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+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <assert.h>
+#include <math.h>
+#include <string.h>
+#include <stdlib.h>
+
+#include "webrtc/common_audio/fft4g.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/modules/audio_processing/ns/include/noise_suppression.h"
+#include "webrtc/modules/audio_processing/ns/ns_core.h"
+#include "webrtc/modules/audio_processing/ns/windows_private.h"
+
+// Set Feature Extraction Parameters.
+static void set_feature_extraction_parameters(NoiseSuppressionC* self) {
+ // Bin size of histogram.
+ self->featureExtractionParams.binSizeLrt = 0.1f;
+ self->featureExtractionParams.binSizeSpecFlat = 0.05f;
+ self->featureExtractionParams.binSizeSpecDiff = 0.1f;
+
+ // Range of histogram over which LRT threshold is computed.
+ self->featureExtractionParams.rangeAvgHistLrt = 1.f;
+
+ // Scale parameters: multiply dominant peaks of the histograms by scale factor
+ // to obtain thresholds for prior model.
+ // For LRT and spectral difference.
+ self->featureExtractionParams.factor1ModelPars = 1.2f;
+ // For spectral_flatness: used when noise is flatter than speech.
+ self->featureExtractionParams.factor2ModelPars = 0.9f;
+
+ // Peak limit for spectral flatness (varies between 0 and 1).
+ self->featureExtractionParams.thresPosSpecFlat = 0.6f;
+
+ // Limit on spacing of two highest peaks in histogram: spacing determined by
+ // bin size.
+ self->featureExtractionParams.limitPeakSpacingSpecFlat =
+ 2 * self->featureExtractionParams.binSizeSpecFlat;
+ self->featureExtractionParams.limitPeakSpacingSpecDiff =
+ 2 * self->featureExtractionParams.binSizeSpecDiff;
+
+ // Limit on relevance of second peak.
+ self->featureExtractionParams.limitPeakWeightsSpecFlat = 0.5f;
+ self->featureExtractionParams.limitPeakWeightsSpecDiff = 0.5f;
+
+ // Fluctuation limit of LRT feature.
+ self->featureExtractionParams.thresFluctLrt = 0.05f;
+
+ // Limit on the max and min values for the feature thresholds.
+ self->featureExtractionParams.maxLrt = 1.f;
+ self->featureExtractionParams.minLrt = 0.2f;
+
+ self->featureExtractionParams.maxSpecFlat = 0.95f;
+ self->featureExtractionParams.minSpecFlat = 0.1f;
+
+ self->featureExtractionParams.maxSpecDiff = 1.f;
+ self->featureExtractionParams.minSpecDiff = 0.16f;
+
+ // Criteria of weight of histogram peak to accept/reject feature.
+ self->featureExtractionParams.thresWeightSpecFlat =
+ (int)(0.3 * (self->modelUpdatePars[1])); // For spectral flatness.
+ self->featureExtractionParams.thresWeightSpecDiff =
+ (int)(0.3 * (self->modelUpdatePars[1])); // For spectral difference.
+}
+
+// Initialize state.
+int WebRtcNs_InitCore(NoiseSuppressionC* self, uint32_t fs) {
+ int i;
+ // Check for valid pointer.
+ if (self == NULL) {
+ return -1;
+ }
+
+ // Initialization of struct.
+ if (fs == 8000 || fs == 16000 || fs == 32000 || fs == 48000) {
+ self->fs = fs;
+ } else {
+ return -1;
+ }
+ self->windShift = 0;
+ // We only support 10ms frames.
+ if (fs == 8000) {
+ self->blockLen = 80;
+ self->anaLen = 128;
+ self->window = kBlocks80w128;
+ } else {
+ self->blockLen = 160;
+ self->anaLen = 256;
+ self->window = kBlocks160w256;
+ }
+ self->magnLen = self->anaLen / 2 + 1; // Number of frequency bins.
+
+ // Initialize FFT work arrays.
+ self->ip[0] = 0; // Setting this triggers initialization.
+ memset(self->dataBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
+ WebRtc_rdft(self->anaLen, 1, self->dataBuf, self->ip, self->wfft);
+
+ memset(self->analyzeBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
+ memset(self->dataBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
+ memset(self->syntBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
+
+ // For HB processing.
+ memset(self->dataBufHB,
+ 0,
+ sizeof(float) * NUM_HIGH_BANDS_MAX * ANAL_BLOCKL_MAX);
+
+ // For quantile noise estimation.
+ memset(self->quantile, 0, sizeof(float) * HALF_ANAL_BLOCKL);
+ for (i = 0; i < SIMULT * HALF_ANAL_BLOCKL; i++) {
+ self->lquantile[i] = 8.f;
+ self->density[i] = 0.3f;
+ }
+
+ for (i = 0; i < SIMULT; i++) {
+ self->counter[i] =
+ (int)floor((float)(END_STARTUP_LONG * (i + 1)) / (float)SIMULT);
+ }
+
+ self->updates = 0;
+
+ // Wiener filter initialization.
+ for (i = 0; i < HALF_ANAL_BLOCKL; i++) {
+ self->smooth[i] = 1.f;
+ }
+
+ // Set the aggressiveness: default.
+ self->aggrMode = 0;
+
+ // Initialize variables for new method.
+ self->priorSpeechProb = 0.5f; // Prior prob for speech/noise.
+ // Previous analyze mag spectrum.
+ memset(self->magnPrevAnalyze, 0, sizeof(float) * HALF_ANAL_BLOCKL);
+ // Previous process mag spectrum.
+ memset(self->magnPrevProcess, 0, sizeof(float) * HALF_ANAL_BLOCKL);
+ // Current noise-spectrum.
+ memset(self->noise, 0, sizeof(float) * HALF_ANAL_BLOCKL);
+ // Previous noise-spectrum.
+ memset(self->noisePrev, 0, sizeof(float) * HALF_ANAL_BLOCKL);
+ // Conservative noise spectrum estimate.
+ memset(self->magnAvgPause, 0, sizeof(float) * HALF_ANAL_BLOCKL);
+ // For estimation of HB in second pass.
+ memset(self->speechProb, 0, sizeof(float) * HALF_ANAL_BLOCKL);
+ // Initial average magnitude spectrum.
+ memset(self->initMagnEst, 0, sizeof(float) * HALF_ANAL_BLOCKL);
+ for (i = 0; i < HALF_ANAL_BLOCKL; i++) {
+ // Smooth LR (same as threshold).
+ self->logLrtTimeAvg[i] = LRT_FEATURE_THR;
+ }
+
+ // Feature quantities.
+ // Spectral flatness (start on threshold).
+ self->featureData[0] = SF_FEATURE_THR;
+ self->featureData[1] = 0.f; // Spectral entropy: not used in this version.
+ self->featureData[2] = 0.f; // Spectral variance: not used in this version.
+ // Average LRT factor (start on threshold).
+ self->featureData[3] = LRT_FEATURE_THR;
+ // Spectral template diff (start on threshold).
+ self->featureData[4] = SF_FEATURE_THR;
+ self->featureData[5] = 0.f; // Normalization for spectral difference.
+ // Window time-average of input magnitude spectrum.
+ self->featureData[6] = 0.f;
+
+ // Histogram quantities: used to estimate/update thresholds for features.
+ memset(self->histLrt, 0, sizeof(int) * HIST_PAR_EST);
+ memset(self->histSpecFlat, 0, sizeof(int) * HIST_PAR_EST);
+ memset(self->histSpecDiff, 0, sizeof(int) * HIST_PAR_EST);
+
+
+ self->blockInd = -1; // Frame counter.
+ // Default threshold for LRT feature.
+ self->priorModelPars[0] = LRT_FEATURE_THR;
+ // Threshold for spectral flatness: determined on-line.
+ self->priorModelPars[1] = 0.5f;
+ // sgn_map par for spectral measure: 1 for flatness measure.
+ self->priorModelPars[2] = 1.f;
+ // Threshold for template-difference feature: determined on-line.
+ self->priorModelPars[3] = 0.5f;
+ // Default weighting parameter for LRT feature.
+ self->priorModelPars[4] = 1.f;
+ // Default weighting parameter for spectral flatness feature.
+ self->priorModelPars[5] = 0.f;
+ // Default weighting parameter for spectral difference feature.
+ self->priorModelPars[6] = 0.f;
+
+ // Update flag for parameters:
+ // 0 no update, 1 = update once, 2 = update every window.
+ self->modelUpdatePars[0] = 2;
+ self->modelUpdatePars[1] = 500; // Window for update.
+ // Counter for update of conservative noise spectrum.
+ self->modelUpdatePars[2] = 0;
+ // Counter if the feature thresholds are updated during the sequence.
+ self->modelUpdatePars[3] = self->modelUpdatePars[1];
+
+ self->signalEnergy = 0.0;
+ self->sumMagn = 0.0;
+ self->whiteNoiseLevel = 0.0;
+ self->pinkNoiseNumerator = 0.0;
+ self->pinkNoiseExp = 0.0;
+
+ set_feature_extraction_parameters(self);
+
+ // Default mode.
+ WebRtcNs_set_policy_core(self, 0);
+
+ self->initFlag = 1;
+ return 0;
+}
+
+// Estimate noise.
+static void NoiseEstimation(NoiseSuppressionC* self,
+ float* magn,
+ float* noise) {
+ size_t i, s, offset;
+ float lmagn[HALF_ANAL_BLOCKL], delta;
+
+ if (self->updates < END_STARTUP_LONG) {
+ self->updates++;
+ }
+
+ for (i = 0; i < self->magnLen; i++) {
+ lmagn[i] = (float)log(magn[i]);
+ }
+
+ // Loop over simultaneous estimates.
+ for (s = 0; s < SIMULT; s++) {
+ offset = s * self->magnLen;
+
+ // newquantest(...)
+ for (i = 0; i < self->magnLen; i++) {
+ // Compute delta.
+ if (self->density[offset + i] > 1.0) {
+ delta = FACTOR * 1.f / self->density[offset + i];
+ } else {
+ delta = FACTOR;
+ }
+
+ // Update log quantile estimate.
+ if (lmagn[i] > self->lquantile[offset + i]) {
+ self->lquantile[offset + i] +=
+ QUANTILE * delta / (float)(self->counter[s] + 1);
+ } else {
+ self->lquantile[offset + i] -=
+ (1.f - QUANTILE) * delta / (float)(self->counter[s] + 1);
+ }
+
+ // Update density estimate.
+ if (fabs(lmagn[i] - self->lquantile[offset + i]) < WIDTH) {
+ self->density[offset + i] =
+ ((float)self->counter[s] * self->density[offset + i] +
+ 1.f / (2.f * WIDTH)) /
+ (float)(self->counter[s] + 1);
+ }
+ } // End loop over magnitude spectrum.
+
+ if (self->counter[s] >= END_STARTUP_LONG) {
+ self->counter[s] = 0;
+ if (self->updates >= END_STARTUP_LONG) {
+ for (i = 0; i < self->magnLen; i++) {
+ self->quantile[i] = (float)exp(self->lquantile[offset + i]);
+ }
+ }
+ }
+
+ self->counter[s]++;
+ } // End loop over simultaneous estimates.
+
+ // Sequentially update the noise during startup.
+ if (self->updates < END_STARTUP_LONG) {
+ // Use the last "s" to get noise during startup that differ from zero.
+ for (i = 0; i < self->magnLen; i++) {
+ self->quantile[i] = (float)exp(self->lquantile[offset + i]);
+ }
+ }
+
+ for (i = 0; i < self->magnLen; i++) {
+ noise[i] = self->quantile[i];
+ }
+}
+
+// Extract thresholds for feature parameters.
+// Histograms are computed over some window size (given by
+// self->modelUpdatePars[1]).
+// Thresholds and weights are extracted every window.
+// |flag| = 0 updates histogram only, |flag| = 1 computes the threshold/weights.
+// Threshold and weights are returned in: self->priorModelPars.
+static void FeatureParameterExtraction(NoiseSuppressionC* self, int flag) {
+ int i, useFeatureSpecFlat, useFeatureSpecDiff, numHistLrt;
+ int maxPeak1, maxPeak2;
+ int weightPeak1SpecFlat, weightPeak2SpecFlat, weightPeak1SpecDiff,
+ weightPeak2SpecDiff;
+
+ float binMid, featureSum;
+ float posPeak1SpecFlat, posPeak2SpecFlat, posPeak1SpecDiff, posPeak2SpecDiff;
+ float fluctLrt, avgHistLrt, avgSquareHistLrt, avgHistLrtCompl;
+
+ // 3 features: LRT, flatness, difference.
+ // lrt_feature = self->featureData[3];
+ // flat_feature = self->featureData[0];
+ // diff_feature = self->featureData[4];
+
+ // Update histograms.
+ if (flag == 0) {
+ // LRT
+ if ((self->featureData[3] <
+ HIST_PAR_EST * self->featureExtractionParams.binSizeLrt) &&
+ (self->featureData[3] >= 0.0)) {
+ i = (int)(self->featureData[3] /
+ self->featureExtractionParams.binSizeLrt);
+ self->histLrt[i]++;
+ }
+ // Spectral flatness.
+ if ((self->featureData[0] <
+ HIST_PAR_EST * self->featureExtractionParams.binSizeSpecFlat) &&
+ (self->featureData[0] >= 0.0)) {
+ i = (int)(self->featureData[0] /
+ self->featureExtractionParams.binSizeSpecFlat);
+ self->histSpecFlat[i]++;
+ }
+ // Spectral difference.
+ if ((self->featureData[4] <
+ HIST_PAR_EST * self->featureExtractionParams.binSizeSpecDiff) &&
+ (self->featureData[4] >= 0.0)) {
+ i = (int)(self->featureData[4] /
+ self->featureExtractionParams.binSizeSpecDiff);
+ self->histSpecDiff[i]++;
+ }
+ }
+
+ // Extract parameters for speech/noise probability.
+ if (flag == 1) {
+ // LRT feature: compute the average over
+ // self->featureExtractionParams.rangeAvgHistLrt.
+ avgHistLrt = 0.0;
+ avgHistLrtCompl = 0.0;
+ avgSquareHistLrt = 0.0;
+ numHistLrt = 0;
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ binMid = ((float)i + 0.5f) * self->featureExtractionParams.binSizeLrt;
+ if (binMid <= self->featureExtractionParams.rangeAvgHistLrt) {
+ avgHistLrt += self->histLrt[i] * binMid;
+ numHistLrt += self->histLrt[i];
+ }
+ avgSquareHistLrt += self->histLrt[i] * binMid * binMid;
+ avgHistLrtCompl += self->histLrt[i] * binMid;
+ }
+ if (numHistLrt > 0) {
+ avgHistLrt = avgHistLrt / ((float)numHistLrt);
+ }
+ avgHistLrtCompl = avgHistLrtCompl / ((float)self->modelUpdatePars[1]);
+ avgSquareHistLrt = avgSquareHistLrt / ((float)self->modelUpdatePars[1]);
+ fluctLrt = avgSquareHistLrt - avgHistLrt * avgHistLrtCompl;
+ // Get threshold for LRT feature.
+ if (fluctLrt < self->featureExtractionParams.thresFluctLrt) {
+ // Very low fluctuation, so likely noise.
+ self->priorModelPars[0] = self->featureExtractionParams.maxLrt;
+ } else {
+ self->priorModelPars[0] =
+ self->featureExtractionParams.factor1ModelPars * avgHistLrt;
+ // Check if value is within min/max range.
+ if (self->priorModelPars[0] < self->featureExtractionParams.minLrt) {
+ self->priorModelPars[0] = self->featureExtractionParams.minLrt;
+ }
+ if (self->priorModelPars[0] > self->featureExtractionParams.maxLrt) {
+ self->priorModelPars[0] = self->featureExtractionParams.maxLrt;
+ }
+ }
+ // Done with LRT feature.
+
+ // For spectral flatness and spectral difference: compute the main peaks of
+ // histogram.
+ maxPeak1 = 0;
+ maxPeak2 = 0;
+ posPeak1SpecFlat = 0.0;
+ posPeak2SpecFlat = 0.0;
+ weightPeak1SpecFlat = 0;
+ weightPeak2SpecFlat = 0;
+
+ // Peaks for flatness.
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ binMid =
+ (i + 0.5f) * self->featureExtractionParams.binSizeSpecFlat;
+ if (self->histSpecFlat[i] > maxPeak1) {
+ // Found new "first" peak.
+ maxPeak2 = maxPeak1;
+ weightPeak2SpecFlat = weightPeak1SpecFlat;
+ posPeak2SpecFlat = posPeak1SpecFlat;
+
+ maxPeak1 = self->histSpecFlat[i];
+ weightPeak1SpecFlat = self->histSpecFlat[i];
+ posPeak1SpecFlat = binMid;
+ } else if (self->histSpecFlat[i] > maxPeak2) {
+ // Found new "second" peak.
+ maxPeak2 = self->histSpecFlat[i];
+ weightPeak2SpecFlat = self->histSpecFlat[i];
+ posPeak2SpecFlat = binMid;
+ }
+ }
+
+ // Compute two peaks for spectral difference.
+ maxPeak1 = 0;
+ maxPeak2 = 0;
+ posPeak1SpecDiff = 0.0;
+ posPeak2SpecDiff = 0.0;
+ weightPeak1SpecDiff = 0;
+ weightPeak2SpecDiff = 0;
+ // Peaks for spectral difference.
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ binMid =
+ ((float)i + 0.5f) * self->featureExtractionParams.binSizeSpecDiff;
+ if (self->histSpecDiff[i] > maxPeak1) {
+ // Found new "first" peak.
+ maxPeak2 = maxPeak1;
+ weightPeak2SpecDiff = weightPeak1SpecDiff;
+ posPeak2SpecDiff = posPeak1SpecDiff;
+
+ maxPeak1 = self->histSpecDiff[i];
+ weightPeak1SpecDiff = self->histSpecDiff[i];
+ posPeak1SpecDiff = binMid;
+ } else if (self->histSpecDiff[i] > maxPeak2) {
+ // Found new "second" peak.
+ maxPeak2 = self->histSpecDiff[i];
+ weightPeak2SpecDiff = self->histSpecDiff[i];
+ posPeak2SpecDiff = binMid;
+ }
+ }
+
+ // For spectrum flatness feature.
+ useFeatureSpecFlat = 1;
+ // Merge the two peaks if they are close.
+ if ((fabs(posPeak2SpecFlat - posPeak1SpecFlat) <
+ self->featureExtractionParams.limitPeakSpacingSpecFlat) &&
+ (weightPeak2SpecFlat >
+ self->featureExtractionParams.limitPeakWeightsSpecFlat *
+ weightPeak1SpecFlat)) {
+ weightPeak1SpecFlat += weightPeak2SpecFlat;
+ posPeak1SpecFlat = 0.5f * (posPeak1SpecFlat + posPeak2SpecFlat);
+ }
+ // Reject if weight of peaks is not large enough, or peak value too small.
+ if (weightPeak1SpecFlat <
+ self->featureExtractionParams.thresWeightSpecFlat ||
+ posPeak1SpecFlat < self->featureExtractionParams.thresPosSpecFlat) {
+ useFeatureSpecFlat = 0;
+ }
+ // If selected, get the threshold.
+ if (useFeatureSpecFlat == 1) {
+ // Compute the threshold.
+ self->priorModelPars[1] =
+ self->featureExtractionParams.factor2ModelPars * posPeak1SpecFlat;
+ // Check if value is within min/max range.
+ if (self->priorModelPars[1] < self->featureExtractionParams.minSpecFlat) {
+ self->priorModelPars[1] = self->featureExtractionParams.minSpecFlat;
+ }
+ if (self->priorModelPars[1] > self->featureExtractionParams.maxSpecFlat) {
+ self->priorModelPars[1] = self->featureExtractionParams.maxSpecFlat;
+ }
+ }
+ // Done with flatness feature.
+
+ // For template feature.
+ useFeatureSpecDiff = 1;
+ // Merge the two peaks if they are close.
+ if ((fabs(posPeak2SpecDiff - posPeak1SpecDiff) <
+ self->featureExtractionParams.limitPeakSpacingSpecDiff) &&
+ (weightPeak2SpecDiff >
+ self->featureExtractionParams.limitPeakWeightsSpecDiff *
+ weightPeak1SpecDiff)) {
+ weightPeak1SpecDiff += weightPeak2SpecDiff;
+ posPeak1SpecDiff = 0.5f * (posPeak1SpecDiff + posPeak2SpecDiff);
+ }
+ // Get the threshold value.
+ self->priorModelPars[3] =
+ self->featureExtractionParams.factor1ModelPars * posPeak1SpecDiff;
+ // Reject if weight of peaks is not large enough.
+ if (weightPeak1SpecDiff <
+ self->featureExtractionParams.thresWeightSpecDiff) {
+ useFeatureSpecDiff = 0;
+ }
+ // Check if value is within min/max range.
+ if (self->priorModelPars[3] < self->featureExtractionParams.minSpecDiff) {
+ self->priorModelPars[3] = self->featureExtractionParams.minSpecDiff;
+ }
+ if (self->priorModelPars[3] > self->featureExtractionParams.maxSpecDiff) {
+ self->priorModelPars[3] = self->featureExtractionParams.maxSpecDiff;
+ }
+ // Done with spectral difference feature.
+
+ // Don't use template feature if fluctuation of LRT feature is very low:
+ // most likely just noise state.
+ if (fluctLrt < self->featureExtractionParams.thresFluctLrt) {
+ useFeatureSpecDiff = 0;
+ }
+
+ // Select the weights between the features.
+ // self->priorModelPars[4] is weight for LRT: always selected.
+ // self->priorModelPars[5] is weight for spectral flatness.
+ // self->priorModelPars[6] is weight for spectral difference.
+ featureSum = (float)(1 + useFeatureSpecFlat + useFeatureSpecDiff);
+ self->priorModelPars[4] = 1.f / featureSum;
+ self->priorModelPars[5] = ((float)useFeatureSpecFlat) / featureSum;
+ self->priorModelPars[6] = ((float)useFeatureSpecDiff) / featureSum;
+
+ // Set hists to zero for next update.
+ if (self->modelUpdatePars[0] >= 1) {
+ for (i = 0; i < HIST_PAR_EST; i++) {
+ self->histLrt[i] = 0;
+ self->histSpecFlat[i] = 0;
+ self->histSpecDiff[i] = 0;
+ }
+ }
+ } // End of flag == 1.
+}
+
+// Compute spectral flatness on input spectrum.
+// |magnIn| is the magnitude spectrum.
+// Spectral flatness is returned in self->featureData[0].
+static void ComputeSpectralFlatness(NoiseSuppressionC* self,
+ const float* magnIn) {
+ size_t i;
+ size_t shiftLP = 1; // Option to remove first bin(s) from spectral measures.
+ float avgSpectralFlatnessNum, avgSpectralFlatnessDen, spectralTmp;
+
+ // Compute spectral measures.
+ // For flatness.
+ avgSpectralFlatnessNum = 0.0;
+ avgSpectralFlatnessDen = self->sumMagn;
+ for (i = 0; i < shiftLP; i++) {
+ avgSpectralFlatnessDen -= magnIn[i];
+ }
+ // Compute log of ratio of the geometric to arithmetic mean: check for log(0)
+ // case.
+ for (i = shiftLP; i < self->magnLen; i++) {
+ if (magnIn[i] > 0.0) {
+ avgSpectralFlatnessNum += (float)log(magnIn[i]);
+ } else {
+ self->featureData[0] -= SPECT_FL_TAVG * self->featureData[0];
+ return;
+ }
+ }
+ // Normalize.
+ avgSpectralFlatnessDen = avgSpectralFlatnessDen / self->magnLen;
+ avgSpectralFlatnessNum = avgSpectralFlatnessNum / self->magnLen;
+
+ // Ratio and inverse log: check for case of log(0).
+ spectralTmp = (float)exp(avgSpectralFlatnessNum) / avgSpectralFlatnessDen;
+
+ // Time-avg update of spectral flatness feature.
+ self->featureData[0] += SPECT_FL_TAVG * (spectralTmp - self->featureData[0]);
+ // Done with flatness feature.
+}
+
+// Compute prior and post SNR based on quantile noise estimation.
+// Compute DD estimate of prior SNR.
+// Inputs:
+// * |magn| is the signal magnitude spectrum estimate.
+// * |noise| is the magnitude noise spectrum estimate.
+// Outputs:
+// * |snrLocPrior| is the computed prior SNR.
+// * |snrLocPost| is the computed post SNR.
+static void ComputeSnr(const NoiseSuppressionC* self,
+ const float* magn,
+ const float* noise,
+ float* snrLocPrior,
+ float* snrLocPost) {
+ size_t i;
+
+ for (i = 0; i < self->magnLen; i++) {
+ // Previous post SNR.
+ // Previous estimate: based on previous frame with gain filter.
+ float previousEstimateStsa = self->magnPrevAnalyze[i] /
+ (self->noisePrev[i] + 0.0001f) * self->smooth[i];
+ // Post SNR.
+ snrLocPost[i] = 0.f;
+ if (magn[i] > noise[i]) {
+ snrLocPost[i] = magn[i] / (noise[i] + 0.0001f) - 1.f;
+ }
+ // DD estimate is sum of two terms: current estimate and previous estimate.
+ // Directed decision update of snrPrior.
+ snrLocPrior[i] =
+ DD_PR_SNR * previousEstimateStsa + (1.f - DD_PR_SNR) * snrLocPost[i];
+ } // End of loop over frequencies.
+}
+
+// Compute the difference measure between input spectrum and a template/learned
+// noise spectrum.
+// |magnIn| is the input spectrum.
+// The reference/template spectrum is self->magnAvgPause[i].
+// Returns (normalized) spectral difference in self->featureData[4].
+static void ComputeSpectralDifference(NoiseSuppressionC* self,
+ const float* magnIn) {
+ // avgDiffNormMagn = var(magnIn) - cov(magnIn, magnAvgPause)^2 /
+ // var(magnAvgPause)
+ size_t i;
+ float avgPause, avgMagn, covMagnPause, varPause, varMagn, avgDiffNormMagn;
+
+ avgPause = 0.0;
+ avgMagn = self->sumMagn;
+ // Compute average quantities.
+ for (i = 0; i < self->magnLen; i++) {
+ // Conservative smooth noise spectrum from pause frames.
+ avgPause += self->magnAvgPause[i];
+ }
+ avgPause /= self->magnLen;
+ avgMagn /= self->magnLen;
+
+ covMagnPause = 0.0;
+ varPause = 0.0;
+ varMagn = 0.0;
+ // Compute variance and covariance quantities.
+ for (i = 0; i < self->magnLen; i++) {
+ covMagnPause += (magnIn[i] - avgMagn) * (self->magnAvgPause[i] - avgPause);
+ varPause +=
+ (self->magnAvgPause[i] - avgPause) * (self->magnAvgPause[i] - avgPause);
+ varMagn += (magnIn[i] - avgMagn) * (magnIn[i] - avgMagn);
+ }
+ covMagnPause /= self->magnLen;
+ varPause /= self->magnLen;
+ varMagn /= self->magnLen;
+ // Update of average magnitude spectrum.
+ self->featureData[6] += self->signalEnergy;
+
+ avgDiffNormMagn =
+ varMagn - (covMagnPause * covMagnPause) / (varPause + 0.0001f);
+ // Normalize and compute time-avg update of difference feature.
+ avgDiffNormMagn = (float)(avgDiffNormMagn / (self->featureData[5] + 0.0001f));
+ self->featureData[4] +=
+ SPECT_DIFF_TAVG * (avgDiffNormMagn - self->featureData[4]);
+}
+
+// Compute speech/noise probability.
+// Speech/noise probability is returned in |probSpeechFinal|.
+// |magn| is the input magnitude spectrum.
+// |noise| is the noise spectrum.
+// |snrLocPrior| is the prior SNR for each frequency.
+// |snrLocPost| is the post SNR for each frequency.
+static void SpeechNoiseProb(NoiseSuppressionC* self,
+ float* probSpeechFinal,
+ const float* snrLocPrior,
+ const float* snrLocPost) {
+ size_t i;
+ int sgnMap;
+ float invLrt, gainPrior, indPrior;
+ float logLrtTimeAvgKsum, besselTmp;
+ float indicator0, indicator1, indicator2;
+ float tmpFloat1, tmpFloat2;
+ float weightIndPrior0, weightIndPrior1, weightIndPrior2;
+ float threshPrior0, threshPrior1, threshPrior2;
+ float widthPrior, widthPrior0, widthPrior1, widthPrior2;
+
+ widthPrior0 = WIDTH_PR_MAP;
+ // Width for pause region: lower range, so increase width in tanh map.
+ widthPrior1 = 2.f * WIDTH_PR_MAP;
+ widthPrior2 = 2.f * WIDTH_PR_MAP; // For spectral-difference measure.
+
+ // Threshold parameters for features.
+ threshPrior0 = self->priorModelPars[0];
+ threshPrior1 = self->priorModelPars[1];
+ threshPrior2 = self->priorModelPars[3];
+
+ // Sign for flatness feature.
+ sgnMap = (int)(self->priorModelPars[2]);
+
+ // Weight parameters for features.
+ weightIndPrior0 = self->priorModelPars[4];
+ weightIndPrior1 = self->priorModelPars[5];
+ weightIndPrior2 = self->priorModelPars[6];
+
+ // Compute feature based on average LR factor.
+ // This is the average over all frequencies of the smooth log LRT.
+ logLrtTimeAvgKsum = 0.0;
+ for (i = 0; i < self->magnLen; i++) {
+ tmpFloat1 = 1.f + 2.f * snrLocPrior[i];
+ tmpFloat2 = 2.f * snrLocPrior[i] / (tmpFloat1 + 0.0001f);
+ besselTmp = (snrLocPost[i] + 1.f) * tmpFloat2;
+ self->logLrtTimeAvg[i] +=
+ LRT_TAVG * (besselTmp - (float)log(tmpFloat1) - self->logLrtTimeAvg[i]);
+ logLrtTimeAvgKsum += self->logLrtTimeAvg[i];
+ }
+ logLrtTimeAvgKsum = (float)logLrtTimeAvgKsum / (self->magnLen);
+ self->featureData[3] = logLrtTimeAvgKsum;
+ // Done with computation of LR factor.
+
+ // Compute the indicator functions.
+ // Average LRT feature.
+ widthPrior = widthPrior0;
+ // Use larger width in tanh map for pause regions.
+ if (logLrtTimeAvgKsum < threshPrior0) {
+ widthPrior = widthPrior1;
+ }
+ // Compute indicator function: sigmoid map.
+ indicator0 =
+ 0.5f *
+ ((float)tanh(widthPrior * (logLrtTimeAvgKsum - threshPrior0)) + 1.f);
+
+ // Spectral flatness feature.
+ tmpFloat1 = self->featureData[0];
+ widthPrior = widthPrior0;
+ // Use larger width in tanh map for pause regions.
+ if (sgnMap == 1 && (tmpFloat1 > threshPrior1)) {
+ widthPrior = widthPrior1;
+ }
+ if (sgnMap == -1 && (tmpFloat1 < threshPrior1)) {
+ widthPrior = widthPrior1;
+ }
+ // Compute indicator function: sigmoid map.
+ indicator1 =
+ 0.5f *
+ ((float)tanh((float)sgnMap * widthPrior * (threshPrior1 - tmpFloat1)) +
+ 1.f);
+
+ // For template spectrum-difference.
+ tmpFloat1 = self->featureData[4];
+ widthPrior = widthPrior0;
+ // Use larger width in tanh map for pause regions.
+ if (tmpFloat1 < threshPrior2) {
+ widthPrior = widthPrior2;
+ }
+ // Compute indicator function: sigmoid map.
+ indicator2 =
+ 0.5f * ((float)tanh(widthPrior * (tmpFloat1 - threshPrior2)) + 1.f);
+
+ // Combine the indicator function with the feature weights.
+ indPrior = weightIndPrior0 * indicator0 + weightIndPrior1 * indicator1 +
+ weightIndPrior2 * indicator2;
+ // Done with computing indicator function.
+
+ // Compute the prior probability.
+ self->priorSpeechProb += PRIOR_UPDATE * (indPrior - self->priorSpeechProb);
+ // Make sure probabilities are within range: keep floor to 0.01.
+ if (self->priorSpeechProb > 1.f) {
+ self->priorSpeechProb = 1.f;
+ }
+ if (self->priorSpeechProb < 0.01f) {
+ self->priorSpeechProb = 0.01f;
+ }
+
+ // Final speech probability: combine prior model with LR factor:.
+ gainPrior = (1.f - self->priorSpeechProb) / (self->priorSpeechProb + 0.0001f);
+ for (i = 0; i < self->magnLen; i++) {
+ invLrt = (float)exp(-self->logLrtTimeAvg[i]);
+ invLrt = (float)gainPrior * invLrt;
+ probSpeechFinal[i] = 1.f / (1.f + invLrt);
+ }
+}
+
+// Update the noise features.
+// Inputs:
+// * |magn| is the signal magnitude spectrum estimate.
+// * |updateParsFlag| is an update flag for parameters.
+static void FeatureUpdate(NoiseSuppressionC* self,
+ const float* magn,
+ int updateParsFlag) {
+ // Compute spectral flatness on input spectrum.
+ ComputeSpectralFlatness(self, magn);
+ // Compute difference of input spectrum with learned/estimated noise spectrum.
+ ComputeSpectralDifference(self, magn);
+ // Compute histograms for parameter decisions (thresholds and weights for
+ // features).
+ // Parameters are extracted once every window time.
+ // (=self->modelUpdatePars[1])
+ if (updateParsFlag >= 1) {
+ // Counter update.
+ self->modelUpdatePars[3]--;
+ // Update histogram.
+ if (self->modelUpdatePars[3] > 0) {
+ FeatureParameterExtraction(self, 0);
+ }
+ // Compute model parameters.
+ if (self->modelUpdatePars[3] == 0) {
+ FeatureParameterExtraction(self, 1);
+ self->modelUpdatePars[3] = self->modelUpdatePars[1];
+ // If wish to update only once, set flag to zero.
+ if (updateParsFlag == 1) {
+ self->modelUpdatePars[0] = 0;
+ } else {
+ // Update every window:
+ // Get normalization for spectral difference for next window estimate.
+ self->featureData[6] =
+ self->featureData[6] / ((float)self->modelUpdatePars[1]);
+ self->featureData[5] =
+ 0.5f * (self->featureData[6] + self->featureData[5]);
+ self->featureData[6] = 0.f;
+ }
+ }
+ }
+}
+
+// Update the noise estimate.
+// Inputs:
+// * |magn| is the signal magnitude spectrum estimate.
+// * |snrLocPrior| is the prior SNR.
+// * |snrLocPost| is the post SNR.
+// Output:
+// * |noise| is the updated noise magnitude spectrum estimate.
+static void UpdateNoiseEstimate(NoiseSuppressionC* self,
+ const float* magn,
+ const float* snrLocPrior,
+ const float* snrLocPost,
+ float* noise) {
+ size_t i;
+ float probSpeech, probNonSpeech;
+ // Time-avg parameter for noise update.
+ float gammaNoiseTmp = NOISE_UPDATE;
+ float gammaNoiseOld;
+ float noiseUpdateTmp;
+
+ for (i = 0; i < self->magnLen; i++) {
+ probSpeech = self->speechProb[i];
+ probNonSpeech = 1.f - probSpeech;
+ // Temporary noise update:
+ // Use it for speech frames if update value is less than previous.
+ noiseUpdateTmp = gammaNoiseTmp * self->noisePrev[i] +
+ (1.f - gammaNoiseTmp) * (probNonSpeech * magn[i] +
+ probSpeech * self->noisePrev[i]);
+ // Time-constant based on speech/noise state.
+ gammaNoiseOld = gammaNoiseTmp;
+ gammaNoiseTmp = NOISE_UPDATE;
+ // Increase gamma (i.e., less noise update) for frame likely to be speech.
+ if (probSpeech > PROB_RANGE) {
+ gammaNoiseTmp = SPEECH_UPDATE;
+ }
+ // Conservative noise update.
+ if (probSpeech < PROB_RANGE) {
+ self->magnAvgPause[i] += GAMMA_PAUSE * (magn[i] - self->magnAvgPause[i]);
+ }
+ // Noise update.
+ if (gammaNoiseTmp == gammaNoiseOld) {
+ noise[i] = noiseUpdateTmp;
+ } else {
+ noise[i] = gammaNoiseTmp * self->noisePrev[i] +
+ (1.f - gammaNoiseTmp) * (probNonSpeech * magn[i] +
+ probSpeech * self->noisePrev[i]);
+ // Allow for noise update downwards:
+ // If noise update decreases the noise, it is safe, so allow it to
+ // happen.
+ if (noiseUpdateTmp < noise[i]) {
+ noise[i] = noiseUpdateTmp;
+ }
+ }
+ } // End of freq loop.
+}
+
+// Updates |buffer| with a new |frame|.
+// Inputs:
+// * |frame| is a new speech frame or NULL for setting to zero.
+// * |frame_length| is the length of the new frame.
+// * |buffer_length| is the length of the buffer.
+// Output:
+// * |buffer| is the updated buffer.
+static void UpdateBuffer(const float* frame,
+ size_t frame_length,
+ size_t buffer_length,
+ float* buffer) {
+ assert(buffer_length < 2 * frame_length);
+
+ memcpy(buffer,
+ buffer + frame_length,
+ sizeof(*buffer) * (buffer_length - frame_length));
+ if (frame) {
+ memcpy(buffer + buffer_length - frame_length,
+ frame,
+ sizeof(*buffer) * frame_length);
+ } else {
+ memset(buffer + buffer_length - frame_length,
+ 0,
+ sizeof(*buffer) * frame_length);
+ }
+}
+
+// Transforms the signal from time to frequency domain.
+// Inputs:
+// * |time_data| is the signal in the time domain.
+// * |time_data_length| is the length of the analysis buffer.
+// * |magnitude_length| is the length of the spectrum magnitude, which equals
+// the length of both |real| and |imag| (time_data_length / 2 + 1).
+// Outputs:
+// * |time_data| is the signal in the frequency domain.
+// * |real| is the real part of the frequency domain.
+// * |imag| is the imaginary part of the frequency domain.
+// * |magn| is the calculated signal magnitude in the frequency domain.
+static void FFT(NoiseSuppressionC* self,
+ float* time_data,
+ size_t time_data_length,
+ size_t magnitude_length,
+ float* real,
+ float* imag,
+ float* magn) {
+ size_t i;
+
+ assert(magnitude_length == time_data_length / 2 + 1);
+
+ WebRtc_rdft(time_data_length, 1, time_data, self->ip, self->wfft);
+
+ imag[0] = 0;
+ real[0] = time_data[0];
+ magn[0] = fabsf(real[0]) + 1.f;
+ imag[magnitude_length - 1] = 0;
+ real[magnitude_length - 1] = time_data[1];
+ magn[magnitude_length - 1] = fabsf(real[magnitude_length - 1]) + 1.f;
+ for (i = 1; i < magnitude_length - 1; ++i) {
+ real[i] = time_data[2 * i];
+ imag[i] = time_data[2 * i + 1];
+ // Magnitude spectrum.
+ magn[i] = sqrtf(real[i] * real[i] + imag[i] * imag[i]) + 1.f;
+ }
+}
+
+// Transforms the signal from frequency to time domain.
+// Inputs:
+// * |real| is the real part of the frequency domain.
+// * |imag| is the imaginary part of the frequency domain.
+// * |magnitude_length| is the length of the spectrum magnitude, which equals
+// the length of both |real| and |imag|.
+// * |time_data_length| is the length of the analysis buffer
+// (2 * (magnitude_length - 1)).
+// Output:
+// * |time_data| is the signal in the time domain.
+static void IFFT(NoiseSuppressionC* self,
+ const float* real,
+ const float* imag,
+ size_t magnitude_length,
+ size_t time_data_length,
+ float* time_data) {
+ size_t i;
+
+ assert(time_data_length == 2 * (magnitude_length - 1));
+
+ time_data[0] = real[0];
+ time_data[1] = real[magnitude_length - 1];
+ for (i = 1; i < magnitude_length - 1; ++i) {
+ time_data[2 * i] = real[i];
+ time_data[2 * i + 1] = imag[i];
+ }
+ WebRtc_rdft(time_data_length, -1, time_data, self->ip, self->wfft);
+
+ for (i = 0; i < time_data_length; ++i) {
+ time_data[i] *= 2.f / time_data_length; // FFT scaling.
+ }
+}
+
+// Calculates the energy of a buffer.
+// Inputs:
+// * |buffer| is the buffer over which the energy is calculated.
+// * |length| is the length of the buffer.
+// Returns the calculated energy.
+static float Energy(const float* buffer, size_t length) {
+ size_t i;
+ float energy = 0.f;
+
+ for (i = 0; i < length; ++i) {
+ energy += buffer[i] * buffer[i];
+ }
+
+ return energy;
+}
+
+// Windows a buffer.
+// Inputs:
+// * |window| is the window by which to multiply.
+// * |data| is the data without windowing.
+// * |length| is the length of the window and data.
+// Output:
+// * |data_windowed| is the windowed data.
+static void Windowing(const float* window,
+ const float* data,
+ size_t length,
+ float* data_windowed) {
+ size_t i;
+
+ for (i = 0; i < length; ++i) {
+ data_windowed[i] = window[i] * data[i];
+ }
+}
+
+// Estimate prior SNR decision-directed and compute DD based Wiener Filter.
+// Input:
+// * |magn| is the signal magnitude spectrum estimate.
+// Output:
+// * |theFilter| is the frequency response of the computed Wiener filter.
+static void ComputeDdBasedWienerFilter(const NoiseSuppressionC* self,
+ const float* magn,
+ float* theFilter) {
+ size_t i;
+ float snrPrior, previousEstimateStsa, currentEstimateStsa;
+
+ for (i = 0; i < self->magnLen; i++) {
+ // Previous estimate: based on previous frame with gain filter.
+ previousEstimateStsa = self->magnPrevProcess[i] /
+ (self->noisePrev[i] + 0.0001f) * self->smooth[i];
+ // Post and prior SNR.
+ currentEstimateStsa = 0.f;
+ if (magn[i] > self->noise[i]) {
+ currentEstimateStsa = magn[i] / (self->noise[i] + 0.0001f) - 1.f;
+ }
+ // DD estimate is sum of two terms: current estimate and previous estimate.
+ // Directed decision update of |snrPrior|.
+ snrPrior = DD_PR_SNR * previousEstimateStsa +
+ (1.f - DD_PR_SNR) * currentEstimateStsa;
+ // Gain filter.
+ theFilter[i] = snrPrior / (self->overdrive + snrPrior);
+ } // End of loop over frequencies.
+}
+
+// Changes the aggressiveness of the noise suppression method.
+// |mode| = 0 is mild (6dB), |mode| = 1 is medium (10dB) and |mode| = 2 is
+// aggressive (15dB).
+// Returns 0 on success and -1 otherwise.
+int WebRtcNs_set_policy_core(NoiseSuppressionC* self, int mode) {
+ // Allow for modes: 0, 1, 2, 3.
+ if (mode < 0 || mode > 3) {
+ return (-1);
+ }
+
+ self->aggrMode = mode;
+ if (mode == 0) {
+ self->overdrive = 1.f;
+ self->denoiseBound = 0.5f;
+ self->gainmap = 0;
+ } else if (mode == 1) {
+ // self->overdrive = 1.25f;
+ self->overdrive = 1.f;
+ self->denoiseBound = 0.25f;
+ self->gainmap = 1;
+ } else if (mode == 2) {
+ // self->overdrive = 1.25f;
+ self->overdrive = 1.1f;
+ self->denoiseBound = 0.125f;
+ self->gainmap = 1;
+ } else if (mode == 3) {
+ // self->overdrive = 1.3f;
+ self->overdrive = 1.25f;
+ self->denoiseBound = 0.09f;
+ self->gainmap = 1;
+ }
+ return 0;
+}
+
+void WebRtcNs_AnalyzeCore(NoiseSuppressionC* self, const float* speechFrame) {
+ size_t i;
+ const size_t kStartBand = 5; // Skip first frequency bins during estimation.
+ int updateParsFlag;
+ float energy;
+ float signalEnergy = 0.f;
+ float sumMagn = 0.f;
+ float tmpFloat1, tmpFloat2, tmpFloat3;
+ float winData[ANAL_BLOCKL_MAX];
+ float magn[HALF_ANAL_BLOCKL], noise[HALF_ANAL_BLOCKL];
+ float snrLocPost[HALF_ANAL_BLOCKL], snrLocPrior[HALF_ANAL_BLOCKL];
+ float real[ANAL_BLOCKL_MAX], imag[HALF_ANAL_BLOCKL];
+ // Variables during startup.
+ float sum_log_i = 0.0;
+ float sum_log_i_square = 0.0;
+ float sum_log_magn = 0.0;
+ float sum_log_i_log_magn = 0.0;
+ float parametric_exp = 0.0;
+ float parametric_num = 0.0;
+
+ // Check that initiation has been done.
+ assert(self->initFlag == 1);
+ updateParsFlag = self->modelUpdatePars[0];
+
+ // Update analysis buffer for L band.
+ UpdateBuffer(speechFrame, self->blockLen, self->anaLen, self->analyzeBuf);
+
+ Windowing(self->window, self->analyzeBuf, self->anaLen, winData);
+ energy = Energy(winData, self->anaLen);
+ if (energy == 0.0) {
+ // We want to avoid updating statistics in this case:
+ // Updating feature statistics when we have zeros only will cause
+ // thresholds to move towards zero signal situations. This in turn has the
+ // effect that once the signal is "turned on" (non-zero values) everything
+ // will be treated as speech and there is no noise suppression effect.
+ // Depending on the duration of the inactive signal it takes a
+ // considerable amount of time for the system to learn what is noise and
+ // what is speech.
+ return;
+ }
+
+ self->blockInd++; // Update the block index only when we process a block.
+
+ FFT(self, winData, self->anaLen, self->magnLen, real, imag, magn);
+
+ for (i = 0; i < self->magnLen; i++) {
+ signalEnergy += real[i] * real[i] + imag[i] * imag[i];
+ sumMagn += magn[i];
+ if (self->blockInd < END_STARTUP_SHORT) {
+ if (i >= kStartBand) {
+ tmpFloat2 = logf((float)i);
+ sum_log_i += tmpFloat2;
+ sum_log_i_square += tmpFloat2 * tmpFloat2;
+ tmpFloat1 = logf(magn[i]);
+ sum_log_magn += tmpFloat1;
+ sum_log_i_log_magn += tmpFloat2 * tmpFloat1;
+ }
+ }
+ }
+ signalEnergy /= self->magnLen;
+ self->signalEnergy = signalEnergy;
+ self->sumMagn = sumMagn;
+
+ // Quantile noise estimate.
+ NoiseEstimation(self, magn, noise);
+ // Compute simplified noise model during startup.
+ if (self->blockInd < END_STARTUP_SHORT) {
+ // Estimate White noise.
+ self->whiteNoiseLevel += sumMagn / self->magnLen * self->overdrive;
+ // Estimate Pink noise parameters.
+ tmpFloat1 = sum_log_i_square * (self->magnLen - kStartBand);
+ tmpFloat1 -= (sum_log_i * sum_log_i);
+ tmpFloat2 =
+ (sum_log_i_square * sum_log_magn - sum_log_i * sum_log_i_log_magn);
+ tmpFloat3 = tmpFloat2 / tmpFloat1;
+ // Constrain the estimated spectrum to be positive.
+ if (tmpFloat3 < 0.f) {
+ tmpFloat3 = 0.f;
+ }
+ self->pinkNoiseNumerator += tmpFloat3;
+ tmpFloat2 = (sum_log_i * sum_log_magn);
+ tmpFloat2 -= (self->magnLen - kStartBand) * sum_log_i_log_magn;
+ tmpFloat3 = tmpFloat2 / tmpFloat1;
+ // Constrain the pink noise power to be in the interval [0, 1].
+ if (tmpFloat3 < 0.f) {
+ tmpFloat3 = 0.f;
+ }
+ if (tmpFloat3 > 1.f) {
+ tmpFloat3 = 1.f;
+ }
+ self->pinkNoiseExp += tmpFloat3;
+
+ // Calculate frequency independent parts of parametric noise estimate.
+ if (self->pinkNoiseExp > 0.f) {
+ // Use pink noise estimate.
+ parametric_num =
+ expf(self->pinkNoiseNumerator / (float)(self->blockInd + 1));
+ parametric_num *= (float)(self->blockInd + 1);
+ parametric_exp = self->pinkNoiseExp / (float)(self->blockInd + 1);
+ }
+ for (i = 0; i < self->magnLen; i++) {
+ // Estimate the background noise using the white and pink noise
+ // parameters.
+ if (self->pinkNoiseExp == 0.f) {
+ // Use white noise estimate.
+ self->parametricNoise[i] = self->whiteNoiseLevel;
+ } else {
+ // Use pink noise estimate.
+ float use_band = (float)(i < kStartBand ? kStartBand : i);
+ self->parametricNoise[i] =
+ parametric_num / powf(use_band, parametric_exp);
+ }
+ // Weight quantile noise with modeled noise.
+ noise[i] *= (self->blockInd);
+ tmpFloat2 =
+ self->parametricNoise[i] * (END_STARTUP_SHORT - self->blockInd);
+ noise[i] += (tmpFloat2 / (float)(self->blockInd + 1));
+ noise[i] /= END_STARTUP_SHORT;
+ }
+ }
+ // Compute average signal during END_STARTUP_LONG time:
+ // used to normalize spectral difference measure.
+ if (self->blockInd < END_STARTUP_LONG) {
+ self->featureData[5] *= self->blockInd;
+ self->featureData[5] += signalEnergy;
+ self->featureData[5] /= (self->blockInd + 1);
+ }
+
+ // Post and prior SNR needed for SpeechNoiseProb.
+ ComputeSnr(self, magn, noise, snrLocPrior, snrLocPost);
+
+ FeatureUpdate(self, magn, updateParsFlag);
+ SpeechNoiseProb(self, self->speechProb, snrLocPrior, snrLocPost);
+ UpdateNoiseEstimate(self, magn, snrLocPrior, snrLocPost, noise);
+
+ // Keep track of noise spectrum for next frame.
+ memcpy(self->noise, noise, sizeof(*noise) * self->magnLen);
+ memcpy(self->magnPrevAnalyze, magn, sizeof(*magn) * self->magnLen);
+}
+
+void WebRtcNs_ProcessCore(NoiseSuppressionC* self,
+ const float* const* speechFrame,
+ size_t num_bands,
+ float* const* outFrame) {
+ // Main routine for noise reduction.
+ int flagHB = 0;
+ size_t i, j;
+
+ float energy1, energy2, gain, factor, factor1, factor2;
+ float fout[BLOCKL_MAX];
+ float winData[ANAL_BLOCKL_MAX];
+ float magn[HALF_ANAL_BLOCKL];
+ float theFilter[HALF_ANAL_BLOCKL], theFilterTmp[HALF_ANAL_BLOCKL];
+ float real[ANAL_BLOCKL_MAX], imag[HALF_ANAL_BLOCKL];
+
+ // SWB variables.
+ int deltaBweHB = 1;
+ int deltaGainHB = 1;
+ float decayBweHB = 1.0;
+ float gainMapParHB = 1.0;
+ float gainTimeDomainHB = 1.0;
+ float avgProbSpeechHB, avgProbSpeechHBTmp, avgFilterGainHB, gainModHB;
+ float sumMagnAnalyze, sumMagnProcess;
+
+ // Check that initiation has been done.
+ assert(self->initFlag == 1);
+ assert((num_bands - 1) <= NUM_HIGH_BANDS_MAX);
+
+ const float* const* speechFrameHB = NULL;
+ float* const* outFrameHB = NULL;
+ size_t num_high_bands = 0;
+ if (num_bands > 1) {
+ speechFrameHB = &speechFrame[1];
+ outFrameHB = &outFrame[1];
+ num_high_bands = num_bands - 1;
+ flagHB = 1;
+ // Range for averaging low band quantities for H band gain.
+ deltaBweHB = (int)self->magnLen / 4;
+ deltaGainHB = deltaBweHB;
+ }
+
+ // Update analysis buffer for L band.
+ UpdateBuffer(speechFrame[0], self->blockLen, self->anaLen, self->dataBuf);
+
+ if (flagHB == 1) {
+ // Update analysis buffer for H bands.
+ for (i = 0; i < num_high_bands; ++i) {
+ UpdateBuffer(speechFrameHB[i],
+ self->blockLen,
+ self->anaLen,
+ self->dataBufHB[i]);
+ }
+ }
+
+ Windowing(self->window, self->dataBuf, self->anaLen, winData);
+ energy1 = Energy(winData, self->anaLen);
+ if (energy1 == 0.0) {
+ // Synthesize the special case of zero input.
+ // Read out fully processed segment.
+ for (i = self->windShift; i < self->blockLen + self->windShift; i++) {
+ fout[i - self->windShift] = self->syntBuf[i];
+ }
+ // Update synthesis buffer.
+ UpdateBuffer(NULL, self->blockLen, self->anaLen, self->syntBuf);
+
+ for (i = 0; i < self->blockLen; ++i)
+ outFrame[0][i] =
+ WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, fout[i], WEBRTC_SPL_WORD16_MIN);
+
+ // For time-domain gain of HB.
+ if (flagHB == 1) {
+ for (i = 0; i < num_high_bands; ++i) {
+ for (j = 0; j < self->blockLen; ++j) {
+ outFrameHB[i][j] = WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
+ self->dataBufHB[i][j],
+ WEBRTC_SPL_WORD16_MIN);
+ }
+ }
+ }
+
+ return;
+ }
+
+ FFT(self, winData, self->anaLen, self->magnLen, real, imag, magn);
+
+ if (self->blockInd < END_STARTUP_SHORT) {
+ for (i = 0; i < self->magnLen; i++) {
+ self->initMagnEst[i] += magn[i];
+ }
+ }
+
+ ComputeDdBasedWienerFilter(self, magn, theFilter);
+
+ for (i = 0; i < self->magnLen; i++) {
+ // Flooring bottom.
+ if (theFilter[i] < self->denoiseBound) {
+ theFilter[i] = self->denoiseBound;
+ }
+ // Flooring top.
+ if (theFilter[i] > 1.f) {
+ theFilter[i] = 1.f;
+ }
+ if (self->blockInd < END_STARTUP_SHORT) {
+ theFilterTmp[i] =
+ (self->initMagnEst[i] - self->overdrive * self->parametricNoise[i]);
+ theFilterTmp[i] /= (self->initMagnEst[i] + 0.0001f);
+ // Flooring bottom.
+ if (theFilterTmp[i] < self->denoiseBound) {
+ theFilterTmp[i] = self->denoiseBound;
+ }
+ // Flooring top.
+ if (theFilterTmp[i] > 1.f) {
+ theFilterTmp[i] = 1.f;
+ }
+ // Weight the two suppression filters.
+ theFilter[i] *= (self->blockInd);
+ theFilterTmp[i] *= (END_STARTUP_SHORT - self->blockInd);
+ theFilter[i] += theFilterTmp[i];
+ theFilter[i] /= (END_STARTUP_SHORT);
+ }
+
+ self->smooth[i] = theFilter[i];
+ real[i] *= self->smooth[i];
+ imag[i] *= self->smooth[i];
+ }
+ // Keep track of |magn| spectrum for next frame.
+ memcpy(self->magnPrevProcess, magn, sizeof(*magn) * self->magnLen);
+ memcpy(self->noisePrev, self->noise, sizeof(self->noise[0]) * self->magnLen);
+ // Back to time domain.
+ IFFT(self, real, imag, self->magnLen, self->anaLen, winData);
+
+ // Scale factor: only do it after END_STARTUP_LONG time.
+ factor = 1.f;
+ if (self->gainmap == 1 && self->blockInd > END_STARTUP_LONG) {
+ factor1 = 1.f;
+ factor2 = 1.f;
+
+ energy2 = Energy(winData, self->anaLen);
+ gain = (float)sqrt(energy2 / (energy1 + 1.f));
+
+ // Scaling for new version.
+ if (gain > B_LIM) {
+ factor1 = 1.f + 1.3f * (gain - B_LIM);
+ if (gain * factor1 > 1.f) {
+ factor1 = 1.f / gain;
+ }
+ }
+ if (gain < B_LIM) {
+ // Don't reduce scale too much for pause regions:
+ // attenuation here should be controlled by flooring.
+ if (gain <= self->denoiseBound) {
+ gain = self->denoiseBound;
+ }
+ factor2 = 1.f - 0.3f * (B_LIM - gain);
+ }
+ // Combine both scales with speech/noise prob:
+ // note prior (priorSpeechProb) is not frequency dependent.
+ factor = self->priorSpeechProb * factor1 +
+ (1.f - self->priorSpeechProb) * factor2;
+ } // Out of self->gainmap == 1.
+
+ Windowing(self->window, winData, self->anaLen, winData);
+
+ // Synthesis.
+ for (i = 0; i < self->anaLen; i++) {
+ self->syntBuf[i] += factor * winData[i];
+ }
+ // Read out fully processed segment.
+ for (i = self->windShift; i < self->blockLen + self->windShift; i++) {
+ fout[i - self->windShift] = self->syntBuf[i];
+ }
+ // Update synthesis buffer.
+ UpdateBuffer(NULL, self->blockLen, self->anaLen, self->syntBuf);
+
+ for (i = 0; i < self->blockLen; ++i)
+ outFrame[0][i] =
+ WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, fout[i], WEBRTC_SPL_WORD16_MIN);
+
+ // For time-domain gain of HB.
+ if (flagHB == 1) {
+ // Average speech prob from low band.
+ // Average over second half (i.e., 4->8kHz) of frequencies spectrum.
+ avgProbSpeechHB = 0.0;
+ for (i = self->magnLen - deltaBweHB - 1; i < self->magnLen - 1; i++) {
+ avgProbSpeechHB += self->speechProb[i];
+ }
+ avgProbSpeechHB = avgProbSpeechHB / ((float)deltaBweHB);
+ // If the speech was suppressed by a component between Analyze and
+ // Process, for example the AEC, then it should not be considered speech
+ // for high band suppression purposes.
+ sumMagnAnalyze = 0;
+ sumMagnProcess = 0;
+ for (i = 0; i < self->magnLen; ++i) {
+ sumMagnAnalyze += self->magnPrevAnalyze[i];
+ sumMagnProcess += self->magnPrevProcess[i];
+ }
+ avgProbSpeechHB *= sumMagnProcess / sumMagnAnalyze;
+ // Average filter gain from low band.
+ // Average over second half (i.e., 4->8kHz) of frequencies spectrum.
+ avgFilterGainHB = 0.0;
+ for (i = self->magnLen - deltaGainHB - 1; i < self->magnLen - 1; i++) {
+ avgFilterGainHB += self->smooth[i];
+ }
+ avgFilterGainHB = avgFilterGainHB / ((float)(deltaGainHB));
+ avgProbSpeechHBTmp = 2.f * avgProbSpeechHB - 1.f;
+ // Gain based on speech probability.
+ gainModHB = 0.5f * (1.f + (float)tanh(gainMapParHB * avgProbSpeechHBTmp));
+ // Combine gain with low band gain.
+ gainTimeDomainHB = 0.5f * gainModHB + 0.5f * avgFilterGainHB;
+ if (avgProbSpeechHB >= 0.5f) {
+ gainTimeDomainHB = 0.25f * gainModHB + 0.75f * avgFilterGainHB;
+ }
+ gainTimeDomainHB = gainTimeDomainHB * decayBweHB;
+ // Make sure gain is within flooring range.
+ // Flooring bottom.
+ if (gainTimeDomainHB < self->denoiseBound) {
+ gainTimeDomainHB = self->denoiseBound;
+ }
+ // Flooring top.
+ if (gainTimeDomainHB > 1.f) {
+ gainTimeDomainHB = 1.f;
+ }
+ // Apply gain.
+ for (i = 0; i < num_high_bands; ++i) {
+ for (j = 0; j < self->blockLen; j++) {
+ outFrameHB[i][j] =
+ WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
+ gainTimeDomainHB * self->dataBufHB[i][j],
+ WEBRTC_SPL_WORD16_MIN);
+ }
+ }
+ } // End of H band gain computation.
+}