/*
 *  Copyright (c) 2011 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 "aec_core.h"

#include <assert.h>
#include <math.h>
#include <stddef.h>  // size_t
#include <stdlib.h>
#include <string.h>

#include "aec_rdft.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "delay_estimator_wrapper.h"
#include "ring_buffer.h"
#include "system_wrappers/interface/cpu_features_wrapper.h"
#include "typedefs.h"

// Buffer size (samples)
static const size_t kBufSizePartitions = 250;  // 1 second of audio in 16 kHz.

// Noise suppression
static const int converged = 250;

// Metrics
static const int subCountLen = 4;
static const int countLen = 50;

// 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);
static const float 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);
const float 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);
const float 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
};

// "Private" function prototypes.
static void ProcessBlock(aec_t* aec);

static void NonLinearProcessing(aec_t *aec, short *output, short *outputH);

static void GetHighbandGain(const float *lambda, float *nlpGainHband);

// Comfort_noise also computes noise for H band returned in comfortNoiseHband
static void ComfortNoise(aec_t *aec, float efw[2][PART_LEN1],
                                  complex_t *comfortNoiseHband,
                                  const float *noisePow, const float *lambda);

static void WebRtcAec_InitLevel(power_level_t *level);
static void WebRtcAec_InitStats(stats_t *stats);
static void UpdateLevel(power_level_t* level, float in[2][PART_LEN1]);
static void UpdateMetrics(aec_t *aec);
// Convert from time domain to frequency domain. Note that |time_data| are
// overwritten.
static void TimeToFrequency(float time_data[PART_LEN2],
                            float freq_data[2][PART_LEN1],
                            int window);

__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);
}

int WebRtcAec_CreateAec(aec_t **aecInst)
{
    aec_t *aec = malloc(sizeof(aec_t));
    *aecInst = aec;
    if (aec == NULL) {
        return -1;
    }

    if (WebRtc_CreateBuffer(&aec->nearFrBuf,
                            FRAME_LEN + PART_LEN,
                            sizeof(int16_t)) == -1) {
        WebRtcAec_FreeAec(aec);
        aec = NULL;
        return -1;
    }

    if (WebRtc_CreateBuffer(&aec->outFrBuf,
                            FRAME_LEN + PART_LEN,
                            sizeof(int16_t)) == -1) {
        WebRtcAec_FreeAec(aec);
        aec = NULL;
        return -1;
    }

    if (WebRtc_CreateBuffer(&aec->nearFrBufH,
                            FRAME_LEN + PART_LEN,
                            sizeof(int16_t)) == -1) {
        WebRtcAec_FreeAec(aec);
        aec = NULL;
        return -1;
    }

    if (WebRtc_CreateBuffer(&aec->outFrBufH,
                            FRAME_LEN + PART_LEN,
                            sizeof(int16_t)) == -1) {
        WebRtcAec_FreeAec(aec);
        aec = NULL;
        return -1;
    }

    // Create far-end buffers.
    if (WebRtc_CreateBuffer(&aec->far_buf, kBufSizePartitions,
                            sizeof(float) * 2 * PART_LEN1) == -1) {
        WebRtcAec_FreeAec(aec);
        aec = NULL;
        return -1;
    }
    if (WebRtc_CreateBuffer(&aec->far_buf_windowed, kBufSizePartitions,
                            sizeof(float) * 2 * PART_LEN1) == -1) {
        WebRtcAec_FreeAec(aec);
        aec = NULL;
        return -1;
    }
#ifdef WEBRTC_AEC_DEBUG_DUMP
    if (WebRtc_CreateBuffer(&aec->far_time_buf, kBufSizePartitions,
                            sizeof(int16_t) * PART_LEN) == -1) {
        WebRtcAec_FreeAec(aec);
        aec = NULL;
        return -1;
    }
#endif
    if (WebRtc_CreateDelayEstimator(&aec->delay_estimator,
                                    PART_LEN1,
                                    kMaxDelayBlocks,
                                    kLookaheadBlocks) == -1) {
      WebRtcAec_FreeAec(aec);
      aec = NULL;
      return -1;
    }

    return 0;
}

int WebRtcAec_FreeAec(aec_t *aec)
{
    if (aec == NULL) {
        return -1;
    }

    WebRtc_FreeBuffer(aec->nearFrBuf);
    WebRtc_FreeBuffer(aec->outFrBuf);

    WebRtc_FreeBuffer(aec->nearFrBufH);
    WebRtc_FreeBuffer(aec->outFrBufH);

    WebRtc_FreeBuffer(aec->far_buf);
    WebRtc_FreeBuffer(aec->far_buf_windowed);
#ifdef WEBRTC_AEC_DEBUG_DUMP
    WebRtc_FreeBuffer(aec->far_time_buf);
#endif
    WebRtc_FreeDelayEstimator(aec->delay_estimator);

    free(aec);
    return 0;
}

static void FilterFar(aec_t *aec, float yf[2][PART_LEN1])
{
  int i;
  for (i = 0; i < NR_PART; i++) {
    int j;
    int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
    int pos = i * PART_LEN1;
    // Check for wrap
    if (i + aec->xfBufBlockPos >= NR_PART) {
      xPos -= NR_PART*(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(aec_t *aec, float ef[2][PART_LEN1])
{
  int i;
  float absEf;
  for (i = 0; i < (PART_LEN1); i++) {
    ef[0][i] /= (aec->xPow[i] + 1e-10f);
    ef[1][i] /= (aec->xPow[i] + 1e-10f);
    absEf = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);

    if (absEf > aec->errThresh) {
      absEf = aec->errThresh / (absEf + 1e-10f);
      ef[0][i] *= absEf;
      ef[1][i] *= absEf;
    }

    // Stepsize factor
    ef[0][i] *= aec->mu;
    ef[1][i] *= aec->mu;
  }
}

// Time-unconstrined filter adaptation.
// TODO(andrew): consider for a low-complexity mode.
//static void FilterAdaptationUnconstrained(aec_t *aec, float *fft,
//                                          float ef[2][PART_LEN1]) {
//  int i, j;
//  for (i = 0; i < NR_PART; i++) {
//    int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
//    int pos;
//    // Check for wrap
//    if (i + aec->xfBufBlockPos >= NR_PART) {
//      xPos -= NR_PART * PART_LEN1;
//    }
//
//    pos = i * PART_LEN1;
//
//    for (j = 0; j < PART_LEN1; j++) {
//      aec->wfBuf[pos + j][0] += MulRe(aec->xfBuf[xPos + j][0],
//                                      -aec->xfBuf[xPos + j][1],
//                                      ef[j][0], ef[j][1]);
//      aec->wfBuf[pos + j][1] += MulIm(aec->xfBuf[xPos + j][0],
//                                      -aec->xfBuf[xPos + j][1],
//                                      ef[j][0], ef[j][1]);
//    }
//  }
//}

static void FilterAdaptation(aec_t *aec, float *fft, float ef[2][PART_LEN1]) {
  int i, j;
  for (i = 0; i < NR_PART; i++) {
    int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
    int pos;
    // Check for wrap
    if (i + aec->xfBufBlockPos >= NR_PART) {
      xPos -= NR_PART * 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(aec_t *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;
  }
}

WebRtcAec_FilterFar_t WebRtcAec_FilterFar;
WebRtcAec_ScaleErrorSignal_t WebRtcAec_ScaleErrorSignal;
WebRtcAec_FilterAdaptation_t WebRtcAec_FilterAdaptation;
WebRtcAec_OverdriveAndSuppress_t WebRtcAec_OverdriveAndSuppress;

int WebRtcAec_InitAec(aec_t *aec, int sampFreq)
{
    int i;

    aec->sampFreq = sampFreq;

    if (sampFreq == 8000) {
        aec->mu = 0.6f;
        aec->errThresh = 2e-6f;
    }
    else {
        aec->mu = 0.5f;
        aec->errThresh = 1.5e-6f;
    }

    if (WebRtc_InitBuffer(aec->nearFrBuf) == -1) {
        return -1;
    }

    if (WebRtc_InitBuffer(aec->outFrBuf) == -1) {
        return -1;
    }

    if (WebRtc_InitBuffer(aec->nearFrBufH) == -1) {
        return -1;
    }

    if (WebRtc_InitBuffer(aec->outFrBufH) == -1) {
        return -1;
    }

    // Initialize far-end buffers.
    if (WebRtc_InitBuffer(aec->far_buf) == -1) {
        return -1;
    }
    if (WebRtc_InitBuffer(aec->far_buf_windowed) == -1) {
        return -1;
    }
#ifdef WEBRTC_AEC_DEBUG_DUMP
    if (WebRtc_InitBuffer(aec->far_time_buf) == -1) {
        return -1;
    }
#endif
    aec->system_delay = 0;

    if (WebRtc_InitDelayEstimator(aec->delay_estimator) != 0) {
      return -1;
    }
    aec->delay_logging_enabled = 0;
    memset(aec->delay_histogram, 0, sizeof(aec->delay_histogram));

    // Default target suppression level
    aec->targetSupp = -11.5;
    aec->minOverDrive = 2.0;

    // Sampling frequency multiplier
    // SWB is processed as 160 frame size
    if (aec->sampFreq == 32000) {
      aec->mult = (short)aec->sampFreq / 16000;
    }
    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 band
    memset(aec->dBufH, 0, sizeof(aec->dBufH));

    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) * NR_PART * PART_LEN1);
    memset(aec->wfBuf, 0, sizeof(complex_t) * NR_PART * 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) * NR_PART * 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;
    WebRtcAec_InitMetrics(aec);

    // Assembly optimization
    WebRtcAec_FilterFar = FilterFar;
    WebRtcAec_ScaleErrorSignal = ScaleErrorSignal;
    WebRtcAec_FilterAdaptation = FilterAdaptation;
    WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppress;

#if defined(WEBRTC_ARCH_X86_FAMILY)
    if (WebRtc_GetCPUInfo(kSSE2)) {
      WebRtcAec_InitAec_SSE2();
    }
#endif

    aec_rdft_init();

    return 0;
}

void WebRtcAec_InitMetrics(aec_t *aec)
{
    aec->stateCounter = 0;
    WebRtcAec_InitLevel(&aec->farlevel);
    WebRtcAec_InitLevel(&aec->nearlevel);
    WebRtcAec_InitLevel(&aec->linoutlevel);
    WebRtcAec_InitLevel(&aec->nlpoutlevel);

    WebRtcAec_InitStats(&aec->erl);
    WebRtcAec_InitStats(&aec->erle);
    WebRtcAec_InitStats(&aec->aNlp);
    WebRtcAec_InitStats(&aec->rerl);
}


void WebRtcAec_BufferFarendPartition(aec_t *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) {
    WebRtc_MoveReadPtr(aec->far_buf, 1);
    WebRtc_MoveReadPtr(aec->far_buf_windowed, 1);
    aec->system_delay -= PART_LEN;
#ifdef WEBRTC_AEC_DEBUG_DUMP
    WebRtc_MoveReadPtr(aec->far_time_buf, 1);
#endif
  }
  // 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);
}

void WebRtcAec_ProcessFrame(aec_t *aec,
                            const short *nearend,
                            const short *nearendH,
                            int knownDelay)
{
    // 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) Adjust the buffer to the system delay, by moving the read pointer.
    // 3) 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.

    // 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 = 0;

    // 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, FRAME_LEN);
    // For H band
    if (aec->sampFreq == 32000) {
        WebRtc_WriteBuffer(aec->nearFrBufH, nearendH, 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.
      WebRtc_MoveReadPtr(aec->far_buf_windowed, -(aec->mult + 1));
      aec->system_delay -= WebRtc_MoveReadPtr(aec->far_buf, -(aec->mult + 1)) *
          PART_LEN;
#ifdef WEBRTC_AEC_DEBUG_DUMP
      WebRtc_MoveReadPtr(aec->far_time_buf, -(aec->mult + 1));
#endif
    }

    // 2) Compensate for a possible change in the system delay.

    WebRtc_MoveReadPtr(aec->far_buf_windowed, move_elements);
    moved_elements = WebRtc_MoveReadPtr(aec->far_buf, move_elements);
    aec->knownDelay -= moved_elements * PART_LEN;
#ifdef WEBRTC_AEC_DEBUG_DUMP
    WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
#endif

    // 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;
}

static void ProcessBlock(aec_t* aec) {
    int i;
    float d[PART_LEN], y[PART_LEN], e[PART_LEN], dH[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};

    int16_t nearend[PART_LEN];
    int16_t* nearend_ptr = NULL;
    int16_t output[PART_LEN];
    int16_t outputH[PART_LEN];

    float* xf_ptr = NULL;

    memset(dH, 0, sizeof(dH));
    if (aec->sampFreq == 32000) {
      // Get the upper band first so we can reuse |nearend|.
      WebRtc_ReadBuffer(aec->nearFrBufH,
                        (void**) &nearend_ptr,
                        nearend,
                        PART_LEN);
      for (i = 0; i < PART_LEN; i++) {
          dH[i] = (float) (nearend_ptr[i]);
      }
      memcpy(aec->dBufH + PART_LEN, dH, sizeof(float) * PART_LEN);
    }
    WebRtc_ReadBuffer(aec->nearFrBuf, (void**) &nearend_ptr, nearend, PART_LEN);

    // ---------- Ooura fft ----------
    // Concatenate old and new nearend blocks.
    for (i = 0; i < PART_LEN; i++) {
        d[i] = (float) (nearend_ptr[i]);
    }
    memcpy(aec->dBuf + PART_LEN, d, sizeof(float) * PART_LEN);

#ifdef WEBRTC_AEC_DEBUG_DUMP
    {
        int16_t farend[PART_LEN];
        int16_t* farend_ptr = NULL;
        WebRtc_ReadBuffer(aec->far_time_buf, (void**) &farend_ptr, farend, 1);
        fwrite(farend_ptr, sizeof(int16_t), PART_LEN, aec->farFile);
        fwrite(nearend_ptr, sizeof(int16_t), PART_LEN, aec->nearFile);
    }
#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] * NR_PART * 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) {
      int delay_estimate = 0;
      // Estimate the delay
      delay_estimate = WebRtc_DelayEstimatorProcessFloat(aec->delay_estimator,
                                                         abs_far_spectrum,
                                                         abs_near_spectrum,
                                                         PART_LEN1);
      if (delay_estimate >= 0) {
        // Update delay estimate buffer.
        aec->delay_histogram[delay_estimate]++;
      }
    }

    // Update the xfBuf block position.
    aec->xfBufBlockPos--;
    if (aec->xfBufBlockPos == -1) {
        aec->xfBufBlockPos = NR_PART - 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], 0, sizeof(float) * (PART_LEN1 * 2));

    // 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] = d[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];
    }

    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);

    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 H band
    if (aec->sampFreq == 32000) {
        WebRtc_WriteBuffer(aec->outFrBufH, outputH, PART_LEN);
    }

#ifdef WEBRTC_AEC_DEBUG_DUMP
    {
        int16_t eInt16[PART_LEN];
        for (i = 0; i < PART_LEN; i++) {
            eInt16[i] = (int16_t)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, e[i],
                WEBRTC_SPL_WORD16_MIN);
        }

        fwrite(eInt16, sizeof(int16_t), PART_LEN, aec->outLinearFile);
        fwrite(output, sizeof(int16_t), PART_LEN, aec->outFile);
    }
#endif
}

static void NonLinearProcessing(aec_t *aec, short *output, short *outputH)
{
    float efw[2][PART_LEN1], dfw[2][PART_LEN1], xfw[2][PART_LEN1];
    complex_t comfortNoiseHband[PART_LEN1];
    float fft[PART_LEN2];
    float scale, dtmp;
    float nlpGainHband;
    int i, j, pos;

    // Coherence and non-linear filter
    float cohde[PART_LEN1], cohxd[PART_LEN1];
    float hNlDeAvg, hNlXdAvg;
    float hNl[PART_LEN1];
    float hNlPref[PREF_BAND_SIZE];
    float hNlFb = 0, hNlFbLow = 0;
    const float prefBandQuant = 0.75f, prefBandQuantLow = 0.5f;
    const int prefBandSize = PREF_BAND_SIZE / aec->mult;
    const int minPrefBand = 4 / aec->mult;

    // Near and error power sums
    float sdSum = 0, seSum = 0;

    // Power estimate smoothing coefficients
    const float gCoh[2][2] = {{0.9f, 0.1f}, {0.93f, 0.07f}};
    const float *ptrGCoh = gCoh[aec->mult - 1];

    // Filter energy
    float wfEnMax = 0, wfEn = 0;
    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;

    // Measure energy in each filter partition to determine delay.
    // TODO: Spread by computing one partition per block?
    if (aec->delayEstCtr == 0) {
        wfEnMax = 0;
        aec->delayIdx = 0;
        for (i = 0; i < NR_PART; i++) {
            pos = i * PART_LEN1;
            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;
                aec->delayIdx = i;
            }
        }
    }

    // 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);

    // Use delayed far.
    memcpy(xfw, aec->xfwBuf + aec->delayIdx * PART_LEN1, sizeof(xfw));

    // Windowed near fft
    for (i = 0; i < PART_LEN; i++) {
        fft[i] = aec->dBuf[i] * sqrtHanning[i];
        fft[PART_LEN + i] = aec->dBuf[PART_LEN + i] * sqrtHanning[PART_LEN - i];
    }
    aec_rdft_forward_128(fft);

    dfw[1][0] = 0;
    dfw[1][PART_LEN] = 0;
    dfw[0][0] = fft[0];
    dfw[0][PART_LEN] = fft[1];
    for (i = 1; i < PART_LEN; i++) {
        dfw[0][i] = fft[2 * i];
        dfw[1][i] = fft[2 * i + 1];
    }

    // Windowed error fft
    for (i = 0; i < PART_LEN; i++) {
        fft[i] = aec->eBuf[i] * sqrtHanning[i];
        fft[PART_LEN + i] = aec->eBuf[PART_LEN + i] * sqrtHanning[PART_LEN - i];
    }
    aec_rdft_forward_128(fft);
    efw[1][0] = 0;
    efw[1][PART_LEN] = 0;
    efw[0][0] = fft[0];
    efw[0][PART_LEN] = fft[1];
    for (i = 1; i < PART_LEN; i++) {
        efw[0][i] = fft[2 * i];
        efw[1][i] = fft[2 * i + 1];
    }

    // Smoothed PSD
    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: 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], 15);

        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.
    if (aec->divergeState == 0) {
        if (seSum > sdSum) {
            aec->divergeState = 1;
        }
    }
    else {
        if (seSum * 1.05f < sdSum) {
            aec->divergeState = 0;
        }
    }

    if (aec->divergeState == 1) {
        memcpy(efw, dfw, sizeof(efw));
    }

    // Reset if error is significantly larger than nearend (13 dB).
    if (seSum > (19.95f * sdSum)) {
        memset(aec->wfBuf, 0, sizeof(aec->wfBuf));
    }

    // 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);
    }

    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 = aec->minOverDrive;

        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(aec->targetSupp /
            ((float)log(aec->hNlFbMin + 1e-10f) + 1e-10f), aec->minOverDrive);
    }

    // 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.
    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]*sqrtHanning[i] + aec->outBuf[i];

        // Saturation protection
        output[i] = (short)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, fft[i],
            WEBRTC_SPL_WORD16_MIN);

        fft[PART_LEN + i] *= scale; // fft scaling
        aec->outBuf[i] = fft[PART_LEN + i] * sqrtHanning[PART_LEN - i];
    }

    // For H band
    if (aec->sampFreq == 32000) {

        // 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 (i = 0; i < PART_LEN; i++) {
            dtmp = (float)aec->dBufH[i];
            dtmp = (float)dtmp * nlpGainHband; // for variable gain

            // add some comfort noise where Hband is attenuated
            if (flagHbandCn == 1) {
                fft[i] *= scale; // fft scaling
                dtmp += cnScaleHband * fft[i];
            }

            // Saturation protection
            outputH[i] = (short)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
    if (aec->sampFreq == 32000) {
        memcpy(aec->dBufH, aec->dBufH + PART_LEN, sizeof(float) * PART_LEN);
    }

    memmove(aec->xfwBuf + PART_LEN1, aec->xfwBuf, sizeof(aec->xfwBuf) -
        sizeof(complex_t) * PART_LEN1);
}

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(aec_t *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;
    WebRtc_Word16 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 * (float)cos(tmp);
        u[i][1] = -noise * (float)sin(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->sampFreq == 32000 && 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 WebRtcAec_InitLevel(power_level_t *level)
{
    const float bigFloat = 1E17f;

    level->averagelevel = 0;
    level->framelevel = 0;
    level->minlevel = bigFloat;
    level->frsum = 0;
    level->sfrsum = 0;
    level->frcounter = 0;
    level->sfrcounter = 0;
}

static void WebRtcAec_InitStats(stats_t *stats)
{
    stats->instant = offsetLevel;
    stats->average = offsetLevel;
    stats->max = offsetLevel;
    stats->min = offsetLevel * (-1);
    stats->sum = 0;
    stats->hisum = 0;
    stats->himean = offsetLevel;
    stats->counter = 0;
    stats->hicounter = 0;
}

static void UpdateLevel(power_level_t* 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(aec_t *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 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] *= sqrtHanning[i];
      time_data[PART_LEN + i] *= 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];
  }
}