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diff --git a/third_party/webrtc/src/webrtc/modules/audio_processing/aec/aec_core.c b/third_party/webrtc/src/webrtc/modules/audio_processing/aec/aec_core.c
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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.
+ */
+
+/*
+ * The core AEC algorithm, which is presented with time-aligned signals.
+ */
+
+#include "webrtc/modules/audio_processing/aec/aec_core.h"
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+#include <stdio.h>
+#endif
+
+#include <assert.h>
+#include <math.h>
+#include <stddef.h> // size_t
+#include <stdlib.h>
+#include <string.h>
+
+#include "webrtc/common_audio/ring_buffer.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/modules/audio_processing/aec/aec_common.h"
+#include "webrtc/modules/audio_processing/aec/aec_core_internal.h"
+#include "webrtc/modules/audio_processing/aec/aec_rdft.h"
+#include "webrtc/modules/audio_processing/logging/aec_logging.h"
+#include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
+#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/typedefs.h"
+
+
+// Buffer size (samples)
+static const size_t kBufSizePartitions = 250; // 1 second of audio in 16 kHz.
+
+// Metrics
+static const int subCountLen = 4;
+static const int countLen = 50;
+static const int kDelayMetricsAggregationWindow = 1250; // 5 seconds at 16 kHz.
+
+// Quantities to control H band scaling for SWB input
+static const int flagHbandCn = 1; // flag for adding comfort noise in H band
+static const float cnScaleHband =
+ (float)0.4; // scale for comfort noise in H band
+// Initial bin for averaging nlp gain in low band
+static const int freqAvgIc = PART_LEN / 2;
+
+// Matlab code to produce table:
+// win = sqrt(hanning(63)); win = [0 ; win(1:32)];
+// fprintf(1, '\t%.14f, %.14f, %.14f,\n', win);
+ALIGN16_BEG const float ALIGN16_END WebRtcAec_sqrtHanning[65] = {
+ 0.00000000000000f, 0.02454122852291f, 0.04906767432742f, 0.07356456359967f,
+ 0.09801714032956f, 0.12241067519922f, 0.14673047445536f, 0.17096188876030f,
+ 0.19509032201613f, 0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
+ 0.29028467725446f, 0.31368174039889f, 0.33688985339222f, 0.35989503653499f,
+ 0.38268343236509f, 0.40524131400499f, 0.42755509343028f, 0.44961132965461f,
+ 0.47139673682600f, 0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
+ 0.55557023301960f, 0.57580819141785f, 0.59569930449243f, 0.61523159058063f,
+ 0.63439328416365f, 0.65317284295378f, 0.67155895484702f, 0.68954054473707f,
+ 0.70710678118655f, 0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
+ 0.77301045336274f, 0.78834642762661f, 0.80320753148064f, 0.81758481315158f,
+ 0.83146961230255f, 0.84485356524971f, 0.85772861000027f, 0.87008699110871f,
+ 0.88192126434835f, 0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
+ 0.92387953251129f, 0.93299279883474f, 0.94154406518302f, 0.94952818059304f,
+ 0.95694033573221f, 0.96377606579544f, 0.97003125319454f, 0.97570213003853f,
+ 0.98078528040323f, 0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
+ 0.99518472667220f, 0.99729045667869f, 0.99879545620517f, 0.99969881869620f,
+ 1.00000000000000f};
+
+// Matlab code to produce table:
+// weightCurve = [0 ; 0.3 * sqrt(linspace(0,1,64))' + 0.1];
+// fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', weightCurve);
+ALIGN16_BEG const float ALIGN16_END WebRtcAec_weightCurve[65] = {
+ 0.0000f, 0.1000f, 0.1378f, 0.1535f, 0.1655f, 0.1756f, 0.1845f, 0.1926f,
+ 0.2000f, 0.2069f, 0.2134f, 0.2195f, 0.2254f, 0.2309f, 0.2363f, 0.2414f,
+ 0.2464f, 0.2512f, 0.2558f, 0.2604f, 0.2648f, 0.2690f, 0.2732f, 0.2773f,
+ 0.2813f, 0.2852f, 0.2890f, 0.2927f, 0.2964f, 0.3000f, 0.3035f, 0.3070f,
+ 0.3104f, 0.3138f, 0.3171f, 0.3204f, 0.3236f, 0.3268f, 0.3299f, 0.3330f,
+ 0.3360f, 0.3390f, 0.3420f, 0.3449f, 0.3478f, 0.3507f, 0.3535f, 0.3563f,
+ 0.3591f, 0.3619f, 0.3646f, 0.3673f, 0.3699f, 0.3726f, 0.3752f, 0.3777f,
+ 0.3803f, 0.3828f, 0.3854f, 0.3878f, 0.3903f, 0.3928f, 0.3952f, 0.3976f,
+ 0.4000f};
+
+// Matlab code to produce table:
+// overDriveCurve = [sqrt(linspace(0,1,65))' + 1];
+// fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', overDriveCurve);
+ALIGN16_BEG const float ALIGN16_END WebRtcAec_overDriveCurve[65] = {
+ 1.0000f, 1.1250f, 1.1768f, 1.2165f, 1.2500f, 1.2795f, 1.3062f, 1.3307f,
+ 1.3536f, 1.3750f, 1.3953f, 1.4146f, 1.4330f, 1.4507f, 1.4677f, 1.4841f,
+ 1.5000f, 1.5154f, 1.5303f, 1.5449f, 1.5590f, 1.5728f, 1.5863f, 1.5995f,
+ 1.6124f, 1.6250f, 1.6374f, 1.6495f, 1.6614f, 1.6731f, 1.6847f, 1.6960f,
+ 1.7071f, 1.7181f, 1.7289f, 1.7395f, 1.7500f, 1.7603f, 1.7706f, 1.7806f,
+ 1.7906f, 1.8004f, 1.8101f, 1.8197f, 1.8292f, 1.8385f, 1.8478f, 1.8570f,
+ 1.8660f, 1.8750f, 1.8839f, 1.8927f, 1.9014f, 1.9100f, 1.9186f, 1.9270f,
+ 1.9354f, 1.9437f, 1.9520f, 1.9601f, 1.9682f, 1.9763f, 1.9843f, 1.9922f,
+ 2.0000f};
+
+// Delay Agnostic AEC parameters, still under development and may change.
+static const float kDelayQualityThresholdMax = 0.07f;
+static const float kDelayQualityThresholdMin = 0.01f;
+static const int kInitialShiftOffset = 5;
+#if !defined(WEBRTC_ANDROID)
+static const int kDelayCorrectionStart = 1500; // 10 ms chunks
+#endif
+
+// Target suppression levels for nlp modes.
+// log{0.001, 0.00001, 0.00000001}
+static const float kTargetSupp[3] = {-6.9f, -11.5f, -18.4f};
+
+// Two sets of parameters, one for the extended filter mode.
+static const float kExtendedMinOverDrive[3] = {3.0f, 6.0f, 15.0f};
+static const float kNormalMinOverDrive[3] = {1.0f, 2.0f, 5.0f};
+const float WebRtcAec_kExtendedSmoothingCoefficients[2][2] = {{0.9f, 0.1f},
+ {0.92f, 0.08f}};
+const float WebRtcAec_kNormalSmoothingCoefficients[2][2] = {{0.9f, 0.1f},
+ {0.93f, 0.07f}};
+
+// Number of partitions forming the NLP's "preferred" bands.
+enum {
+ kPrefBandSize = 24
+};
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+extern int webrtc_aec_instance_count;
+#endif
+
+WebRtcAecFilterFar WebRtcAec_FilterFar;
+WebRtcAecScaleErrorSignal WebRtcAec_ScaleErrorSignal;
+WebRtcAecFilterAdaptation WebRtcAec_FilterAdaptation;
+WebRtcAecOverdriveAndSuppress WebRtcAec_OverdriveAndSuppress;
+WebRtcAecComfortNoise WebRtcAec_ComfortNoise;
+WebRtcAecSubBandCoherence WebRtcAec_SubbandCoherence;
+
+__inline static float MulRe(float aRe, float aIm, float bRe, float bIm) {
+ return aRe * bRe - aIm * bIm;
+}
+
+__inline static float MulIm(float aRe, float aIm, float bRe, float bIm) {
+ return aRe * bIm + aIm * bRe;
+}
+
+static int CmpFloat(const void* a, const void* b) {
+ const float* da = (const float*)a;
+ const float* db = (const float*)b;
+
+ return (*da > *db) - (*da < *db);
+}
+
+static void FilterFar(AecCore* aec, float yf[2][PART_LEN1]) {
+ int i;
+ for (i = 0; i < aec->num_partitions; i++) {
+ int j;
+ int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
+ int pos = i * PART_LEN1;
+ // Check for wrap
+ if (i + aec->xfBufBlockPos >= aec->num_partitions) {
+ xPos -= aec->num_partitions * (PART_LEN1);
+ }
+
+ for (j = 0; j < PART_LEN1; j++) {
+ yf[0][j] += MulRe(aec->xfBuf[0][xPos + j],
+ aec->xfBuf[1][xPos + j],
+ aec->wfBuf[0][pos + j],
+ aec->wfBuf[1][pos + j]);
+ yf[1][j] += MulIm(aec->xfBuf[0][xPos + j],
+ aec->xfBuf[1][xPos + j],
+ aec->wfBuf[0][pos + j],
+ aec->wfBuf[1][pos + j]);
+ }
+ }
+}
+
+static void ScaleErrorSignal(AecCore* aec, float ef[2][PART_LEN1]) {
+ const float mu = aec->extended_filter_enabled ? kExtendedMu : aec->normal_mu;
+ const float error_threshold = aec->extended_filter_enabled
+ ? kExtendedErrorThreshold
+ : aec->normal_error_threshold;
+ int i;
+ float abs_ef;
+ for (i = 0; i < (PART_LEN1); i++) {
+ ef[0][i] /= (aec->xPow[i] + 1e-10f);
+ ef[1][i] /= (aec->xPow[i] + 1e-10f);
+ abs_ef = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
+
+ if (abs_ef > error_threshold) {
+ abs_ef = error_threshold / (abs_ef + 1e-10f);
+ ef[0][i] *= abs_ef;
+ ef[1][i] *= abs_ef;
+ }
+
+ // Stepsize factor
+ ef[0][i] *= mu;
+ ef[1][i] *= mu;
+ }
+}
+
+// Time-unconstrined filter adaptation.
+// TODO(andrew): consider for a low-complexity mode.
+// static void FilterAdaptationUnconstrained(AecCore* aec, float *fft,
+// float ef[2][PART_LEN1]) {
+// int i, j;
+// for (i = 0; i < aec->num_partitions; i++) {
+// int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
+// int pos;
+// // Check for wrap
+// if (i + aec->xfBufBlockPos >= aec->num_partitions) {
+// xPos -= aec->num_partitions * PART_LEN1;
+// }
+//
+// pos = i * PART_LEN1;
+//
+// for (j = 0; j < PART_LEN1; j++) {
+// aec->wfBuf[0][pos + j] += MulRe(aec->xfBuf[0][xPos + j],
+// -aec->xfBuf[1][xPos + j],
+// ef[0][j], ef[1][j]);
+// aec->wfBuf[1][pos + j] += MulIm(aec->xfBuf[0][xPos + j],
+// -aec->xfBuf[1][xPos + j],
+// ef[0][j], ef[1][j]);
+// }
+// }
+//}
+
+static void FilterAdaptation(AecCore* aec, float* fft, float ef[2][PART_LEN1]) {
+ int i, j;
+ for (i = 0; i < aec->num_partitions; i++) {
+ int xPos = (i + aec->xfBufBlockPos) * (PART_LEN1);
+ int pos;
+ // Check for wrap
+ if (i + aec->xfBufBlockPos >= aec->num_partitions) {
+ xPos -= aec->num_partitions * PART_LEN1;
+ }
+
+ pos = i * PART_LEN1;
+
+ for (j = 0; j < PART_LEN; j++) {
+
+ fft[2 * j] = MulRe(aec->xfBuf[0][xPos + j],
+ -aec->xfBuf[1][xPos + j],
+ ef[0][j],
+ ef[1][j]);
+ fft[2 * j + 1] = MulIm(aec->xfBuf[0][xPos + j],
+ -aec->xfBuf[1][xPos + j],
+ ef[0][j],
+ ef[1][j]);
+ }
+ fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
+ -aec->xfBuf[1][xPos + PART_LEN],
+ ef[0][PART_LEN],
+ ef[1][PART_LEN]);
+
+ aec_rdft_inverse_128(fft);
+ memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
+
+ // fft scaling
+ {
+ float scale = 2.0f / PART_LEN2;
+ for (j = 0; j < PART_LEN; j++) {
+ fft[j] *= scale;
+ }
+ }
+ aec_rdft_forward_128(fft);
+
+ aec->wfBuf[0][pos] += fft[0];
+ aec->wfBuf[0][pos + PART_LEN] += fft[1];
+
+ for (j = 1; j < PART_LEN; j++) {
+ aec->wfBuf[0][pos + j] += fft[2 * j];
+ aec->wfBuf[1][pos + j] += fft[2 * j + 1];
+ }
+ }
+}
+
+static void OverdriveAndSuppress(AecCore* aec,
+ float hNl[PART_LEN1],
+ const float hNlFb,
+ float efw[2][PART_LEN1]) {
+ int i;
+ for (i = 0; i < PART_LEN1; i++) {
+ // Weight subbands
+ if (hNl[i] > hNlFb) {
+ hNl[i] = WebRtcAec_weightCurve[i] * hNlFb +
+ (1 - WebRtcAec_weightCurve[i]) * hNl[i];
+ }
+ hNl[i] = powf(hNl[i], aec->overDriveSm * WebRtcAec_overDriveCurve[i]);
+
+ // Suppress error signal
+ efw[0][i] *= hNl[i];
+ efw[1][i] *= hNl[i];
+
+ // Ooura fft returns incorrect sign on imaginary component. It matters here
+ // because we are making an additive change with comfort noise.
+ efw[1][i] *= -1;
+ }
+}
+
+static int PartitionDelay(const AecCore* aec) {
+ // Measures the energy in each filter partition and returns the partition with
+ // highest energy.
+ // TODO(bjornv): Spread computational cost by computing one partition per
+ // block?
+ float wfEnMax = 0;
+ int i;
+ int delay = 0;
+
+ for (i = 0; i < aec->num_partitions; i++) {
+ int j;
+ int pos = i * PART_LEN1;
+ float wfEn = 0;
+ for (j = 0; j < PART_LEN1; j++) {
+ wfEn += aec->wfBuf[0][pos + j] * aec->wfBuf[0][pos + j] +
+ aec->wfBuf[1][pos + j] * aec->wfBuf[1][pos + j];
+ }
+
+ if (wfEn > wfEnMax) {
+ wfEnMax = wfEn;
+ delay = i;
+ }
+ }
+ return delay;
+}
+
+// Threshold to protect against the ill-effects of a zero far-end.
+const float WebRtcAec_kMinFarendPSD = 15;
+
+// Updates the following smoothed Power Spectral Densities (PSD):
+// - sd : near-end
+// - se : residual echo
+// - sx : far-end
+// - sde : cross-PSD of near-end and residual echo
+// - sxd : cross-PSD of near-end and far-end
+//
+// In addition to updating the PSDs, also the filter diverge state is determined
+// upon actions are taken.
+static void SmoothedPSD(AecCore* aec,
+ float efw[2][PART_LEN1],
+ float dfw[2][PART_LEN1],
+ float xfw[2][PART_LEN1]) {
+ // Power estimate smoothing coefficients.
+ const float* ptrGCoh = aec->extended_filter_enabled
+ ? WebRtcAec_kExtendedSmoothingCoefficients[aec->mult - 1]
+ : WebRtcAec_kNormalSmoothingCoefficients[aec->mult - 1];
+ int i;
+ float sdSum = 0, seSum = 0;
+
+ for (i = 0; i < PART_LEN1; i++) {
+ aec->sd[i] = ptrGCoh[0] * aec->sd[i] +
+ ptrGCoh[1] * (dfw[0][i] * dfw[0][i] + dfw[1][i] * dfw[1][i]);
+ aec->se[i] = ptrGCoh[0] * aec->se[i] +
+ ptrGCoh[1] * (efw[0][i] * efw[0][i] + efw[1][i] * efw[1][i]);
+ // We threshold here to protect against the ill-effects of a zero farend.
+ // The threshold is not arbitrarily chosen, but balances protection and
+ // adverse interaction with the algorithm's tuning.
+ // TODO(bjornv): investigate further why this is so sensitive.
+ aec->sx[i] =
+ ptrGCoh[0] * aec->sx[i] +
+ ptrGCoh[1] * WEBRTC_SPL_MAX(
+ xfw[0][i] * xfw[0][i] + xfw[1][i] * xfw[1][i],
+ WebRtcAec_kMinFarendPSD);
+
+ aec->sde[i][0] =
+ ptrGCoh[0] * aec->sde[i][0] +
+ ptrGCoh[1] * (dfw[0][i] * efw[0][i] + dfw[1][i] * efw[1][i]);
+ aec->sde[i][1] =
+ ptrGCoh[0] * aec->sde[i][1] +
+ ptrGCoh[1] * (dfw[0][i] * efw[1][i] - dfw[1][i] * efw[0][i]);
+
+ aec->sxd[i][0] =
+ ptrGCoh[0] * aec->sxd[i][0] +
+ ptrGCoh[1] * (dfw[0][i] * xfw[0][i] + dfw[1][i] * xfw[1][i]);
+ aec->sxd[i][1] =
+ ptrGCoh[0] * aec->sxd[i][1] +
+ ptrGCoh[1] * (dfw[0][i] * xfw[1][i] - dfw[1][i] * xfw[0][i]);
+
+ sdSum += aec->sd[i];
+ seSum += aec->se[i];
+ }
+
+ // Divergent filter safeguard.
+ aec->divergeState = (aec->divergeState ? 1.05f : 1.0f) * seSum > sdSum;
+
+ if (aec->divergeState)
+ memcpy(efw, dfw, sizeof(efw[0][0]) * 2 * PART_LEN1);
+
+ // Reset if error is significantly larger than nearend (13 dB).
+ if (!aec->extended_filter_enabled && seSum > (19.95f * sdSum))
+ memset(aec->wfBuf, 0, sizeof(aec->wfBuf));
+}
+
+// Window time domain data to be used by the fft.
+__inline static void WindowData(float* x_windowed, const float* x) {
+ int i;
+ for (i = 0; i < PART_LEN; i++) {
+ x_windowed[i] = x[i] * WebRtcAec_sqrtHanning[i];
+ x_windowed[PART_LEN + i] =
+ x[PART_LEN + i] * WebRtcAec_sqrtHanning[PART_LEN - i];
+ }
+}
+
+// Puts fft output data into a complex valued array.
+__inline static void StoreAsComplex(const float* data,
+ float data_complex[2][PART_LEN1]) {
+ int i;
+ data_complex[0][0] = data[0];
+ data_complex[1][0] = 0;
+ for (i = 1; i < PART_LEN; i++) {
+ data_complex[0][i] = data[2 * i];
+ data_complex[1][i] = data[2 * i + 1];
+ }
+ data_complex[0][PART_LEN] = data[1];
+ data_complex[1][PART_LEN] = 0;
+}
+
+static void SubbandCoherence(AecCore* aec,
+ float efw[2][PART_LEN1],
+ float xfw[2][PART_LEN1],
+ float* fft,
+ float* cohde,
+ float* cohxd) {
+ float dfw[2][PART_LEN1];
+ int i;
+
+ if (aec->delayEstCtr == 0)
+ aec->delayIdx = PartitionDelay(aec);
+
+ // Use delayed far.
+ memcpy(xfw,
+ aec->xfwBuf + aec->delayIdx * PART_LEN1,
+ sizeof(xfw[0][0]) * 2 * PART_LEN1);
+
+ // Windowed near fft
+ WindowData(fft, aec->dBuf);
+ aec_rdft_forward_128(fft);
+ StoreAsComplex(fft, dfw);
+
+ // Windowed error fft
+ WindowData(fft, aec->eBuf);
+ aec_rdft_forward_128(fft);
+ StoreAsComplex(fft, efw);
+
+ SmoothedPSD(aec, efw, dfw, xfw);
+
+ // Subband coherence
+ for (i = 0; i < PART_LEN1; i++) {
+ cohde[i] =
+ (aec->sde[i][0] * aec->sde[i][0] + aec->sde[i][1] * aec->sde[i][1]) /
+ (aec->sd[i] * aec->se[i] + 1e-10f);
+ cohxd[i] =
+ (aec->sxd[i][0] * aec->sxd[i][0] + aec->sxd[i][1] * aec->sxd[i][1]) /
+ (aec->sx[i] * aec->sd[i] + 1e-10f);
+ }
+}
+
+static void GetHighbandGain(const float* lambda, float* nlpGainHband) {
+ int i;
+
+ nlpGainHband[0] = (float)0.0;
+ for (i = freqAvgIc; i < PART_LEN1 - 1; i++) {
+ nlpGainHband[0] += lambda[i];
+ }
+ nlpGainHband[0] /= (float)(PART_LEN1 - 1 - freqAvgIc);
+}
+
+static void ComfortNoise(AecCore* aec,
+ float efw[2][PART_LEN1],
+ complex_t* comfortNoiseHband,
+ const float* noisePow,
+ const float* lambda) {
+ int i, num;
+ float rand[PART_LEN];
+ float noise, noiseAvg, tmp, tmpAvg;
+ int16_t randW16[PART_LEN];
+ complex_t u[PART_LEN1];
+
+ const float pi2 = 6.28318530717959f;
+
+ // Generate a uniform random array on [0 1]
+ WebRtcSpl_RandUArray(randW16, PART_LEN, &aec->seed);
+ for (i = 0; i < PART_LEN; i++) {
+ rand[i] = ((float)randW16[i]) / 32768;
+ }
+
+ // Reject LF noise
+ u[0][0] = 0;
+ u[0][1] = 0;
+ for (i = 1; i < PART_LEN1; i++) {
+ tmp = pi2 * rand[i - 1];
+
+ noise = sqrtf(noisePow[i]);
+ u[i][0] = noise * cosf(tmp);
+ u[i][1] = -noise * sinf(tmp);
+ }
+ u[PART_LEN][1] = 0;
+
+ for (i = 0; i < PART_LEN1; i++) {
+ // This is the proper weighting to match the background noise power
+ tmp = sqrtf(WEBRTC_SPL_MAX(1 - lambda[i] * lambda[i], 0));
+ // tmp = 1 - lambda[i];
+ efw[0][i] += tmp * u[i][0];
+ efw[1][i] += tmp * u[i][1];
+ }
+
+ // For H band comfort noise
+ // TODO: don't compute noise and "tmp" twice. Use the previous results.
+ noiseAvg = 0.0;
+ tmpAvg = 0.0;
+ num = 0;
+ if (aec->num_bands > 1 && flagHbandCn == 1) {
+
+ // average noise scale
+ // average over second half of freq spectrum (i.e., 4->8khz)
+ // TODO: we shouldn't need num. We know how many elements we're summing.
+ for (i = PART_LEN1 >> 1; i < PART_LEN1; i++) {
+ num++;
+ noiseAvg += sqrtf(noisePow[i]);
+ }
+ noiseAvg /= (float)num;
+
+ // average nlp scale
+ // average over second half of freq spectrum (i.e., 4->8khz)
+ // TODO: we shouldn't need num. We know how many elements we're summing.
+ num = 0;
+ for (i = PART_LEN1 >> 1; i < PART_LEN1; i++) {
+ num++;
+ tmpAvg += sqrtf(WEBRTC_SPL_MAX(1 - lambda[i] * lambda[i], 0));
+ }
+ tmpAvg /= (float)num;
+
+ // Use average noise for H band
+ // TODO: we should probably have a new random vector here.
+ // Reject LF noise
+ u[0][0] = 0;
+ u[0][1] = 0;
+ for (i = 1; i < PART_LEN1; i++) {
+ tmp = pi2 * rand[i - 1];
+
+ // Use average noise for H band
+ u[i][0] = noiseAvg * (float)cos(tmp);
+ u[i][1] = -noiseAvg * (float)sin(tmp);
+ }
+ u[PART_LEN][1] = 0;
+
+ for (i = 0; i < PART_LEN1; i++) {
+ // Use average NLP weight for H band
+ comfortNoiseHband[i][0] = tmpAvg * u[i][0];
+ comfortNoiseHband[i][1] = tmpAvg * u[i][1];
+ }
+ }
+}
+
+static void InitLevel(PowerLevel* level) {
+ const float kBigFloat = 1E17f;
+
+ level->averagelevel = 0;
+ level->framelevel = 0;
+ level->minlevel = kBigFloat;
+ level->frsum = 0;
+ level->sfrsum = 0;
+ level->frcounter = 0;
+ level->sfrcounter = 0;
+}
+
+static void InitStats(Stats* stats) {
+ stats->instant = kOffsetLevel;
+ stats->average = kOffsetLevel;
+ stats->max = kOffsetLevel;
+ stats->min = kOffsetLevel * (-1);
+ stats->sum = 0;
+ stats->hisum = 0;
+ stats->himean = kOffsetLevel;
+ stats->counter = 0;
+ stats->hicounter = 0;
+}
+
+static void InitMetrics(AecCore* self) {
+ self->stateCounter = 0;
+ InitLevel(&self->farlevel);
+ InitLevel(&self->nearlevel);
+ InitLevel(&self->linoutlevel);
+ InitLevel(&self->nlpoutlevel);
+
+ InitStats(&self->erl);
+ InitStats(&self->erle);
+ InitStats(&self->aNlp);
+ InitStats(&self->rerl);
+}
+
+static void UpdateLevel(PowerLevel* level, float in[2][PART_LEN1]) {
+ // Do the energy calculation in the frequency domain. The FFT is performed on
+ // a segment of PART_LEN2 samples due to overlap, but we only want the energy
+ // of half that data (the last PART_LEN samples). Parseval's relation states
+ // that the energy is preserved according to
+ //
+ // \sum_{n=0}^{N-1} |x(n)|^2 = 1/N * \sum_{n=0}^{N-1} |X(n)|^2
+ // = ENERGY,
+ //
+ // where N = PART_LEN2. Since we are only interested in calculating the energy
+ // for the last PART_LEN samples we approximate by calculating ENERGY and
+ // divide by 2,
+ //
+ // \sum_{n=N/2}^{N-1} |x(n)|^2 ~= ENERGY / 2
+ //
+ // Since we deal with real valued time domain signals we only store frequency
+ // bins [0, PART_LEN], which is what |in| consists of. To calculate ENERGY we
+ // need to add the contribution from the missing part in
+ // [PART_LEN+1, PART_LEN2-1]. These values are, up to a phase shift, identical
+ // with the values in [1, PART_LEN-1], hence multiply those values by 2. This
+ // is the values in the for loop below, but multiplication by 2 and division
+ // by 2 cancel.
+
+ // TODO(bjornv): Investigate reusing energy calculations performed at other
+ // places in the code.
+ int k = 1;
+ // Imaginary parts are zero at end points and left out of the calculation.
+ float energy = (in[0][0] * in[0][0]) / 2;
+ energy += (in[0][PART_LEN] * in[0][PART_LEN]) / 2;
+
+ for (k = 1; k < PART_LEN; k++) {
+ energy += (in[0][k] * in[0][k] + in[1][k] * in[1][k]);
+ }
+ energy /= PART_LEN2;
+
+ level->sfrsum += energy;
+ level->sfrcounter++;
+
+ if (level->sfrcounter > subCountLen) {
+ level->framelevel = level->sfrsum / (subCountLen * PART_LEN);
+ level->sfrsum = 0;
+ level->sfrcounter = 0;
+ if (level->framelevel > 0) {
+ if (level->framelevel < level->minlevel) {
+ level->minlevel = level->framelevel; // New minimum.
+ } else {
+ level->minlevel *= (1 + 0.001f); // Small increase.
+ }
+ }
+ level->frcounter++;
+ level->frsum += level->framelevel;
+ if (level->frcounter > countLen) {
+ level->averagelevel = level->frsum / countLen;
+ level->frsum = 0;
+ level->frcounter = 0;
+ }
+ }
+}
+
+static void UpdateMetrics(AecCore* aec) {
+ float dtmp, dtmp2;
+
+ const float actThresholdNoisy = 8.0f;
+ const float actThresholdClean = 40.0f;
+ const float safety = 0.99995f;
+ const float noisyPower = 300000.0f;
+
+ float actThreshold;
+ float echo, suppressedEcho;
+
+ if (aec->echoState) { // Check if echo is likely present
+ aec->stateCounter++;
+ }
+
+ if (aec->farlevel.frcounter == 0) {
+
+ if (aec->farlevel.minlevel < noisyPower) {
+ actThreshold = actThresholdClean;
+ } else {
+ actThreshold = actThresholdNoisy;
+ }
+
+ if ((aec->stateCounter > (0.5f * countLen * subCountLen)) &&
+ (aec->farlevel.sfrcounter == 0)
+
+ // Estimate in active far-end segments only
+ &&
+ (aec->farlevel.averagelevel >
+ (actThreshold * aec->farlevel.minlevel))) {
+
+ // Subtract noise power
+ echo = aec->nearlevel.averagelevel - safety * aec->nearlevel.minlevel;
+
+ // ERL
+ dtmp = 10 * (float)log10(aec->farlevel.averagelevel /
+ aec->nearlevel.averagelevel +
+ 1e-10f);
+ dtmp2 = 10 * (float)log10(aec->farlevel.averagelevel / echo + 1e-10f);
+
+ aec->erl.instant = dtmp;
+ if (dtmp > aec->erl.max) {
+ aec->erl.max = dtmp;
+ }
+
+ if (dtmp < aec->erl.min) {
+ aec->erl.min = dtmp;
+ }
+
+ aec->erl.counter++;
+ aec->erl.sum += dtmp;
+ aec->erl.average = aec->erl.sum / aec->erl.counter;
+
+ // Upper mean
+ if (dtmp > aec->erl.average) {
+ aec->erl.hicounter++;
+ aec->erl.hisum += dtmp;
+ aec->erl.himean = aec->erl.hisum / aec->erl.hicounter;
+ }
+
+ // A_NLP
+ dtmp = 10 * (float)log10(aec->nearlevel.averagelevel /
+ (2 * aec->linoutlevel.averagelevel) +
+ 1e-10f);
+
+ // subtract noise power
+ suppressedEcho = 2 * (aec->linoutlevel.averagelevel -
+ safety * aec->linoutlevel.minlevel);
+
+ dtmp2 = 10 * (float)log10(echo / suppressedEcho + 1e-10f);
+
+ aec->aNlp.instant = dtmp2;
+ if (dtmp > aec->aNlp.max) {
+ aec->aNlp.max = dtmp;
+ }
+
+ if (dtmp < aec->aNlp.min) {
+ aec->aNlp.min = dtmp;
+ }
+
+ aec->aNlp.counter++;
+ aec->aNlp.sum += dtmp;
+ aec->aNlp.average = aec->aNlp.sum / aec->aNlp.counter;
+
+ // Upper mean
+ if (dtmp > aec->aNlp.average) {
+ aec->aNlp.hicounter++;
+ aec->aNlp.hisum += dtmp;
+ aec->aNlp.himean = aec->aNlp.hisum / aec->aNlp.hicounter;
+ }
+
+ // ERLE
+
+ // subtract noise power
+ suppressedEcho = 2 * (aec->nlpoutlevel.averagelevel -
+ safety * aec->nlpoutlevel.minlevel);
+
+ dtmp = 10 * (float)log10(aec->nearlevel.averagelevel /
+ (2 * aec->nlpoutlevel.averagelevel) +
+ 1e-10f);
+ dtmp2 = 10 * (float)log10(echo / suppressedEcho + 1e-10f);
+
+ dtmp = dtmp2;
+ aec->erle.instant = dtmp;
+ if (dtmp > aec->erle.max) {
+ aec->erle.max = dtmp;
+ }
+
+ if (dtmp < aec->erle.min) {
+ aec->erle.min = dtmp;
+ }
+
+ aec->erle.counter++;
+ aec->erle.sum += dtmp;
+ aec->erle.average = aec->erle.sum / aec->erle.counter;
+
+ // Upper mean
+ if (dtmp > aec->erle.average) {
+ aec->erle.hicounter++;
+ aec->erle.hisum += dtmp;
+ aec->erle.himean = aec->erle.hisum / aec->erle.hicounter;
+ }
+ }
+
+ aec->stateCounter = 0;
+ }
+}
+
+static void UpdateDelayMetrics(AecCore* self) {
+ int i = 0;
+ int delay_values = 0;
+ int median = 0;
+ int lookahead = WebRtc_lookahead(self->delay_estimator);
+ const int kMsPerBlock = PART_LEN / (self->mult * 8);
+ int64_t l1_norm = 0;
+
+ if (self->num_delay_values == 0) {
+ // We have no new delay value data. Even though -1 is a valid |median| in
+ // the sense that we allow negative values, it will practically never be
+ // used since multiples of |kMsPerBlock| will always be returned.
+ // We therefore use -1 to indicate in the logs that the delay estimator was
+ // not able to estimate the delay.
+ self->delay_median = -1;
+ self->delay_std = -1;
+ self->fraction_poor_delays = -1;
+ return;
+ }
+
+ // Start value for median count down.
+ delay_values = self->num_delay_values >> 1;
+ // Get median of delay values since last update.
+ for (i = 0; i < kHistorySizeBlocks; i++) {
+ delay_values -= self->delay_histogram[i];
+ if (delay_values < 0) {
+ median = i;
+ break;
+ }
+ }
+ // Account for lookahead.
+ self->delay_median = (median - lookahead) * kMsPerBlock;
+
+ // Calculate the L1 norm, with median value as central moment.
+ for (i = 0; i < kHistorySizeBlocks; i++) {
+ l1_norm += abs(i - median) * self->delay_histogram[i];
+ }
+ self->delay_std = (int)((l1_norm + self->num_delay_values / 2) /
+ self->num_delay_values) * kMsPerBlock;
+
+ // Determine fraction of delays that are out of bounds, that is, either
+ // negative (anti-causal system) or larger than the AEC filter length.
+ {
+ int num_delays_out_of_bounds = self->num_delay_values;
+ const int histogram_length = sizeof(self->delay_histogram) /
+ sizeof(self->delay_histogram[0]);
+ for (i = lookahead; i < lookahead + self->num_partitions; ++i) {
+ if (i < histogram_length)
+ num_delays_out_of_bounds -= self->delay_histogram[i];
+ }
+ self->fraction_poor_delays = (float)num_delays_out_of_bounds /
+ self->num_delay_values;
+ }
+
+ // Reset histogram.
+ memset(self->delay_histogram, 0, sizeof(self->delay_histogram));
+ self->num_delay_values = 0;
+
+ return;
+}
+
+static void TimeToFrequency(float time_data[PART_LEN2],
+ float freq_data[2][PART_LEN1],
+ int window) {
+ int i = 0;
+
+ // TODO(bjornv): Should we have a different function/wrapper for windowed FFT?
+ if (window) {
+ for (i = 0; i < PART_LEN; i++) {
+ time_data[i] *= WebRtcAec_sqrtHanning[i];
+ time_data[PART_LEN + i] *= WebRtcAec_sqrtHanning[PART_LEN - i];
+ }
+ }
+
+ aec_rdft_forward_128(time_data);
+ // Reorder.
+ freq_data[1][0] = 0;
+ freq_data[1][PART_LEN] = 0;
+ freq_data[0][0] = time_data[0];
+ freq_data[0][PART_LEN] = time_data[1];
+ for (i = 1; i < PART_LEN; i++) {
+ freq_data[0][i] = time_data[2 * i];
+ freq_data[1][i] = time_data[2 * i + 1];
+ }
+}
+
+static int MoveFarReadPtrWithoutSystemDelayUpdate(AecCore* self, int elements) {
+ WebRtc_MoveReadPtr(self->far_buf_windowed, elements);
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_MoveReadPtr(self->far_time_buf, elements);
+#endif
+ return WebRtc_MoveReadPtr(self->far_buf, elements);
+}
+
+static int SignalBasedDelayCorrection(AecCore* self) {
+ int delay_correction = 0;
+ int last_delay = -2;
+ assert(self != NULL);
+#if !defined(WEBRTC_ANDROID)
+ // On desktops, turn on correction after |kDelayCorrectionStart| frames. This
+ // is to let the delay estimation get a chance to converge. Also, if the
+ // playout audio volume is low (or even muted) the delay estimation can return
+ // a very large delay, which will break the AEC if it is applied.
+ if (self->frame_count < kDelayCorrectionStart) {
+ return 0;
+ }
+#endif
+
+ // 1. Check for non-negative delay estimate. Note that the estimates we get
+ // from the delay estimation are not compensated for lookahead. Hence, a
+ // negative |last_delay| is an invalid one.
+ // 2. Verify that there is a delay change. In addition, only allow a change
+ // if the delay is outside a certain region taking the AEC filter length
+ // into account.
+ // TODO(bjornv): Investigate if we can remove the non-zero delay change check.
+ // 3. Only allow delay correction if the delay estimation quality exceeds
+ // |delay_quality_threshold|.
+ // 4. Finally, verify that the proposed |delay_correction| is feasible by
+ // comparing with the size of the far-end buffer.
+ last_delay = WebRtc_last_delay(self->delay_estimator);
+ if ((last_delay >= 0) &&
+ (last_delay != self->previous_delay) &&
+ (WebRtc_last_delay_quality(self->delay_estimator) >
+ self->delay_quality_threshold)) {
+ int delay = last_delay - WebRtc_lookahead(self->delay_estimator);
+ // Allow for a slack in the actual delay, defined by a |lower_bound| and an
+ // |upper_bound|. The adaptive echo cancellation filter is currently
+ // |num_partitions| (of 64 samples) long. If the delay estimate is negative
+ // or at least 3/4 of the filter length we open up for correction.
+ const int lower_bound = 0;
+ const int upper_bound = self->num_partitions * 3 / 4;
+ const int do_correction = delay <= lower_bound || delay > upper_bound;
+ if (do_correction == 1) {
+ int available_read = (int)WebRtc_available_read(self->far_buf);
+ // With |shift_offset| we gradually rely on the delay estimates. For
+ // positive delays we reduce the correction by |shift_offset| to lower the
+ // risk of pushing the AEC into a non causal state. For negative delays
+ // we rely on the values up to a rounding error, hence compensate by 1
+ // element to make sure to push the delay into the causal region.
+ delay_correction = -delay;
+ delay_correction += delay > self->shift_offset ? self->shift_offset : 1;
+ self->shift_offset--;
+ self->shift_offset = (self->shift_offset <= 1 ? 1 : self->shift_offset);
+ if (delay_correction > available_read - self->mult - 1) {
+ // There is not enough data in the buffer to perform this shift. Hence,
+ // we do not rely on the delay estimate and do nothing.
+ delay_correction = 0;
+ } else {
+ self->previous_delay = last_delay;
+ ++self->delay_correction_count;
+ }
+ }
+ }
+ // Update the |delay_quality_threshold| once we have our first delay
+ // correction.
+ if (self->delay_correction_count > 0) {
+ float delay_quality = WebRtc_last_delay_quality(self->delay_estimator);
+ delay_quality = (delay_quality > kDelayQualityThresholdMax ?
+ kDelayQualityThresholdMax : delay_quality);
+ self->delay_quality_threshold =
+ (delay_quality > self->delay_quality_threshold ? delay_quality :
+ self->delay_quality_threshold);
+ }
+ return delay_correction;
+}
+
+static void NonLinearProcessing(AecCore* aec,
+ float* output,
+ float* const* outputH) {
+ float efw[2][PART_LEN1], xfw[2][PART_LEN1];
+ complex_t comfortNoiseHband[PART_LEN1];
+ float fft[PART_LEN2];
+ float scale, dtmp;
+ float nlpGainHband;
+ int i;
+ size_t j;
+
+ // Coherence and non-linear filter
+ float cohde[PART_LEN1], cohxd[PART_LEN1];
+ float hNlDeAvg, hNlXdAvg;
+ float hNl[PART_LEN1];
+ float hNlPref[kPrefBandSize];
+ float hNlFb = 0, hNlFbLow = 0;
+ const float prefBandQuant = 0.75f, prefBandQuantLow = 0.5f;
+ const int prefBandSize = kPrefBandSize / aec->mult;
+ const int minPrefBand = 4 / aec->mult;
+ // Power estimate smoothing coefficients.
+ const float* min_overdrive = aec->extended_filter_enabled
+ ? kExtendedMinOverDrive
+ : kNormalMinOverDrive;
+
+ // Filter energy
+ const int delayEstInterval = 10 * aec->mult;
+
+ float* xfw_ptr = NULL;
+
+ aec->delayEstCtr++;
+ if (aec->delayEstCtr == delayEstInterval) {
+ aec->delayEstCtr = 0;
+ }
+
+ // initialize comfort noise for H band
+ memset(comfortNoiseHband, 0, sizeof(comfortNoiseHband));
+ nlpGainHband = (float)0.0;
+ dtmp = (float)0.0;
+
+ // We should always have at least one element stored in |far_buf|.
+ assert(WebRtc_available_read(aec->far_buf_windowed) > 0);
+ // NLP
+ WebRtc_ReadBuffer(aec->far_buf_windowed, (void**)&xfw_ptr, &xfw[0][0], 1);
+
+ // TODO(bjornv): Investigate if we can reuse |far_buf_windowed| instead of
+ // |xfwBuf|.
+ // Buffer far.
+ memcpy(aec->xfwBuf, xfw_ptr, sizeof(float) * 2 * PART_LEN1);
+
+ WebRtcAec_SubbandCoherence(aec, efw, xfw, fft, cohde, cohxd);
+
+ hNlXdAvg = 0;
+ for (i = minPrefBand; i < prefBandSize + minPrefBand; i++) {
+ hNlXdAvg += cohxd[i];
+ }
+ hNlXdAvg /= prefBandSize;
+ hNlXdAvg = 1 - hNlXdAvg;
+
+ hNlDeAvg = 0;
+ for (i = minPrefBand; i < prefBandSize + minPrefBand; i++) {
+ hNlDeAvg += cohde[i];
+ }
+ hNlDeAvg /= prefBandSize;
+
+ if (hNlXdAvg < 0.75f && hNlXdAvg < aec->hNlXdAvgMin) {
+ aec->hNlXdAvgMin = hNlXdAvg;
+ }
+
+ if (hNlDeAvg > 0.98f && hNlXdAvg > 0.9f) {
+ aec->stNearState = 1;
+ } else if (hNlDeAvg < 0.95f || hNlXdAvg < 0.8f) {
+ aec->stNearState = 0;
+ }
+
+ if (aec->hNlXdAvgMin == 1) {
+ aec->echoState = 0;
+ aec->overDrive = min_overdrive[aec->nlp_mode];
+
+ if (aec->stNearState == 1) {
+ memcpy(hNl, cohde, sizeof(hNl));
+ hNlFb = hNlDeAvg;
+ hNlFbLow = hNlDeAvg;
+ } else {
+ for (i = 0; i < PART_LEN1; i++) {
+ hNl[i] = 1 - cohxd[i];
+ }
+ hNlFb = hNlXdAvg;
+ hNlFbLow = hNlXdAvg;
+ }
+ } else {
+
+ if (aec->stNearState == 1) {
+ aec->echoState = 0;
+ memcpy(hNl, cohde, sizeof(hNl));
+ hNlFb = hNlDeAvg;
+ hNlFbLow = hNlDeAvg;
+ } else {
+ aec->echoState = 1;
+ for (i = 0; i < PART_LEN1; i++) {
+ hNl[i] = WEBRTC_SPL_MIN(cohde[i], 1 - cohxd[i]);
+ }
+
+ // Select an order statistic from the preferred bands.
+ // TODO: Using quicksort now, but a selection algorithm may be preferred.
+ memcpy(hNlPref, &hNl[minPrefBand], sizeof(float) * prefBandSize);
+ qsort(hNlPref, prefBandSize, sizeof(float), CmpFloat);
+ hNlFb = hNlPref[(int)floor(prefBandQuant * (prefBandSize - 1))];
+ hNlFbLow = hNlPref[(int)floor(prefBandQuantLow * (prefBandSize - 1))];
+ }
+ }
+
+ // Track the local filter minimum to determine suppression overdrive.
+ if (hNlFbLow < 0.6f && hNlFbLow < aec->hNlFbLocalMin) {
+ aec->hNlFbLocalMin = hNlFbLow;
+ aec->hNlFbMin = hNlFbLow;
+ aec->hNlNewMin = 1;
+ aec->hNlMinCtr = 0;
+ }
+ aec->hNlFbLocalMin =
+ WEBRTC_SPL_MIN(aec->hNlFbLocalMin + 0.0008f / aec->mult, 1);
+ aec->hNlXdAvgMin = WEBRTC_SPL_MIN(aec->hNlXdAvgMin + 0.0006f / aec->mult, 1);
+
+ if (aec->hNlNewMin == 1) {
+ aec->hNlMinCtr++;
+ }
+ if (aec->hNlMinCtr == 2) {
+ aec->hNlNewMin = 0;
+ aec->hNlMinCtr = 0;
+ aec->overDrive =
+ WEBRTC_SPL_MAX(kTargetSupp[aec->nlp_mode] /
+ ((float)log(aec->hNlFbMin + 1e-10f) + 1e-10f),
+ min_overdrive[aec->nlp_mode]);
+ }
+
+ // Smooth the overdrive.
+ if (aec->overDrive < aec->overDriveSm) {
+ aec->overDriveSm = 0.99f * aec->overDriveSm + 0.01f * aec->overDrive;
+ } else {
+ aec->overDriveSm = 0.9f * aec->overDriveSm + 0.1f * aec->overDrive;
+ }
+
+ WebRtcAec_OverdriveAndSuppress(aec, hNl, hNlFb, efw);
+
+ // Add comfort noise.
+ WebRtcAec_ComfortNoise(aec, efw, comfortNoiseHband, aec->noisePow, hNl);
+
+ // TODO(bjornv): Investigate how to take the windowing below into account if
+ // needed.
+ if (aec->metricsMode == 1) {
+ // Note that we have a scaling by two in the time domain |eBuf|.
+ // In addition the time domain signal is windowed before transformation,
+ // losing half the energy on the average. We take care of the first
+ // scaling only in UpdateMetrics().
+ UpdateLevel(&aec->nlpoutlevel, efw);
+ }
+ // Inverse error fft.
+ fft[0] = efw[0][0];
+ fft[1] = efw[0][PART_LEN];
+ for (i = 1; i < PART_LEN; i++) {
+ fft[2 * i] = efw[0][i];
+ // Sign change required by Ooura fft.
+ fft[2 * i + 1] = -efw[1][i];
+ }
+ aec_rdft_inverse_128(fft);
+
+ // Overlap and add to obtain output.
+ scale = 2.0f / PART_LEN2;
+ for (i = 0; i < PART_LEN; i++) {
+ fft[i] *= scale; // fft scaling
+ fft[i] = fft[i] * WebRtcAec_sqrtHanning[i] + aec->outBuf[i];
+
+ fft[PART_LEN + i] *= scale; // fft scaling
+ aec->outBuf[i] = fft[PART_LEN + i] * WebRtcAec_sqrtHanning[PART_LEN - i];
+
+ // Saturate output to keep it in the allowed range.
+ output[i] = WEBRTC_SPL_SAT(
+ WEBRTC_SPL_WORD16_MAX, fft[i], WEBRTC_SPL_WORD16_MIN);
+ }
+
+ // For H band
+ if (aec->num_bands > 1) {
+
+ // H band gain
+ // average nlp over low band: average over second half of freq spectrum
+ // (4->8khz)
+ GetHighbandGain(hNl, &nlpGainHband);
+
+ // Inverse comfort_noise
+ if (flagHbandCn == 1) {
+ fft[0] = comfortNoiseHband[0][0];
+ fft[1] = comfortNoiseHband[PART_LEN][0];
+ for (i = 1; i < PART_LEN; i++) {
+ fft[2 * i] = comfortNoiseHband[i][0];
+ fft[2 * i + 1] = comfortNoiseHband[i][1];
+ }
+ aec_rdft_inverse_128(fft);
+ scale = 2.0f / PART_LEN2;
+ }
+
+ // compute gain factor
+ for (j = 0; j < aec->num_bands - 1; ++j) {
+ for (i = 0; i < PART_LEN; i++) {
+ dtmp = aec->dBufH[j][i];
+ dtmp = dtmp * nlpGainHband; // for variable gain
+
+ // add some comfort noise where Hband is attenuated
+ if (flagHbandCn == 1 && j == 0) {
+ fft[i] *= scale; // fft scaling
+ dtmp += cnScaleHband * fft[i];
+ }
+
+ // Saturate output to keep it in the allowed range.
+ outputH[j][i] = WEBRTC_SPL_SAT(
+ WEBRTC_SPL_WORD16_MAX, dtmp, WEBRTC_SPL_WORD16_MIN);
+ }
+ }
+ }
+
+ // Copy the current block to the old position.
+ memcpy(aec->dBuf, aec->dBuf + PART_LEN, sizeof(float) * PART_LEN);
+ memcpy(aec->eBuf, aec->eBuf + PART_LEN, sizeof(float) * PART_LEN);
+
+ // Copy the current block to the old position for H band
+ for (j = 0; j < aec->num_bands - 1; ++j) {
+ memcpy(aec->dBufH[j], aec->dBufH[j] + PART_LEN, sizeof(float) * PART_LEN);
+ }
+
+ memmove(aec->xfwBuf + PART_LEN1,
+ aec->xfwBuf,
+ sizeof(aec->xfwBuf) - sizeof(complex_t) * PART_LEN1);
+}
+
+static void ProcessBlock(AecCore* aec) {
+ size_t i;
+ float y[PART_LEN], e[PART_LEN];
+ float scale;
+
+ float fft[PART_LEN2];
+ float xf[2][PART_LEN1], yf[2][PART_LEN1], ef[2][PART_LEN1];
+ float df[2][PART_LEN1];
+ float far_spectrum = 0.0f;
+ float near_spectrum = 0.0f;
+ float abs_far_spectrum[PART_LEN1];
+ float abs_near_spectrum[PART_LEN1];
+
+ const float gPow[2] = {0.9f, 0.1f};
+
+ // Noise estimate constants.
+ const int noiseInitBlocks = 500 * aec->mult;
+ const float step = 0.1f;
+ const float ramp = 1.0002f;
+ const float gInitNoise[2] = {0.999f, 0.001f};
+
+ float nearend[PART_LEN];
+ float* nearend_ptr = NULL;
+ float output[PART_LEN];
+ float outputH[NUM_HIGH_BANDS_MAX][PART_LEN];
+ float* outputH_ptr[NUM_HIGH_BANDS_MAX];
+ for (i = 0; i < NUM_HIGH_BANDS_MAX; ++i) {
+ outputH_ptr[i] = outputH[i];
+ }
+
+ float* xf_ptr = NULL;
+
+ // Concatenate old and new nearend blocks.
+ for (i = 0; i < aec->num_bands - 1; ++i) {
+ WebRtc_ReadBuffer(aec->nearFrBufH[i],
+ (void**)&nearend_ptr,
+ nearend,
+ PART_LEN);
+ memcpy(aec->dBufH[i] + PART_LEN, nearend_ptr, sizeof(nearend));
+ }
+ WebRtc_ReadBuffer(aec->nearFrBuf, (void**)&nearend_ptr, nearend, PART_LEN);
+ memcpy(aec->dBuf + PART_LEN, nearend_ptr, sizeof(nearend));
+
+ // ---------- Ooura fft ----------
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ {
+ float farend[PART_LEN];
+ float* farend_ptr = NULL;
+ WebRtc_ReadBuffer(aec->far_time_buf, (void**)&farend_ptr, farend, 1);
+ RTC_AEC_DEBUG_WAV_WRITE(aec->farFile, farend_ptr, PART_LEN);
+ RTC_AEC_DEBUG_WAV_WRITE(aec->nearFile, nearend_ptr, PART_LEN);
+ }
+#endif
+
+ // We should always have at least one element stored in |far_buf|.
+ assert(WebRtc_available_read(aec->far_buf) > 0);
+ WebRtc_ReadBuffer(aec->far_buf, (void**)&xf_ptr, &xf[0][0], 1);
+
+ // Near fft
+ memcpy(fft, aec->dBuf, sizeof(float) * PART_LEN2);
+ TimeToFrequency(fft, df, 0);
+
+ // Power smoothing
+ for (i = 0; i < PART_LEN1; i++) {
+ far_spectrum = (xf_ptr[i] * xf_ptr[i]) +
+ (xf_ptr[PART_LEN1 + i] * xf_ptr[PART_LEN1 + i]);
+ aec->xPow[i] =
+ gPow[0] * aec->xPow[i] + gPow[1] * aec->num_partitions * far_spectrum;
+ // Calculate absolute spectra
+ abs_far_spectrum[i] = sqrtf(far_spectrum);
+
+ near_spectrum = df[0][i] * df[0][i] + df[1][i] * df[1][i];
+ aec->dPow[i] = gPow[0] * aec->dPow[i] + gPow[1] * near_spectrum;
+ // Calculate absolute spectra
+ abs_near_spectrum[i] = sqrtf(near_spectrum);
+ }
+
+ // Estimate noise power. Wait until dPow is more stable.
+ if (aec->noiseEstCtr > 50) {
+ for (i = 0; i < PART_LEN1; i++) {
+ if (aec->dPow[i] < aec->dMinPow[i]) {
+ aec->dMinPow[i] =
+ (aec->dPow[i] + step * (aec->dMinPow[i] - aec->dPow[i])) * ramp;
+ } else {
+ aec->dMinPow[i] *= ramp;
+ }
+ }
+ }
+
+ // Smooth increasing noise power from zero at the start,
+ // to avoid a sudden burst of comfort noise.
+ if (aec->noiseEstCtr < noiseInitBlocks) {
+ aec->noiseEstCtr++;
+ for (i = 0; i < PART_LEN1; i++) {
+ if (aec->dMinPow[i] > aec->dInitMinPow[i]) {
+ aec->dInitMinPow[i] = gInitNoise[0] * aec->dInitMinPow[i] +
+ gInitNoise[1] * aec->dMinPow[i];
+ } else {
+ aec->dInitMinPow[i] = aec->dMinPow[i];
+ }
+ }
+ aec->noisePow = aec->dInitMinPow;
+ } else {
+ aec->noisePow = aec->dMinPow;
+ }
+
+ // Block wise delay estimation used for logging
+ if (aec->delay_logging_enabled) {
+ if (WebRtc_AddFarSpectrumFloat(
+ aec->delay_estimator_farend, abs_far_spectrum, PART_LEN1) == 0) {
+ int delay_estimate = WebRtc_DelayEstimatorProcessFloat(
+ aec->delay_estimator, abs_near_spectrum, PART_LEN1);
+ if (delay_estimate >= 0) {
+ // Update delay estimate buffer.
+ aec->delay_histogram[delay_estimate]++;
+ aec->num_delay_values++;
+ }
+ if (aec->delay_metrics_delivered == 1 &&
+ aec->num_delay_values >= kDelayMetricsAggregationWindow) {
+ UpdateDelayMetrics(aec);
+ }
+ }
+ }
+
+ // Update the xfBuf block position.
+ aec->xfBufBlockPos--;
+ if (aec->xfBufBlockPos == -1) {
+ aec->xfBufBlockPos = aec->num_partitions - 1;
+ }
+
+ // Buffer xf
+ memcpy(aec->xfBuf[0] + aec->xfBufBlockPos * PART_LEN1,
+ xf_ptr,
+ sizeof(float) * PART_LEN1);
+ memcpy(aec->xfBuf[1] + aec->xfBufBlockPos * PART_LEN1,
+ &xf_ptr[PART_LEN1],
+ sizeof(float) * PART_LEN1);
+
+ memset(yf, 0, sizeof(yf));
+
+ // Filter far
+ WebRtcAec_FilterFar(aec, yf);
+
+ // Inverse fft to obtain echo estimate and error.
+ fft[0] = yf[0][0];
+ fft[1] = yf[0][PART_LEN];
+ for (i = 1; i < PART_LEN; i++) {
+ fft[2 * i] = yf[0][i];
+ fft[2 * i + 1] = yf[1][i];
+ }
+ aec_rdft_inverse_128(fft);
+
+ scale = 2.0f / PART_LEN2;
+ for (i = 0; i < PART_LEN; i++) {
+ y[i] = fft[PART_LEN + i] * scale; // fft scaling
+ }
+
+ for (i = 0; i < PART_LEN; i++) {
+ e[i] = nearend_ptr[i] - y[i];
+ }
+
+ // Error fft
+ memcpy(aec->eBuf + PART_LEN, e, sizeof(float) * PART_LEN);
+ memset(fft, 0, sizeof(float) * PART_LEN);
+ memcpy(fft + PART_LEN, e, sizeof(float) * PART_LEN);
+ // TODO(bjornv): Change to use TimeToFrequency().
+ aec_rdft_forward_128(fft);
+
+ ef[1][0] = 0;
+ ef[1][PART_LEN] = 0;
+ ef[0][0] = fft[0];
+ ef[0][PART_LEN] = fft[1];
+ for (i = 1; i < PART_LEN; i++) {
+ ef[0][i] = fft[2 * i];
+ ef[1][i] = fft[2 * i + 1];
+ }
+
+ RTC_AEC_DEBUG_RAW_WRITE(aec->e_fft_file,
+ &ef[0][0],
+ sizeof(ef[0][0]) * PART_LEN1 * 2);
+
+ if (aec->metricsMode == 1) {
+ // Note that the first PART_LEN samples in fft (before transformation) are
+ // zero. Hence, the scaling by two in UpdateLevel() should not be
+ // performed. That scaling is taken care of in UpdateMetrics() instead.
+ UpdateLevel(&aec->linoutlevel, ef);
+ }
+
+ // Scale error signal inversely with far power.
+ WebRtcAec_ScaleErrorSignal(aec, ef);
+ WebRtcAec_FilterAdaptation(aec, fft, ef);
+ NonLinearProcessing(aec, output, outputH_ptr);
+
+ if (aec->metricsMode == 1) {
+ // Update power levels and echo metrics
+ UpdateLevel(&aec->farlevel, (float(*)[PART_LEN1])xf_ptr);
+ UpdateLevel(&aec->nearlevel, df);
+ UpdateMetrics(aec);
+ }
+
+ // Store the output block.
+ WebRtc_WriteBuffer(aec->outFrBuf, output, PART_LEN);
+ // For high bands
+ for (i = 0; i < aec->num_bands - 1; ++i) {
+ WebRtc_WriteBuffer(aec->outFrBufH[i], outputH[i], PART_LEN);
+ }
+
+ RTC_AEC_DEBUG_WAV_WRITE(aec->outLinearFile, e, PART_LEN);
+ RTC_AEC_DEBUG_WAV_WRITE(aec->outFile, output, PART_LEN);
+}
+
+AecCore* WebRtcAec_CreateAec() {
+ int i;
+ AecCore* aec = malloc(sizeof(AecCore));
+ if (!aec) {
+ return NULL;
+ }
+
+ aec->nearFrBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(float));
+ if (!aec->nearFrBuf) {
+ WebRtcAec_FreeAec(aec);
+ return NULL;
+ }
+
+ aec->outFrBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(float));
+ if (!aec->outFrBuf) {
+ WebRtcAec_FreeAec(aec);
+ return NULL;
+ }
+
+ for (i = 0; i < NUM_HIGH_BANDS_MAX; ++i) {
+ aec->nearFrBufH[i] = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
+ sizeof(float));
+ if (!aec->nearFrBufH[i]) {
+ WebRtcAec_FreeAec(aec);
+ return NULL;
+ }
+ aec->outFrBufH[i] = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
+ sizeof(float));
+ if (!aec->outFrBufH[i]) {
+ WebRtcAec_FreeAec(aec);
+ return NULL;
+ }
+ }
+
+ // Create far-end buffers.
+ aec->far_buf =
+ WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * 2 * PART_LEN1);
+ if (!aec->far_buf) {
+ WebRtcAec_FreeAec(aec);
+ return NULL;
+ }
+ aec->far_buf_windowed =
+ WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * 2 * PART_LEN1);
+ if (!aec->far_buf_windowed) {
+ WebRtcAec_FreeAec(aec);
+ return NULL;
+ }
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ aec->instance_index = webrtc_aec_instance_count;
+ aec->far_time_buf =
+ WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * PART_LEN);
+ if (!aec->far_time_buf) {
+ WebRtcAec_FreeAec(aec);
+ return NULL;
+ }
+ aec->farFile = aec->nearFile = aec->outFile = aec->outLinearFile = NULL;
+ aec->debug_dump_count = 0;
+#endif
+ aec->delay_estimator_farend =
+ WebRtc_CreateDelayEstimatorFarend(PART_LEN1, kHistorySizeBlocks);
+ if (aec->delay_estimator_farend == NULL) {
+ WebRtcAec_FreeAec(aec);
+ return NULL;
+ }
+ // We create the delay_estimator with the same amount of maximum lookahead as
+ // the delay history size (kHistorySizeBlocks) for symmetry reasons.
+ aec->delay_estimator = WebRtc_CreateDelayEstimator(
+ aec->delay_estimator_farend, kHistorySizeBlocks);
+ if (aec->delay_estimator == NULL) {
+ WebRtcAec_FreeAec(aec);
+ return NULL;
+ }
+#ifdef WEBRTC_ANDROID
+ aec->delay_agnostic_enabled = 1; // DA-AEC enabled by default.
+ // DA-AEC assumes the system is causal from the beginning and will self adjust
+ // the lookahead when shifting is required.
+ WebRtc_set_lookahead(aec->delay_estimator, 0);
+#else
+ aec->delay_agnostic_enabled = 0;
+ WebRtc_set_lookahead(aec->delay_estimator, kLookaheadBlocks);
+#endif
+ aec->extended_filter_enabled = 0;
+
+ // Assembly optimization
+ WebRtcAec_FilterFar = FilterFar;
+ WebRtcAec_ScaleErrorSignal = ScaleErrorSignal;
+ WebRtcAec_FilterAdaptation = FilterAdaptation;
+ WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppress;
+ WebRtcAec_ComfortNoise = ComfortNoise;
+ WebRtcAec_SubbandCoherence = SubbandCoherence;
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+ if (WebRtc_GetCPUInfo(kSSE2)) {
+ WebRtcAec_InitAec_SSE2();
+ }
+#endif
+
+#if defined(MIPS_FPU_LE)
+ WebRtcAec_InitAec_mips();
+#endif
+
+#if defined(WEBRTC_HAS_NEON)
+ WebRtcAec_InitAec_neon();
+#elif defined(WEBRTC_DETECT_NEON)
+ if ((WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON) != 0) {
+ WebRtcAec_InitAec_neon();
+ }
+#endif
+
+ aec_rdft_init();
+
+ return aec;
+}
+
+void WebRtcAec_FreeAec(AecCore* aec) {
+ int i;
+ if (aec == NULL) {
+ return;
+ }
+
+ WebRtc_FreeBuffer(aec->nearFrBuf);
+ WebRtc_FreeBuffer(aec->outFrBuf);
+
+ for (i = 0; i < NUM_HIGH_BANDS_MAX; ++i) {
+ WebRtc_FreeBuffer(aec->nearFrBufH[i]);
+ WebRtc_FreeBuffer(aec->outFrBufH[i]);
+ }
+
+ WebRtc_FreeBuffer(aec->far_buf);
+ WebRtc_FreeBuffer(aec->far_buf_windowed);
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_FreeBuffer(aec->far_time_buf);
+#endif
+ RTC_AEC_DEBUG_WAV_CLOSE(aec->farFile);
+ RTC_AEC_DEBUG_WAV_CLOSE(aec->nearFile);
+ RTC_AEC_DEBUG_WAV_CLOSE(aec->outFile);
+ RTC_AEC_DEBUG_WAV_CLOSE(aec->outLinearFile);
+ RTC_AEC_DEBUG_RAW_CLOSE(aec->e_fft_file);
+
+ WebRtc_FreeDelayEstimator(aec->delay_estimator);
+ WebRtc_FreeDelayEstimatorFarend(aec->delay_estimator_farend);
+
+ free(aec);
+}
+
+int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
+ int i;
+
+ aec->sampFreq = sampFreq;
+
+ if (sampFreq == 8000) {
+ aec->normal_mu = 0.6f;
+ aec->normal_error_threshold = 2e-6f;
+ aec->num_bands = 1;
+ } else {
+ aec->normal_mu = 0.5f;
+ aec->normal_error_threshold = 1.5e-6f;
+ aec->num_bands = (size_t)(sampFreq / 16000);
+ }
+
+ WebRtc_InitBuffer(aec->nearFrBuf);
+ WebRtc_InitBuffer(aec->outFrBuf);
+ for (i = 0; i < NUM_HIGH_BANDS_MAX; ++i) {
+ WebRtc_InitBuffer(aec->nearFrBufH[i]);
+ WebRtc_InitBuffer(aec->outFrBufH[i]);
+ }
+
+ // Initialize far-end buffers.
+ WebRtc_InitBuffer(aec->far_buf);
+ WebRtc_InitBuffer(aec->far_buf_windowed);
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_InitBuffer(aec->far_time_buf);
+ {
+ int process_rate = sampFreq > 16000 ? 16000 : sampFreq;
+ RTC_AEC_DEBUG_WAV_REOPEN("aec_far", aec->instance_index,
+ aec->debug_dump_count, process_rate,
+ &aec->farFile );
+ RTC_AEC_DEBUG_WAV_REOPEN("aec_near", aec->instance_index,
+ aec->debug_dump_count, process_rate,
+ &aec->nearFile);
+ RTC_AEC_DEBUG_WAV_REOPEN("aec_out", aec->instance_index,
+ aec->debug_dump_count, process_rate,
+ &aec->outFile );
+ RTC_AEC_DEBUG_WAV_REOPEN("aec_out_linear", aec->instance_index,
+ aec->debug_dump_count, process_rate,
+ &aec->outLinearFile);
+ }
+
+ RTC_AEC_DEBUG_RAW_OPEN("aec_e_fft",
+ aec->debug_dump_count,
+ &aec->e_fft_file);
+
+ ++aec->debug_dump_count;
+#endif
+ aec->system_delay = 0;
+
+ if (WebRtc_InitDelayEstimatorFarend(aec->delay_estimator_farend) != 0) {
+ return -1;
+ }
+ if (WebRtc_InitDelayEstimator(aec->delay_estimator) != 0) {
+ return -1;
+ }
+ aec->delay_logging_enabled = 0;
+ aec->delay_metrics_delivered = 0;
+ memset(aec->delay_histogram, 0, sizeof(aec->delay_histogram));
+ aec->num_delay_values = 0;
+ aec->delay_median = -1;
+ aec->delay_std = -1;
+ aec->fraction_poor_delays = -1.0f;
+
+ aec->signal_delay_correction = 0;
+ aec->previous_delay = -2; // (-2): Uninitialized.
+ aec->delay_correction_count = 0;
+ aec->shift_offset = kInitialShiftOffset;
+ aec->delay_quality_threshold = kDelayQualityThresholdMin;
+
+ aec->num_partitions = kNormalNumPartitions;
+
+ // Update the delay estimator with filter length. We use half the
+ // |num_partitions| to take the echo path into account. In practice we say
+ // that the echo has a duration of maximum half |num_partitions|, which is not
+ // true, but serves as a crude measure.
+ WebRtc_set_allowed_offset(aec->delay_estimator, aec->num_partitions / 2);
+ // TODO(bjornv): I currently hard coded the enable. Once we've established
+ // that AECM has no performance regression, robust_validation will be enabled
+ // all the time and the APIs to turn it on/off will be removed. Hence, remove
+ // this line then.
+ WebRtc_enable_robust_validation(aec->delay_estimator, 1);
+ aec->frame_count = 0;
+
+ // Default target suppression mode.
+ aec->nlp_mode = 1;
+
+ // Sampling frequency multiplier w.r.t. 8 kHz.
+ // In case of multiple bands we process the lower band in 16 kHz, hence the
+ // multiplier is always 2.
+ if (aec->num_bands > 1) {
+ aec->mult = 2;
+ } else {
+ aec->mult = (short)aec->sampFreq / 8000;
+ }
+
+ aec->farBufWritePos = 0;
+ aec->farBufReadPos = 0;
+
+ aec->inSamples = 0;
+ aec->outSamples = 0;
+ aec->knownDelay = 0;
+
+ // Initialize buffers
+ memset(aec->dBuf, 0, sizeof(aec->dBuf));
+ memset(aec->eBuf, 0, sizeof(aec->eBuf));
+ // For H bands
+ for (i = 0; i < NUM_HIGH_BANDS_MAX; ++i) {
+ memset(aec->dBufH[i], 0, sizeof(aec->dBufH[i]));
+ }
+
+ memset(aec->xPow, 0, sizeof(aec->xPow));
+ memset(aec->dPow, 0, sizeof(aec->dPow));
+ memset(aec->dInitMinPow, 0, sizeof(aec->dInitMinPow));
+ aec->noisePow = aec->dInitMinPow;
+ aec->noiseEstCtr = 0;
+
+ // Initial comfort noise power
+ for (i = 0; i < PART_LEN1; i++) {
+ aec->dMinPow[i] = 1.0e6f;
+ }
+
+ // Holds the last block written to
+ aec->xfBufBlockPos = 0;
+ // TODO: Investigate need for these initializations. Deleting them doesn't
+ // change the output at all and yields 0.4% overall speedup.
+ memset(aec->xfBuf, 0, sizeof(complex_t) * kExtendedNumPartitions * PART_LEN1);
+ memset(aec->wfBuf, 0, sizeof(complex_t) * kExtendedNumPartitions * PART_LEN1);
+ memset(aec->sde, 0, sizeof(complex_t) * PART_LEN1);
+ memset(aec->sxd, 0, sizeof(complex_t) * PART_LEN1);
+ memset(
+ aec->xfwBuf, 0, sizeof(complex_t) * kExtendedNumPartitions * PART_LEN1);
+ memset(aec->se, 0, sizeof(float) * PART_LEN1);
+
+ // To prevent numerical instability in the first block.
+ for (i = 0; i < PART_LEN1; i++) {
+ aec->sd[i] = 1;
+ }
+ for (i = 0; i < PART_LEN1; i++) {
+ aec->sx[i] = 1;
+ }
+
+ memset(aec->hNs, 0, sizeof(aec->hNs));
+ memset(aec->outBuf, 0, sizeof(float) * PART_LEN);
+
+ aec->hNlFbMin = 1;
+ aec->hNlFbLocalMin = 1;
+ aec->hNlXdAvgMin = 1;
+ aec->hNlNewMin = 0;
+ aec->hNlMinCtr = 0;
+ aec->overDrive = 2;
+ aec->overDriveSm = 2;
+ aec->delayIdx = 0;
+ aec->stNearState = 0;
+ aec->echoState = 0;
+ aec->divergeState = 0;
+
+ aec->seed = 777;
+ aec->delayEstCtr = 0;
+
+ // Metrics disabled by default
+ aec->metricsMode = 0;
+ InitMetrics(aec);
+
+ return 0;
+}
+
+void WebRtcAec_BufferFarendPartition(AecCore* aec, const float* farend) {
+ float fft[PART_LEN2];
+ float xf[2][PART_LEN1];
+
+ // Check if the buffer is full, and in that case flush the oldest data.
+ if (WebRtc_available_write(aec->far_buf) < 1) {
+ WebRtcAec_MoveFarReadPtr(aec, 1);
+ }
+ // Convert far-end partition to the frequency domain without windowing.
+ memcpy(fft, farend, sizeof(float) * PART_LEN2);
+ TimeToFrequency(fft, xf, 0);
+ WebRtc_WriteBuffer(aec->far_buf, &xf[0][0], 1);
+
+ // Convert far-end partition to the frequency domain with windowing.
+ memcpy(fft, farend, sizeof(float) * PART_LEN2);
+ TimeToFrequency(fft, xf, 1);
+ WebRtc_WriteBuffer(aec->far_buf_windowed, &xf[0][0], 1);
+}
+
+int WebRtcAec_MoveFarReadPtr(AecCore* aec, int elements) {
+ int elements_moved = MoveFarReadPtrWithoutSystemDelayUpdate(aec, elements);
+ aec->system_delay -= elements_moved * PART_LEN;
+ return elements_moved;
+}
+
+void WebRtcAec_ProcessFrames(AecCore* aec,
+ const float* const* nearend,
+ size_t num_bands,
+ size_t num_samples,
+ int knownDelay,
+ float* const* out) {
+ size_t i, j;
+ int out_elements = 0;
+
+ aec->frame_count++;
+ // For each frame the process is as follows:
+ // 1) If the system_delay indicates on being too small for processing a
+ // frame we stuff the buffer with enough data for 10 ms.
+ // 2 a) Adjust the buffer to the system delay, by moving the read pointer.
+ // b) Apply signal based delay correction, if we have detected poor AEC
+ // performance.
+ // 3) TODO(bjornv): Investigate if we need to add this:
+ // If we can't move read pointer due to buffer size limitations we
+ // flush/stuff the buffer.
+ // 4) Process as many partitions as possible.
+ // 5) Update the |system_delay| with respect to a full frame of FRAME_LEN
+ // samples. Even though we will have data left to process (we work with
+ // partitions) we consider updating a whole frame, since that's the
+ // amount of data we input and output in audio_processing.
+ // 6) Update the outputs.
+
+ // The AEC has two different delay estimation algorithms built in. The
+ // first relies on delay input values from the user and the amount of
+ // shifted buffer elements is controlled by |knownDelay|. This delay will
+ // give a guess on how much we need to shift far-end buffers to align with
+ // the near-end signal. The other delay estimation algorithm uses the
+ // far- and near-end signals to find the offset between them. This one
+ // (called "signal delay") is then used to fine tune the alignment, or
+ // simply compensate for errors in the system based one.
+ // Note that the two algorithms operate independently. Currently, we only
+ // allow one algorithm to be turned on.
+
+ assert(aec->num_bands == num_bands);
+
+ for (j = 0; j < num_samples; j+= FRAME_LEN) {
+ // TODO(bjornv): Change the near-end buffer handling to be the same as for
+ // far-end, that is, with a near_pre_buf.
+ // Buffer the near-end frame.
+ WebRtc_WriteBuffer(aec->nearFrBuf, &nearend[0][j], FRAME_LEN);
+ // For H band
+ for (i = 1; i < num_bands; ++i) {
+ WebRtc_WriteBuffer(aec->nearFrBufH[i - 1], &nearend[i][j], FRAME_LEN);
+ }
+
+ // 1) At most we process |aec->mult|+1 partitions in 10 ms. Make sure we
+ // have enough far-end data for that by stuffing the buffer if the
+ // |system_delay| indicates others.
+ if (aec->system_delay < FRAME_LEN) {
+ // We don't have enough data so we rewind 10 ms.
+ WebRtcAec_MoveFarReadPtr(aec, -(aec->mult + 1));
+ }
+
+ if (!aec->delay_agnostic_enabled) {
+ // 2 a) Compensate for a possible change in the system delay.
+
+ // TODO(bjornv): Investigate how we should round the delay difference;
+ // right now we know that incoming |knownDelay| is underestimated when
+ // it's less than |aec->knownDelay|. We therefore, round (-32) in that
+ // direction. In the other direction, we don't have this situation, but
+ // might flush one partition too little. This can cause non-causality,
+ // which should be investigated. Maybe, allow for a non-symmetric
+ // rounding, like -16.
+ int move_elements = (aec->knownDelay - knownDelay - 32) / PART_LEN;
+ int moved_elements =
+ MoveFarReadPtrWithoutSystemDelayUpdate(aec, move_elements);
+ aec->knownDelay -= moved_elements * PART_LEN;
+ } else {
+ // 2 b) Apply signal based delay correction.
+ int move_elements = SignalBasedDelayCorrection(aec);
+ int moved_elements =
+ MoveFarReadPtrWithoutSystemDelayUpdate(aec, move_elements);
+ int far_near_buffer_diff = WebRtc_available_read(aec->far_buf) -
+ WebRtc_available_read(aec->nearFrBuf) / PART_LEN;
+ WebRtc_SoftResetDelayEstimator(aec->delay_estimator, moved_elements);
+ WebRtc_SoftResetDelayEstimatorFarend(aec->delay_estimator_farend,
+ moved_elements);
+ aec->signal_delay_correction += moved_elements;
+ // If we rely on reported system delay values only, a buffer underrun here
+ // can never occur since we've taken care of that in 1) above. Here, we
+ // apply signal based delay correction and can therefore end up with
+ // buffer underruns since the delay estimation can be wrong. We therefore
+ // stuff the buffer with enough elements if needed.
+ if (far_near_buffer_diff < 0) {
+ WebRtcAec_MoveFarReadPtr(aec, far_near_buffer_diff);
+ }
+ }
+
+ // 4) Process as many blocks as possible.
+ while (WebRtc_available_read(aec->nearFrBuf) >= PART_LEN) {
+ ProcessBlock(aec);
+ }
+
+ // 5) Update system delay with respect to the entire frame.
+ aec->system_delay -= FRAME_LEN;
+
+ // 6) Update output frame.
+ // Stuff the out buffer if we have less than a frame to output.
+ // This should only happen for the first frame.
+ out_elements = (int)WebRtc_available_read(aec->outFrBuf);
+ if (out_elements < FRAME_LEN) {
+ WebRtc_MoveReadPtr(aec->outFrBuf, out_elements - FRAME_LEN);
+ for (i = 0; i < num_bands - 1; ++i) {
+ WebRtc_MoveReadPtr(aec->outFrBufH[i], out_elements - FRAME_LEN);
+ }
+ }
+ // Obtain an output frame.
+ WebRtc_ReadBuffer(aec->outFrBuf, NULL, &out[0][j], FRAME_LEN);
+ // For H bands.
+ for (i = 1; i < num_bands; ++i) {
+ WebRtc_ReadBuffer(aec->outFrBufH[i - 1], NULL, &out[i][j], FRAME_LEN);
+ }
+ }
+}
+
+int WebRtcAec_GetDelayMetricsCore(AecCore* self, int* median, int* std,
+ float* fraction_poor_delays) {
+ assert(self != NULL);
+ assert(median != NULL);
+ assert(std != NULL);
+
+ if (self->delay_logging_enabled == 0) {
+ // Logging disabled.
+ return -1;
+ }
+
+ if (self->delay_metrics_delivered == 0) {
+ UpdateDelayMetrics(self);
+ self->delay_metrics_delivered = 1;
+ }
+ *median = self->delay_median;
+ *std = self->delay_std;
+ *fraction_poor_delays = self->fraction_poor_delays;
+
+ return 0;
+}
+
+int WebRtcAec_echo_state(AecCore* self) { return self->echoState; }
+
+void WebRtcAec_GetEchoStats(AecCore* self,
+ Stats* erl,
+ Stats* erle,
+ Stats* a_nlp) {
+ assert(erl != NULL);
+ assert(erle != NULL);
+ assert(a_nlp != NULL);
+ *erl = self->erl;
+ *erle = self->erle;
+ *a_nlp = self->aNlp;
+}
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+void* WebRtcAec_far_time_buf(AecCore* self) { return self->far_time_buf; }
+#endif
+
+void WebRtcAec_SetConfigCore(AecCore* self,
+ int nlp_mode,
+ int metrics_mode,
+ int delay_logging) {
+ assert(nlp_mode >= 0 && nlp_mode < 3);
+ self->nlp_mode = nlp_mode;
+ self->metricsMode = metrics_mode;
+ if (self->metricsMode) {
+ InitMetrics(self);
+ }
+ // Turn on delay logging if it is either set explicitly or if delay agnostic
+ // AEC is enabled (which requires delay estimates).
+ self->delay_logging_enabled = delay_logging || self->delay_agnostic_enabled;
+ if (self->delay_logging_enabled) {
+ memset(self->delay_histogram, 0, sizeof(self->delay_histogram));
+ }
+}
+
+void WebRtcAec_enable_delay_agnostic(AecCore* self, int enable) {
+ self->delay_agnostic_enabled = enable;
+}
+
+int WebRtcAec_delay_agnostic_enabled(AecCore* self) {
+ return self->delay_agnostic_enabled;
+}
+
+void WebRtcAec_enable_extended_filter(AecCore* self, int enable) {
+ self->extended_filter_enabled = enable;
+ self->num_partitions = enable ? kExtendedNumPartitions : kNormalNumPartitions;
+ // Update the delay estimator with filter length. See InitAEC() for details.
+ WebRtc_set_allowed_offset(self->delay_estimator, self->num_partitions / 2);
+}
+
+int WebRtcAec_extended_filter_enabled(AecCore* self) {
+ return self->extended_filter_enabled;
+}
+
+int WebRtcAec_system_delay(AecCore* self) { return self->system_delay; }
+
+void WebRtcAec_SetSystemDelay(AecCore* self, int delay) {
+ assert(delay >= 0);
+ self->system_delay = delay;
+}
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