modules/audio_processing/residual_echo_detector.cc - src/webrtc - Git at Google

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

 #include "webrtc/modules/audio_processing/residual_echo_detector.h"

 #include <algorithm>
 #include <numeric>

 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
 #include "webrtc/rtc_base/atomicops.h"
 #include "webrtc/rtc_base/logging.h"
 #include "webrtc/system_wrappers/include/metrics.h"

 namespace {

 float Power(rtc::ArrayView<const float> input) {
   if (input.empty()) {
     return 0.f;
   }
   return std::inner_product(input.begin(), input.end(), input.begin(), 0.f) /
          input.size();
 }

 constexpr size_t kLookbackFrames = 650;
 // TODO(ivoc): Verify the size of this buffer.
 constexpr size_t kRenderBufferSize = 30;
 constexpr float kAlpha = 0.001f;
 // 10 seconds of data, updated every 10 ms.
 constexpr size_t kAggregationBufferSize = 10 * 100;

 }  // namespace

 namespace webrtc {

 int ResidualEchoDetector::instance_count_ = 0;

 ResidualEchoDetector::ResidualEchoDetector()
     : data_dumper_(
           new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
       render_buffer_(kRenderBufferSize),
       render_power_(kLookbackFrames),
       render_power_mean_(kLookbackFrames),
       render_power_std_dev_(kLookbackFrames),
       covariances_(kLookbackFrames),
       recent_likelihood_max_(kAggregationBufferSize) {}

 ResidualEchoDetector::~ResidualEchoDetector() = default;

 void ResidualEchoDetector::AnalyzeRenderAudio(
     rtc::ArrayView<const float> render_audio) {
   // Dump debug data assuming 48 kHz sample rate (if this assumption is not
   // valid the dumped audio will need to be converted offline accordingly).
   data_dumper_->DumpWav("ed_render", render_audio.size(), render_audio.data(),
                         48000, 1);

   if (render_buffer_.Size() == 0) {
     frames_since_zero_buffer_size_ = 0;
   } else if (frames_since_zero_buffer_size_ >= kRenderBufferSize) {
     // This can happen in a few cases: at the start of a call, due to a glitch
     // or due to clock drift. The excess capture value will be ignored.
     // TODO(ivoc): Include how often this happens in APM stats.
     render_buffer_.Pop();
     frames_since_zero_buffer_size_ = 0;
   }
   ++frames_since_zero_buffer_size_;
   float power = Power(render_audio);
   render_buffer_.Push(power);
 }

 void ResidualEchoDetector::AnalyzeCaptureAudio(
     rtc::ArrayView<const float> capture_audio) {
   // Dump debug data assuming 48 kHz sample rate (if this assumption is not
   // valid the dumped audio will need to be converted offline accordingly).
   data_dumper_->DumpWav("ed_capture", capture_audio.size(),
                         capture_audio.data(), 48000, 1);

   if (first_process_call_) {
     // On the first process call (so the start of a call), we must flush the
     // render buffer, otherwise the render data will be delayed.
     render_buffer_.Clear();
     first_process_call_ = false;
   }

   // Get the next render value.
   const rtc::Optional<float> buffered_render_power = render_buffer_.Pop();
   if (!buffered_render_power) {
     // This can happen in a few cases: at the start of a call, due to a glitch
     // or due to clock drift. The excess capture value will be ignored.
     // TODO(ivoc): Include how often this happens in APM stats.
     return;
   }
   // Update the render statistics, and store the statistics in circular buffers.
   render_statistics_.Update(*buffered_render_power);
   RTC_DCHECK_LT(next_insertion_index_, kLookbackFrames);
   render_power_[next_insertion_index_] = *buffered_render_power;
   render_power_mean_[next_insertion_index_] = render_statistics_.mean();
   render_power_std_dev_[next_insertion_index_] =
       render_statistics_.std_deviation();

   // Get the next capture value, update capture statistics and add the relevant
   // values to the buffers.
   const float capture_power = Power(capture_audio);
   capture_statistics_.Update(capture_power);
   const float capture_mean = capture_statistics_.mean();
   const float capture_std_deviation = capture_statistics_.std_deviation();

   // Update the covariance values and determine the new echo likelihood.
   echo_likelihood_ = 0.f;
   size_t read_index = next_insertion_index_;

   int best_delay = -1;
   for (size_t delay = 0; delay < covariances_.size(); ++delay) {
     RTC_DCHECK_LT(read_index, render_power_.size());
     covariances_[delay].Update(capture_power, capture_mean,
                                capture_std_deviation, render_power_[read_index],
                                render_power_mean_[read_index],
                                render_power_std_dev_[read_index]);
     read_index = read_index > 0 ? read_index - 1 : kLookbackFrames - 1;

     if (covariances_[delay].normalized_cross_correlation() > echo_likelihood_) {
       echo_likelihood_ = covariances_[delay].normalized_cross_correlation();
       best_delay = static_cast<int>(delay);
     }
   }
   // This is a temporary log message to help find the underlying cause for echo
   // likelihoods > 1.0.
   // TODO(ivoc): Remove once the issue is resolved.
   if (echo_likelihood_ > 1.1f) {
     // Make sure we don't spam the log.
     if (log_counter_ < 5 && best_delay != -1) {
       size_t read_index = kLookbackFrames + next_insertion_index_ - best_delay;
       if (read_index >= kLookbackFrames) {
         read_index -= kLookbackFrames;
       }
       RTC_DCHECK_LT(read_index, render_power_.size());
       LOG_F(LS_ERROR) << "Echo detector internal state: {"
                       << "Echo likelihood: " << echo_likelihood_
                       << ", Best Delay: " << best_delay << ", Covariance: "
                       << covariances_[best_delay].covariance()
                       << ", Last capture power: " << capture_power
                       << ", Capture mean: " << capture_mean
                       << ", Capture_standard deviation: "
                       << capture_std_deviation
                       << ", Last render power: " << render_power_[read_index]
                       << ", Render mean: " << render_power_mean_[read_index]
                       << ", Render standard deviation: "
                       << render_power_std_dev_[read_index]
                       << ", Reliability: " << reliability_ << "}";
       log_counter_++;
     }
   }
   RTC_DCHECK_LT(echo_likelihood_, 1.1f);

   reliability_ = (1.0f - kAlpha) * reliability_ + kAlpha * 1.0f;
   echo_likelihood_ *= reliability_;
   // This is a temporary fix to prevent echo likelihood values > 1.0.
   // TODO(ivoc): Find the root cause of this issue and fix it.
   echo_likelihood_ = std::min(echo_likelihood_, 1.0f);
   int echo_percentage = static_cast<int>(echo_likelihood_ * 100);
   RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ResidualEchoDetector.EchoLikelihood",
                        echo_percentage, 0, 100, 100 /* number of bins */);

   // Update the buffer of recent likelihood values.
   recent_likelihood_max_.Update(echo_likelihood_);

   // Update the next insertion index.
   next_insertion_index_ = next_insertion_index_ < (kLookbackFrames - 1)
                               ? next_insertion_index_ + 1
                               : 0;
 }

 void ResidualEchoDetector::Initialize() {
   render_buffer_.Clear();
   std::fill(render_power_.begin(), render_power_.end(), 0.f);
   std::fill(render_power_mean_.begin(), render_power_mean_.end(), 0.f);
   std::fill(render_power_std_dev_.begin(), render_power_std_dev_.end(), 0.f);
   render_statistics_.Clear();
   capture_statistics_.Clear();
   recent_likelihood_max_.Clear();
   for (auto& cov : covariances_) {
     cov.Clear();
   }
   echo_likelihood_ = 0.f;
   next_insertion_index_ = 0;
   reliability_ = 0.f;
 }

 void ResidualEchoDetector::PackRenderAudioBuffer(
     AudioBuffer* audio,
     std::vector<float>* packed_buffer) {
   packed_buffer->clear();
   packed_buffer->insert(packed_buffer->end(), audio->channels_f()[0],
                         audio->channels_f()[0] + audio->num_frames());
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
	*
	* Use of this source code is governed by a BSD-style license
	* that can be found in the LICENSE file in the root of the source
	* tree. An additional intellectual property rights grant can be found
	* in the file PATENTS. All contributing project authors may
	* be found in the AUTHORS file in the root of the source tree.
	*/

	#include "webrtc/modules/audio_processing/residual_echo_detector.h"

	#include <algorithm>
	#include <numeric>

	#include "webrtc/modules/audio_processing/audio_buffer.h"
	#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
	#include "webrtc/rtc_base/atomicops.h"
	#include "webrtc/rtc_base/logging.h"
	#include "webrtc/system_wrappers/include/metrics.h"

	namespace {

	float Power(rtc::ArrayView<const float> input) {
	if (input.empty()) {
	return 0.f;
	}
	return std::inner_product(input.begin(), input.end(), input.begin(), 0.f) /
	input.size();
	}

	constexpr size_t kLookbackFrames = 650;
	// TODO(ivoc): Verify the size of this buffer.
	constexpr size_t kRenderBufferSize = 30;
	constexpr float kAlpha = 0.001f;
	// 10 seconds of data, updated every 10 ms.
	constexpr size_t kAggregationBufferSize = 10 * 100;

	} // namespace

	namespace webrtc {

	int ResidualEchoDetector::instance_count_ = 0;

	ResidualEchoDetector::ResidualEchoDetector()
	: data_dumper_(
	new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
	render_buffer_(kRenderBufferSize),
	render_power_(kLookbackFrames),
	render_power_mean_(kLookbackFrames),
	render_power_std_dev_(kLookbackFrames),
	covariances_(kLookbackFrames),
	recent_likelihood_max_(kAggregationBufferSize) {}

	ResidualEchoDetector::~ResidualEchoDetector() = default;

	void ResidualEchoDetector::AnalyzeRenderAudio(
	rtc::ArrayView<const float> render_audio) {
	// Dump debug data assuming 48 kHz sample rate (if this assumption is not
	// valid the dumped audio will need to be converted offline accordingly).
	data_dumper_->DumpWav("ed_render", render_audio.size(), render_audio.data(),
	48000, 1);

	if (render_buffer_.Size() == 0) {
	frames_since_zero_buffer_size_ = 0;
	} else if (frames_since_zero_buffer_size_ >= kRenderBufferSize) {
	// This can happen in a few cases: at the start of a call, due to a glitch
	// or due to clock drift. The excess capture value will be ignored.
	// TODO(ivoc): Include how often this happens in APM stats.
	render_buffer_.Pop();
	frames_since_zero_buffer_size_ = 0;
	}
	++frames_since_zero_buffer_size_;
	float power = Power(render_audio);
	render_buffer_.Push(power);
	}

	void ResidualEchoDetector::AnalyzeCaptureAudio(
	rtc::ArrayView<const float> capture_audio) {
	// Dump debug data assuming 48 kHz sample rate (if this assumption is not
	// valid the dumped audio will need to be converted offline accordingly).
	data_dumper_->DumpWav("ed_capture", capture_audio.size(),
	capture_audio.data(), 48000, 1);

	if (first_process_call_) {
	// On the first process call (so the start of a call), we must flush the
	// render buffer, otherwise the render data will be delayed.
	render_buffer_.Clear();
	first_process_call_ = false;
	}

	// Get the next render value.
	const rtc::Optional<float> buffered_render_power = render_buffer_.Pop();
	if (!buffered_render_power) {
	// This can happen in a few cases: at the start of a call, due to a glitch
	// or due to clock drift. The excess capture value will be ignored.
	// TODO(ivoc): Include how often this happens in APM stats.
	return;
	}
	// Update the render statistics, and store the statistics in circular buffers.
	render_statistics_.Update(*buffered_render_power);
	RTC_DCHECK_LT(next_insertion_index_, kLookbackFrames);
	render_power_[next_insertion_index_] = *buffered_render_power;
	render_power_mean_[next_insertion_index_] = render_statistics_.mean();
	render_power_std_dev_[next_insertion_index_] =
	render_statistics_.std_deviation();

	// Get the next capture value, update capture statistics and add the relevant
	// values to the buffers.
	const float capture_power = Power(capture_audio);
	capture_statistics_.Update(capture_power);
	const float capture_mean = capture_statistics_.mean();
	const float capture_std_deviation = capture_statistics_.std_deviation();

	// Update the covariance values and determine the new echo likelihood.
	echo_likelihood_ = 0.f;
	size_t read_index = next_insertion_index_;

	int best_delay = -1;
	for (size_t delay = 0; delay < covariances_.size(); ++delay) {
	RTC_DCHECK_LT(read_index, render_power_.size());
	covariances_[delay].Update(capture_power, capture_mean,
	capture_std_deviation, render_power_[read_index],
	render_power_mean_[read_index],
	render_power_std_dev_[read_index]);
	read_index = read_index > 0 ? read_index - 1 : kLookbackFrames - 1;

	if (covariances_[delay].normalized_cross_correlation() > echo_likelihood_) {
	echo_likelihood_ = covariances_[delay].normalized_cross_correlation();
	best_delay = static_cast<int>(delay);
	}
	}
	// This is a temporary log message to help find the underlying cause for echo
	// likelihoods > 1.0.
	// TODO(ivoc): Remove once the issue is resolved.
	if (echo_likelihood_ > 1.1f) {
	// Make sure we don't spam the log.
	if (log_counter_ < 5 && best_delay != -1) {
	size_t read_index = kLookbackFrames + next_insertion_index_ - best_delay;
	if (read_index >= kLookbackFrames) {
	read_index -= kLookbackFrames;
	}
	RTC_DCHECK_LT(read_index, render_power_.size());
	LOG_F(LS_ERROR) << "Echo detector internal state: {"
	<< "Echo likelihood: " << echo_likelihood_
	<< ", Best Delay: " << best_delay << ", Covariance: "
	<< covariances_[best_delay].covariance()
	<< ", Last capture power: " << capture_power
	<< ", Capture mean: " << capture_mean
	<< ", Capture_standard deviation: "
	<< capture_std_deviation
	<< ", Last render power: " << render_power_[read_index]
	<< ", Render mean: " << render_power_mean_[read_index]
	<< ", Render standard deviation: "
	<< render_power_std_dev_[read_index]
	<< ", Reliability: " << reliability_ << "}";
	log_counter_++;
	}
	}
	RTC_DCHECK_LT(echo_likelihood_, 1.1f);

	reliability_ = (1.0f - kAlpha) * reliability_ + kAlpha * 1.0f;
	echo_likelihood_ *= reliability_;
	// This is a temporary fix to prevent echo likelihood values > 1.0.
	// TODO(ivoc): Find the root cause of this issue and fix it.
	echo_likelihood_ = std::min(echo_likelihood_, 1.0f);
	int echo_percentage = static_cast<int>(echo_likelihood_ * 100);
	RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ResidualEchoDetector.EchoLikelihood",
	echo_percentage, 0, 100, 100 /* number of bins */);

	// Update the buffer of recent likelihood values.
	recent_likelihood_max_.Update(echo_likelihood_);

	// Update the next insertion index.
	next_insertion_index_ = next_insertion_index_ < (kLookbackFrames - 1)
	? next_insertion_index_ + 1
	: 0;
	}

	void ResidualEchoDetector::Initialize() {
	render_buffer_.Clear();
	std::fill(render_power_.begin(), render_power_.end(), 0.f);
	std::fill(render_power_mean_.begin(), render_power_mean_.end(), 0.f);
	std::fill(render_power_std_dev_.begin(), render_power_std_dev_.end(), 0.f);
	render_statistics_.Clear();
	capture_statistics_.Clear();
	recent_likelihood_max_.Clear();
	for (auto& cov : covariances_) {
	cov.Clear();
	}
	echo_likelihood_ = 0.f;
	next_insertion_index_ = 0;
	reliability_ = 0.f;
	}

	void ResidualEchoDetector::PackRenderAudioBuffer(
	AudioBuffer* audio,
	std::vector<float>* packed_buffer) {
	packed_buffer->clear();
	packed_buffer->insert(packed_buffer->end(), audio->channels_f()[0],
	audio->channels_f()[0] + audio->num_frames());
	}

	} // namespace webrtc