnc->sCmn.fs_kHz;
+ psEncCtrl->LF_MA_shp[ 0 ] = -1.0f + b;
+ psEncCtrl->LF_AR_shp[ 0 ] = 1.0f - b - b * strength * 0.6f;
+ for( k = 1; k < psEnc->sCmn.nb_subfr; k++ ) {
+ psEncCtrl->LF_MA_shp[ k ] = psEncCtrl->LF_MA_shp[ 0 ];
+ psEncCtrl->LF_AR_shp[ k ] = psEncCtrl->LF_AR_shp[ 0 ];
+ }
+ Tilt = -HP_NOISE_COEF;
+ }
+
+ /****************************/
+ /* HARMONIC SHAPING CONTROL */
+ /****************************/
+ /* Control boosting of harmonic frequencies */
+ HarmBoost = LOW_RATE_HARMONIC_BOOST * ( 1.0f - psEncCtrl->coding_quality ) * psEnc->LTPCorr;
+
+ /* More harmonic boost for noisy input signals */
+ HarmBoost += LOW_INPUT_QUALITY_HARMONIC_BOOST * ( 1.0f - psEncCtrl->input_quality );
+
+ if( USE_HARM_SHAPING && psEnc->sCmn.indices.signalType == TYPE_VOICED ) {
+ /* Harmonic noise shaping */
+ HarmShapeGain = HARMONIC_SHAPING;
+
+ /* More harmonic noise shaping for high bitrates or noisy input */
+ HarmShapeGain += HIGH_RATE_OR_LOW_QUALITY_HARMONIC_SHAPING *
+ ( 1.0f - ( 1.0f - psEncCtrl->coding_quality ) * psEncCtrl->input_quality );
+
+ /* Less harmonic noise shaping for less periodic signals */
+ HarmShapeGain *= ( silk_float )sqrt( psEnc->LTPCorr );
+ } else {
+ HarmShapeGain = 0.0f;
+ }
+
+ /*************************/
+ /* Smooth over subframes */
+ /*************************/
+ for( k = 0; k < psEnc->sCmn.nb_subfr; k++ ) {
+ psShapeSt->HarmBoost_smth += SUBFR_SMTH_COEF * ( HarmBoost - psShapeSt->HarmBoost_smth );
+ psEncCtrl->HarmBoost[ k ] = psShapeSt->HarmBoost_smth;
+ psShapeSt->HarmShapeGain_smth += SUBFR_SMTH_COEF * ( HarmShapeGain - psShapeSt->HarmShapeGain_smth );
+ psEncCtrl->HarmShapeGain[ k ] = psShapeSt->HarmShapeGain_smth;
+ psShapeSt->Tilt_smth += SUBFR_SMTH_COEF * ( Tilt - psShapeSt->Tilt_smth );
+ psEncCtrl->Tilt[ k ] = psShapeSt->Tilt_smth;
+ }
+}