mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
207 lines
5.2 KiB
C++
207 lines
5.2 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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/* Resamples a signal to an arbitrary rate. Used by the AEC to compensate for
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* clock skew by resampling the farend signal.
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*/
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#include "modules/audio_processing/aec/aec_resampler.h"
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#include <stdlib.h>
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#include <string.h>
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#include "modules/audio_processing/aec/aec_core.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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enum { kEstimateLengthFrames = 400 };
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typedef struct {
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float buffer[kResamplerBufferSize];
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float position;
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int deviceSampleRateHz;
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int skewData[kEstimateLengthFrames];
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int skewDataIndex;
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float skewEstimate;
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} AecResampler;
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static int EstimateSkew(const int* rawSkew,
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int size,
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int deviceSampleRateHz,
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float* skewEst);
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void* WebRtcAec_CreateResampler() {
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return malloc(sizeof(AecResampler));
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}
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int WebRtcAec_InitResampler(void* resampInst, int deviceSampleRateHz) {
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AecResampler* obj = static_cast<AecResampler*>(resampInst);
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memset(obj->buffer, 0, sizeof(obj->buffer));
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obj->position = 0.0;
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obj->deviceSampleRateHz = deviceSampleRateHz;
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memset(obj->skewData, 0, sizeof(obj->skewData));
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obj->skewDataIndex = 0;
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obj->skewEstimate = 0.0;
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return 0;
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}
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void WebRtcAec_FreeResampler(void* resampInst) {
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AecResampler* obj = static_cast<AecResampler*>(resampInst);
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free(obj);
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}
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void WebRtcAec_ResampleLinear(void* resampInst,
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const float* inspeech,
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size_t size,
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float skew,
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float* outspeech,
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size_t* size_out) {
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AecResampler* obj = static_cast<AecResampler*>(resampInst);
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float* y;
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float be, tnew;
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size_t tn, mm;
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RTC_DCHECK_LE(size, 2 * FRAME_LEN);
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RTC_DCHECK(resampInst);
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RTC_DCHECK(inspeech);
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RTC_DCHECK(outspeech);
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RTC_DCHECK(size_out);
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// Add new frame data in lookahead
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memcpy(&obj->buffer[FRAME_LEN + kResamplingDelay], inspeech,
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size * sizeof(inspeech[0]));
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// Sample rate ratio
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be = 1 + skew;
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// Loop over input frame
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mm = 0;
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y = &obj->buffer[FRAME_LEN]; // Point at current frame
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tnew = be * mm + obj->position;
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tn = (size_t)tnew;
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while (tn < size) {
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// Interpolation
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outspeech[mm] = y[tn] + (tnew - tn) * (y[tn + 1] - y[tn]);
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mm++;
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tnew = be * mm + obj->position;
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tn = static_cast<int>(tnew);
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}
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*size_out = mm;
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obj->position += (*size_out) * be - size;
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// Shift buffer
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memmove(obj->buffer, &obj->buffer[size],
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(kResamplerBufferSize - size) * sizeof(obj->buffer[0]));
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}
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int WebRtcAec_GetSkew(void* resampInst, int rawSkew, float* skewEst) {
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AecResampler* obj = static_cast<AecResampler*>(resampInst);
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int err = 0;
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if (obj->skewDataIndex < kEstimateLengthFrames) {
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obj->skewData[obj->skewDataIndex] = rawSkew;
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obj->skewDataIndex++;
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} else if (obj->skewDataIndex == kEstimateLengthFrames) {
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err = EstimateSkew(obj->skewData, kEstimateLengthFrames,
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obj->deviceSampleRateHz, skewEst);
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obj->skewEstimate = *skewEst;
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obj->skewDataIndex++;
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} else {
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*skewEst = obj->skewEstimate;
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}
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return err;
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}
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int EstimateSkew(const int* rawSkew,
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int size,
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int deviceSampleRateHz,
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float* skewEst) {
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const int absLimitOuter = static_cast<int>(0.04f * deviceSampleRateHz);
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const int absLimitInner = static_cast<int>(0.0025f * deviceSampleRateHz);
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int i = 0;
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int n = 0;
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float rawAvg = 0;
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float err = 0;
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float rawAbsDev = 0;
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int upperLimit = 0;
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int lowerLimit = 0;
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float cumSum = 0;
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float x = 0;
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float x2 = 0;
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float y = 0;
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float xy = 0;
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float xAvg = 0;
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float denom = 0;
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float skew = 0;
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*skewEst = 0; // Set in case of error below.
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for (i = 0; i < size; i++) {
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if ((rawSkew[i] < absLimitOuter && rawSkew[i] > -absLimitOuter)) {
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n++;
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rawAvg += rawSkew[i];
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}
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}
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if (n == 0) {
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return -1;
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}
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RTC_DCHECK_GT(n, 0);
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rawAvg /= n;
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for (i = 0; i < size; i++) {
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if ((rawSkew[i] < absLimitOuter && rawSkew[i] > -absLimitOuter)) {
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err = rawSkew[i] - rawAvg;
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rawAbsDev += err >= 0 ? err : -err;
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}
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}
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RTC_DCHECK_GT(n, 0);
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rawAbsDev /= n;
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upperLimit = static_cast<int>(rawAvg + 5 * rawAbsDev + 1); // +1 for ceiling.
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lowerLimit = static_cast<int>(rawAvg - 5 * rawAbsDev - 1); // -1 for floor.
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n = 0;
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for (i = 0; i < size; i++) {
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if ((rawSkew[i] < absLimitInner && rawSkew[i] > -absLimitInner) ||
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(rawSkew[i] < upperLimit && rawSkew[i] > lowerLimit)) {
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n++;
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cumSum += rawSkew[i];
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x += n;
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x2 += n * n;
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y += cumSum;
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xy += n * cumSum;
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}
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}
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if (n == 0) {
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return -1;
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}
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RTC_DCHECK_GT(n, 0);
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xAvg = x / n;
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denom = x2 - xAvg * x;
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if (denom != 0) {
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skew = (xy - xAvg * y) / denom;
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}
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*skewEst = skew;
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return 0;
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}
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} // namespace webrtc
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