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

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

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

 #include <string.h>

 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 #include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"

 namespace webrtc {

 namespace {
 int16_t MapSetting(EchoCancellation::SuppressionLevel level) {
   switch (level) {
     case EchoCancellation::kLowSuppression:
       return kAecNlpConservative;
     case EchoCancellation::kModerateSuppression:
       return kAecNlpModerate;
     case EchoCancellation::kHighSuppression:
       return kAecNlpAggressive;
   }
   RTC_NOTREACHED();
   return -1;
 }

 AudioProcessing::Error MapError(int err) {
   switch (err) {
     case AEC_UNSUPPORTED_FUNCTION_ERROR:
       return AudioProcessing::kUnsupportedFunctionError;
     case AEC_BAD_PARAMETER_ERROR:
       return AudioProcessing::kBadParameterError;
     case AEC_BAD_PARAMETER_WARNING:
       return AudioProcessing::kBadStreamParameterWarning;
     default:
       // AEC_UNSPECIFIED_ERROR
       // AEC_UNINITIALIZED_ERROR
       // AEC_NULL_POINTER_ERROR
       return AudioProcessing::kUnspecifiedError;
   }
 }

 // Maximum length that a frame of samples can have.
 static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
 // Maximum number of frames to buffer in the render queue.
 // TODO(peah): Decrease this once we properly handle hugely unbalanced
 // reverse and forward call numbers.
 static const size_t kMaxNumFramesToBuffer = 100;
 }  // namespace

 struct EchoCancellationImpl::StreamProperties {
   StreamProperties() = delete;
   StreamProperties(int sample_rate_hz,
                    size_t num_reverse_channels,
                    size_t num_output_channels,
                    size_t num_proc_channels)
       : sample_rate_hz(sample_rate_hz),
         num_reverse_channels(num_reverse_channels),
         num_output_channels(num_output_channels),
         num_proc_channels(num_proc_channels) {}

   const int sample_rate_hz;
   const size_t num_reverse_channels;
   const size_t num_output_channels;
   const size_t num_proc_channels;
 };

 class EchoCancellationImpl::Canceller {
  public:
   Canceller() {
     state_ = WebRtcAec_Create();
     RTC_DCHECK(state_);
   }

   ~Canceller() {
     RTC_CHECK(state_);
     WebRtcAec_Free(state_);
   }

   void* state() { return state_; }

   void Initialize(int sample_rate_hz) {
     // TODO(ajm): Drift compensation is disabled in practice. If restored, it
     // should be managed internally and not depend on the hardware sample rate.
     // For now, just hardcode a 48 kHz value.
     const int error = WebRtcAec_Init(state_, sample_rate_hz, 48000);
     RTC_DCHECK_EQ(0, error);
   }

  private:
   void* state_;
 };

 EchoCancellationImpl::EchoCancellationImpl(rtc::CriticalSection* crit_render,
                                            rtc::CriticalSection* crit_capture)
     : crit_render_(crit_render),
       crit_capture_(crit_capture),
       drift_compensation_enabled_(false),
       metrics_enabled_(false),
       suppression_level_(kModerateSuppression),
       stream_drift_samples_(0),
       was_stream_drift_set_(false),
       stream_has_echo_(false),
       delay_logging_enabled_(false),
       extended_filter_enabled_(false),
       delay_agnostic_enabled_(false),
       aec3_enabled_(false),
       render_queue_element_max_size_(0) {
   RTC_DCHECK(crit_render);
   RTC_DCHECK(crit_capture);
 }

 EchoCancellationImpl::~EchoCancellationImpl() {}

 int EchoCancellationImpl::ProcessRenderAudio(const AudioBuffer* audio) {
   rtc::CritScope cs_render(crit_render_);
   if (!enabled_) {
     return AudioProcessing::kNoError;
   }

   RTC_DCHECK(stream_properties_);
   RTC_DCHECK_GE(160u, audio->num_frames_per_band());
   RTC_DCHECK_EQ(audio->num_channels(),
                 stream_properties_->num_reverse_channels);
   RTC_DCHECK_GE(cancellers_.size(), stream_properties_->num_output_channels *
                                         audio->num_channels());

   int err = AudioProcessing::kNoError;

   // The ordering convention must be followed to pass to the correct AEC.
   size_t handle_index = 0;
   render_queue_buffer_.clear();
   for (size_t i = 0; i < stream_properties_->num_output_channels; i++) {
     for (size_t j = 0; j < audio->num_channels(); j++) {
       // Retrieve any error code produced by the buffering of the farend
       // signal.
       err = WebRtcAec_GetBufferFarendError(
           cancellers_[handle_index++]->state(),
           audio->split_bands_const_f(j)[kBand0To8kHz],
           audio->num_frames_per_band());

       if (err != AudioProcessing::kNoError) {
         return MapError(err);  // TODO(ajm): warning possible?
       }

       // Buffer the samples in the render queue.
       render_queue_buffer_.insert(render_queue_buffer_.end(),
                                   audio->split_bands_const_f(j)[kBand0To8kHz],
                                   (audio->split_bands_const_f(j)[kBand0To8kHz] +
                                    audio->num_frames_per_band()));
     }
   }

   // Insert the samples into the queue.
   if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
     // The data queue is full and needs to be emptied.
     ReadQueuedRenderData();

     // Retry the insert (should always work).
     RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
   }

   return AudioProcessing::kNoError;
 }

 // Read chunks of data that were received and queued on the render side from
 // a queue. All the data chunks are buffered into the farend signal of the AEC.
 void EchoCancellationImpl::ReadQueuedRenderData() {
   rtc::CritScope cs_capture(crit_capture_);
   if (!enabled_) {
     return;
   }

   RTC_DCHECK(stream_properties_);
   while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
     size_t handle_index = 0;
     size_t buffer_index = 0;
     const size_t num_frames_per_band =
         capture_queue_buffer_.size() /
         (stream_properties_->num_output_channels *
          stream_properties_->num_reverse_channels);
     for (size_t i = 0; i < stream_properties_->num_output_channels; i++) {
       for (size_t j = 0; j < stream_properties_->num_reverse_channels; j++) {
         WebRtcAec_BufferFarend(cancellers_[handle_index++]->state(),
                                &capture_queue_buffer_[buffer_index],
                                num_frames_per_band);

         buffer_index += num_frames_per_band;
       }
     }
   }
 }

 int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio,
                                               int stream_delay_ms) {
   rtc::CritScope cs_capture(crit_capture_);
   if (!enabled_) {
     return AudioProcessing::kNoError;
   }

   if (drift_compensation_enabled_ && !was_stream_drift_set_) {
     return AudioProcessing::kStreamParameterNotSetError;
   }

   RTC_DCHECK(stream_properties_);
   RTC_DCHECK_GE(160u, audio->num_frames_per_band());
   RTC_DCHECK_EQ(audio->num_channels(), stream_properties_->num_proc_channels);

   int err = AudioProcessing::kNoError;

   // The ordering convention must be followed to pass to the correct AEC.
   size_t handle_index = 0;
   stream_has_echo_ = false;
   for (size_t i = 0; i < audio->num_channels(); i++) {
     for (size_t j = 0; j < stream_properties_->num_reverse_channels; j++) {
       err = WebRtcAec_Process(
           cancellers_[handle_index]->state(), audio->split_bands_const_f(i),
           audio->num_bands(), audio->split_bands_f(i),
           audio->num_frames_per_band(), stream_delay_ms, stream_drift_samples_);

       if (err != AudioProcessing::kNoError) {
         err = MapError(err);
         // TODO(ajm): Figure out how to return warnings properly.
         if (err != AudioProcessing::kBadStreamParameterWarning) {
           return err;
         }
       }

       int status = 0;
       err = WebRtcAec_get_echo_status(cancellers_[handle_index]->state(),
                                       &status);
       if (err != AudioProcessing::kNoError) {
         return MapError(err);
       }

       if (status == 1) {
         stream_has_echo_ = true;
       }

       handle_index++;
     }
   }

   was_stream_drift_set_ = false;
   return AudioProcessing::kNoError;
 }

 int EchoCancellationImpl::Enable(bool enable) {
   // Run in a single-threaded manner.
   rtc::CritScope cs_render(crit_render_);
   rtc::CritScope cs_capture(crit_capture_);

   if (enable && !enabled_) {
     enabled_ = enable;  // Must be set before Initialize() is called.

     // TODO(peah): Simplify once the Enable function has been removed from
     // the public APM API.
     RTC_DCHECK(stream_properties_);
     Initialize(stream_properties_->sample_rate_hz,
                stream_properties_->num_reverse_channels,
                stream_properties_->num_output_channels,
                stream_properties_->num_proc_channels);
   } else {
     enabled_ = enable;
   }
   return AudioProcessing::kNoError;
 }

 bool EchoCancellationImpl::is_enabled() const {
   rtc::CritScope cs(crit_capture_);
   return enabled_;
 }

 int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) {
   {
     if (MapSetting(level) == -1) {
       return AudioProcessing::kBadParameterError;
     }
     rtc::CritScope cs(crit_capture_);
     suppression_level_ = level;
   }
   return Configure();
 }

 EchoCancellation::SuppressionLevel EchoCancellationImpl::suppression_level()
     const {
   rtc::CritScope cs(crit_capture_);
   return suppression_level_;
 }

 int EchoCancellationImpl::enable_drift_compensation(bool enable) {
   {
     rtc::CritScope cs(crit_capture_);
     drift_compensation_enabled_ = enable;
   }
   return Configure();
 }

 bool EchoCancellationImpl::is_drift_compensation_enabled() const {
   rtc::CritScope cs(crit_capture_);
   return drift_compensation_enabled_;
 }

 void EchoCancellationImpl::set_stream_drift_samples(int drift) {
   rtc::CritScope cs(crit_capture_);
   was_stream_drift_set_ = true;
   stream_drift_samples_ = drift;
 }

 int EchoCancellationImpl::stream_drift_samples() const {
   rtc::CritScope cs(crit_capture_);
   return stream_drift_samples_;
 }

 int EchoCancellationImpl::enable_metrics(bool enable) {
   {
     rtc::CritScope cs(crit_capture_);
     metrics_enabled_ = enable;
   }
   return Configure();
 }

 bool EchoCancellationImpl::are_metrics_enabled() const {
   rtc::CritScope cs(crit_capture_);
   return metrics_enabled_;
 }

 // TODO(ajm): we currently just use the metrics from the first AEC. Think more
 //            aboue the best way to extend this to multi-channel.
 int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
   rtc::CritScope cs(crit_capture_);
   if (metrics == NULL) {
     return AudioProcessing::kNullPointerError;
   }

   if (!enabled_ || !metrics_enabled_) {
     return AudioProcessing::kNotEnabledError;
   }

   AecMetrics my_metrics;
   memset(&my_metrics, 0, sizeof(my_metrics));
   memset(metrics, 0, sizeof(Metrics));

   const int err = WebRtcAec_GetMetrics(cancellers_[0]->state(), &my_metrics);
   if (err != AudioProcessing::kNoError) {
     return MapError(err);
   }

   metrics->residual_echo_return_loss.instant = my_metrics.rerl.instant;
   metrics->residual_echo_return_loss.average = my_metrics.rerl.average;
   metrics->residual_echo_return_loss.maximum = my_metrics.rerl.max;
   metrics->residual_echo_return_loss.minimum = my_metrics.rerl.min;

   metrics->echo_return_loss.instant = my_metrics.erl.instant;
   metrics->echo_return_loss.average = my_metrics.erl.average;
   metrics->echo_return_loss.maximum = my_metrics.erl.max;
   metrics->echo_return_loss.minimum = my_metrics.erl.min;

   metrics->echo_return_loss_enhancement.instant = my_metrics.erle.instant;
   metrics->echo_return_loss_enhancement.average = my_metrics.erle.average;
   metrics->echo_return_loss_enhancement.maximum = my_metrics.erle.max;
   metrics->echo_return_loss_enhancement.minimum = my_metrics.erle.min;

   metrics->a_nlp.instant = my_metrics.aNlp.instant;
   metrics->a_nlp.average = my_metrics.aNlp.average;
   metrics->a_nlp.maximum = my_metrics.aNlp.max;
   metrics->a_nlp.minimum = my_metrics.aNlp.min;

   metrics->divergent_filter_fraction = my_metrics.divergent_filter_fraction;
   return AudioProcessing::kNoError;
 }

 bool EchoCancellationImpl::stream_has_echo() const {
   rtc::CritScope cs(crit_capture_);
   return stream_has_echo_;
 }

 int EchoCancellationImpl::enable_delay_logging(bool enable) {
   {
     rtc::CritScope cs(crit_capture_);
     delay_logging_enabled_ = enable;
   }
   return Configure();
 }

 bool EchoCancellationImpl::is_delay_logging_enabled() const {
   rtc::CritScope cs(crit_capture_);
   return delay_logging_enabled_;
 }

 bool EchoCancellationImpl::is_delay_agnostic_enabled() const {
   rtc::CritScope cs(crit_capture_);
   return delay_agnostic_enabled_;
 }

 bool EchoCancellationImpl::is_aec3_enabled() const {
   rtc::CritScope cs(crit_capture_);
   return aec3_enabled_;
 }

 std::string EchoCancellationImpl::GetExperimentsDescription() {
   rtc::CritScope cs(crit_capture_);
   std::string description = (aec3_enabled_ ? "AEC3;" : "");
   if (refined_adaptive_filter_enabled_) {
     description += "RefinedAdaptiveFilter;";
   }
   return description;
 }

 bool EchoCancellationImpl::is_refined_adaptive_filter_enabled() const {
   rtc::CritScope cs(crit_capture_);
   return refined_adaptive_filter_enabled_;
 }

 bool EchoCancellationImpl::is_extended_filter_enabled() const {
   rtc::CritScope cs(crit_capture_);
   return extended_filter_enabled_;
 }

 // TODO(bjornv): How should we handle the multi-channel case?
 int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) {
   rtc::CritScope cs(crit_capture_);
   float fraction_poor_delays = 0;
   return GetDelayMetrics(median, std, &fraction_poor_delays);
 }

 int EchoCancellationImpl::GetDelayMetrics(int* median, int* std,
                                           float* fraction_poor_delays) {
   rtc::CritScope cs(crit_capture_);
   if (median == NULL) {
     return AudioProcessing::kNullPointerError;
   }
   if (std == NULL) {
     return AudioProcessing::kNullPointerError;
   }

   if (!enabled_ || !delay_logging_enabled_) {
     return AudioProcessing::kNotEnabledError;
   }

   const int err = WebRtcAec_GetDelayMetrics(cancellers_[0]->state(), median,
                                             std, fraction_poor_delays);
   if (err != AudioProcessing::kNoError) {
     return MapError(err);
   }

   return AudioProcessing::kNoError;
 }

 struct AecCore* EchoCancellationImpl::aec_core() const {
   rtc::CritScope cs(crit_capture_);
   if (!enabled_) {
     return NULL;
   }
   return WebRtcAec_aec_core(cancellers_[0]->state());
 }

 void EchoCancellationImpl::Initialize(int sample_rate_hz,
                                       size_t num_reverse_channels,
                                       size_t num_output_channels,
                                       size_t num_proc_channels) {
   rtc::CritScope cs_render(crit_render_);
   rtc::CritScope cs_capture(crit_capture_);

   stream_properties_.reset(
       new StreamProperties(sample_rate_hz, num_reverse_channels,
                            num_output_channels, num_proc_channels));

   if (!enabled_) {
     return;
   }

   if (NumCancellersRequired() > cancellers_.size()) {
     const size_t cancellers_old_size = cancellers_.size();
     cancellers_.resize(NumCancellersRequired());

     for (size_t i = cancellers_old_size; i < cancellers_.size(); ++i) {
       cancellers_[i].reset(new Canceller());
     }
   }

   for (auto& canceller : cancellers_) {
     canceller->Initialize(sample_rate_hz);
   }

   Configure();

   AllocateRenderQueue();
 }

 int EchoCancellationImpl::GetSystemDelayInSamples() const {
   rtc::CritScope cs(crit_capture_);
   RTC_DCHECK(enabled_);
   // Report the delay for the first AEC component.
   return WebRtcAec_system_delay(
       WebRtcAec_aec_core(cancellers_[0]->state()));
 }

 void EchoCancellationImpl::AllocateRenderQueue() {
   const size_t new_render_queue_element_max_size = std::max<size_t>(
       static_cast<size_t>(1),
       kMaxAllowedValuesOfSamplesPerFrame * NumCancellersRequired());

   rtc::CritScope cs_render(crit_render_);
   rtc::CritScope cs_capture(crit_capture_);

   // Reallocate the queue if the queue item size is too small to fit the
   // data to put in the queue.
   if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
     render_queue_element_max_size_ = new_render_queue_element_max_size;

     std::vector<float> template_queue_element(render_queue_element_max_size_);

     render_signal_queue_.reset(
         new SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>(
             kMaxNumFramesToBuffer, template_queue_element,
             RenderQueueItemVerifier<float>(render_queue_element_max_size_)));

     render_queue_buffer_.resize(render_queue_element_max_size_);
     capture_queue_buffer_.resize(render_queue_element_max_size_);
   } else {
     render_signal_queue_->Clear();
   }
 }

 void EchoCancellationImpl::SetExtraOptions(const webrtc::Config& config) {
   {
     rtc::CritScope cs(crit_capture_);
     extended_filter_enabled_ = config.Get<ExtendedFilter>().enabled;
     delay_agnostic_enabled_ = config.Get<DelayAgnostic>().enabled;
     refined_adaptive_filter_enabled_ =
         config.Get<RefinedAdaptiveFilter>().enabled;
     aec3_enabled_ = config.Get<EchoCanceller3>().enabled;
   }
   Configure();
 }

 int EchoCancellationImpl::Configure() {
   rtc::CritScope cs_render(crit_render_);
   rtc::CritScope cs_capture(crit_capture_);
   AecConfig config;
   config.metricsMode = metrics_enabled_;
   config.nlpMode = MapSetting(suppression_level_);
   config.skewMode = drift_compensation_enabled_;
   config.delay_logging = delay_logging_enabled_;

   int error = AudioProcessing::kNoError;
   for (auto& canceller : cancellers_) {
     WebRtcAec_enable_extended_filter(WebRtcAec_aec_core(canceller->state()),
                                      extended_filter_enabled_ ? 1 : 0);
     WebRtcAec_enable_delay_agnostic(WebRtcAec_aec_core(canceller->state()),
                                     delay_agnostic_enabled_ ? 1 : 0);
     WebRtcAec_enable_aec3(WebRtcAec_aec_core(canceller->state()),
                           aec3_enabled_ ? 1 : 0);
     WebRtcAec_enable_refined_adaptive_filter(
         WebRtcAec_aec_core(canceller->state()),
         refined_adaptive_filter_enabled_);
     const int handle_error = WebRtcAec_set_config(canceller->state(), config);
     if (handle_error != AudioProcessing::kNoError) {
       error = AudioProcessing::kNoError;
     }
   }
   return error;
 }

 size_t EchoCancellationImpl::NumCancellersRequired() const {
   RTC_DCHECK(stream_properties_);
   return stream_properties_->num_output_channels *
          stream_properties_->num_reverse_channels;
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
	*
	* Use of this source code is governed by a BSD-style license
	* that can be found in the LICENSE file in the root of the source
	* tree. An additional intellectual property rights grant can be found
	* in the file PATENTS. All contributing project authors may
	* be found in the AUTHORS file in the root of the source tree.
	*/

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

	#include <string.h>

	#include "webrtc/base/checks.h"
	#include "webrtc/modules/audio_processing/aec/aec_core.h"
	#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
	#include "webrtc/modules/audio_processing/audio_buffer.h"

	namespace webrtc {

	namespace {
	int16_t MapSetting(EchoCancellation::SuppressionLevel level) {
	switch (level) {
	case EchoCancellation::kLowSuppression:
	return kAecNlpConservative;
	case EchoCancellation::kModerateSuppression:
	return kAecNlpModerate;
	case EchoCancellation::kHighSuppression:
	return kAecNlpAggressive;
	}
	RTC_NOTREACHED();
	return -1;
	}

	AudioProcessing::Error MapError(int err) {
	switch (err) {
	case AEC_UNSUPPORTED_FUNCTION_ERROR:
	return AudioProcessing::kUnsupportedFunctionError;
	case AEC_BAD_PARAMETER_ERROR:
	return AudioProcessing::kBadParameterError;
	case AEC_BAD_PARAMETER_WARNING:
	return AudioProcessing::kBadStreamParameterWarning;
	default:
	// AEC_UNSPECIFIED_ERROR
	// AEC_UNINITIALIZED_ERROR
	// AEC_NULL_POINTER_ERROR
	return AudioProcessing::kUnspecifiedError;
	}
	}

	// Maximum length that a frame of samples can have.
	static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
	// Maximum number of frames to buffer in the render queue.
	// TODO(peah): Decrease this once we properly handle hugely unbalanced
	// reverse and forward call numbers.
	static const size_t kMaxNumFramesToBuffer = 100;
	} // namespace

	struct EchoCancellationImpl::StreamProperties {
	StreamProperties() = delete;
	StreamProperties(int sample_rate_hz,
	size_t num_reverse_channels,
	size_t num_output_channels,
	size_t num_proc_channels)
	: sample_rate_hz(sample_rate_hz),
	num_reverse_channels(num_reverse_channels),
	num_output_channels(num_output_channels),
	num_proc_channels(num_proc_channels) {}

	const int sample_rate_hz;
	const size_t num_reverse_channels;
	const size_t num_output_channels;
	const size_t num_proc_channels;
	};

	class EchoCancellationImpl::Canceller {
	public:
	Canceller() {
	state_ = WebRtcAec_Create();
	RTC_DCHECK(state_);
	}

	~Canceller() {
	RTC_CHECK(state_);
	WebRtcAec_Free(state_);
	}

	void* state() { return state_; }

	void Initialize(int sample_rate_hz) {
	// TODO(ajm): Drift compensation is disabled in practice. If restored, it
	// should be managed internally and not depend on the hardware sample rate.
	// For now, just hardcode a 48 kHz value.
	const int error = WebRtcAec_Init(state_, sample_rate_hz, 48000);
	RTC_DCHECK_EQ(0, error);
	}

	private:
	void* state_;
	};

	EchoCancellationImpl::EchoCancellationImpl(rtc::CriticalSection* crit_render,
	rtc::CriticalSection* crit_capture)
	: crit_render_(crit_render),
	crit_capture_(crit_capture),
	drift_compensation_enabled_(false),
	metrics_enabled_(false),
	suppression_level_(kModerateSuppression),
	stream_drift_samples_(0),
	was_stream_drift_set_(false),
	stream_has_echo_(false),
	delay_logging_enabled_(false),
	extended_filter_enabled_(false),
	delay_agnostic_enabled_(false),
	aec3_enabled_(false),
	render_queue_element_max_size_(0) {
	RTC_DCHECK(crit_render);
	RTC_DCHECK(crit_capture);
	}

	EchoCancellationImpl::~EchoCancellationImpl() {}

	int EchoCancellationImpl::ProcessRenderAudio(const AudioBuffer* audio) {
	rtc::CritScope cs_render(crit_render_);
	if (!enabled_) {
	return AudioProcessing::kNoError;
	}

	RTC_DCHECK(stream_properties_);
	RTC_DCHECK_GE(160u, audio->num_frames_per_band());
	RTC_DCHECK_EQ(audio->num_channels(),
	stream_properties_->num_reverse_channels);
	RTC_DCHECK_GE(cancellers_.size(), stream_properties_->num_output_channels *
	audio->num_channels());

	int err = AudioProcessing::kNoError;

	// The ordering convention must be followed to pass to the correct AEC.
	size_t handle_index = 0;
	render_queue_buffer_.clear();
	for (size_t i = 0; i < stream_properties_->num_output_channels; i++) {
	for (size_t j = 0; j < audio->num_channels(); j++) {
	// Retrieve any error code produced by the buffering of the farend
	// signal.
	err = WebRtcAec_GetBufferFarendError(
	cancellers_[handle_index++]->state(),
	audio->split_bands_const_f(j)[kBand0To8kHz],
	audio->num_frames_per_band());

	if (err != AudioProcessing::kNoError) {
	return MapError(err); // TODO(ajm): warning possible?
	}

	// Buffer the samples in the render queue.
	render_queue_buffer_.insert(render_queue_buffer_.end(),
	audio->split_bands_const_f(j)[kBand0To8kHz],
	(audio->split_bands_const_f(j)[kBand0To8kHz] +
	audio->num_frames_per_band()));
	}
	}

	// Insert the samples into the queue.
	if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
	// The data queue is full and needs to be emptied.
	ReadQueuedRenderData();

	// Retry the insert (should always work).
	RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
	}

	return AudioProcessing::kNoError;
	}

	// Read chunks of data that were received and queued on the render side from
	// a queue. All the data chunks are buffered into the farend signal of the AEC.
	void EchoCancellationImpl::ReadQueuedRenderData() {
	rtc::CritScope cs_capture(crit_capture_);
	if (!enabled_) {
	return;
	}

	RTC_DCHECK(stream_properties_);
	while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
	size_t handle_index = 0;
	size_t buffer_index = 0;
	const size_t num_frames_per_band =
	capture_queue_buffer_.size() /
	(stream_properties_->num_output_channels *
	stream_properties_->num_reverse_channels);
	for (size_t i = 0; i < stream_properties_->num_output_channels; i++) {
	for (size_t j = 0; j < stream_properties_->num_reverse_channels; j++) {
	WebRtcAec_BufferFarend(cancellers_[handle_index++]->state(),
	&capture_queue_buffer_[buffer_index],
	num_frames_per_band);

	buffer_index += num_frames_per_band;
	}
	}
	}
	}

	int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio,
	int stream_delay_ms) {
	rtc::CritScope cs_capture(crit_capture_);
	if (!enabled_) {
	return AudioProcessing::kNoError;
	}

	if (drift_compensation_enabled_ && !was_stream_drift_set_) {
	return AudioProcessing::kStreamParameterNotSetError;
	}

	RTC_DCHECK(stream_properties_);
	RTC_DCHECK_GE(160u, audio->num_frames_per_band());
	RTC_DCHECK_EQ(audio->num_channels(), stream_properties_->num_proc_channels);

	int err = AudioProcessing::kNoError;

	// The ordering convention must be followed to pass to the correct AEC.
	size_t handle_index = 0;
	stream_has_echo_ = false;
	for (size_t i = 0; i < audio->num_channels(); i++) {
	for (size_t j = 0; j < stream_properties_->num_reverse_channels; j++) {
	err = WebRtcAec_Process(
	cancellers_[handle_index]->state(), audio->split_bands_const_f(i),
	audio->num_bands(), audio->split_bands_f(i),
	audio->num_frames_per_band(), stream_delay_ms, stream_drift_samples_);

	if (err != AudioProcessing::kNoError) {
	err = MapError(err);
	// TODO(ajm): Figure out how to return warnings properly.
	if (err != AudioProcessing::kBadStreamParameterWarning) {
	return err;
	}
	}

	int status = 0;
	err = WebRtcAec_get_echo_status(cancellers_[handle_index]->state(),
	&status);
	if (err != AudioProcessing::kNoError) {
	return MapError(err);
	}

	if (status == 1) {
	stream_has_echo_ = true;
	}

	handle_index++;
	}
	}

	was_stream_drift_set_ = false;
	return AudioProcessing::kNoError;
	}

	int EchoCancellationImpl::Enable(bool enable) {
	// Run in a single-threaded manner.
	rtc::CritScope cs_render(crit_render_);
	rtc::CritScope cs_capture(crit_capture_);

	if (enable && !enabled_) {
	enabled_ = enable; // Must be set before Initialize() is called.

	// TODO(peah): Simplify once the Enable function has been removed from
	// the public APM API.
	RTC_DCHECK(stream_properties_);
	Initialize(stream_properties_->sample_rate_hz,
	stream_properties_->num_reverse_channels,
	stream_properties_->num_output_channels,
	stream_properties_->num_proc_channels);
	} else {
	enabled_ = enable;
	}
	return AudioProcessing::kNoError;
	}

	bool EchoCancellationImpl::is_enabled() const {
	rtc::CritScope cs(crit_capture_);
	return enabled_;
	}

	int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) {
	{
	if (MapSetting(level) == -1) {
	return AudioProcessing::kBadParameterError;
	}
	rtc::CritScope cs(crit_capture_);
	suppression_level_ = level;
	}
	return Configure();
	}

	EchoCancellation::SuppressionLevel EchoCancellationImpl::suppression_level()
	const {
	rtc::CritScope cs(crit_capture_);
	return suppression_level_;
	}

	int EchoCancellationImpl::enable_drift_compensation(bool enable) {
	{
	rtc::CritScope cs(crit_capture_);
	drift_compensation_enabled_ = enable;
	}
	return Configure();
	}

	bool EchoCancellationImpl::is_drift_compensation_enabled() const {
	rtc::CritScope cs(crit_capture_);
	return drift_compensation_enabled_;
	}

	void EchoCancellationImpl::set_stream_drift_samples(int drift) {
	rtc::CritScope cs(crit_capture_);
	was_stream_drift_set_ = true;
	stream_drift_samples_ = drift;
	}

	int EchoCancellationImpl::stream_drift_samples() const {
	rtc::CritScope cs(crit_capture_);
	return stream_drift_samples_;
	}

	int EchoCancellationImpl::enable_metrics(bool enable) {
	{
	rtc::CritScope cs(crit_capture_);
	metrics_enabled_ = enable;
	}
	return Configure();
	}

	bool EchoCancellationImpl::are_metrics_enabled() const {
	rtc::CritScope cs(crit_capture_);
	return metrics_enabled_;
	}

	// TODO(ajm): we currently just use the metrics from the first AEC. Think more
	// aboue the best way to extend this to multi-channel.
	int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
	rtc::CritScope cs(crit_capture_);
	if (metrics == NULL) {
	return AudioProcessing::kNullPointerError;
	}

	if (!enabled_ \|\| !metrics_enabled_) {
	return AudioProcessing::kNotEnabledError;
	}

	AecMetrics my_metrics;
	memset(&my_metrics, 0, sizeof(my_metrics));
	memset(metrics, 0, sizeof(Metrics));

	const int err = WebRtcAec_GetMetrics(cancellers_[0]->state(), &my_metrics);
	if (err != AudioProcessing::kNoError) {
	return MapError(err);
	}

	metrics->residual_echo_return_loss.instant = my_metrics.rerl.instant;
	metrics->residual_echo_return_loss.average = my_metrics.rerl.average;
	metrics->residual_echo_return_loss.maximum = my_metrics.rerl.max;
	metrics->residual_echo_return_loss.minimum = my_metrics.rerl.min;

	metrics->echo_return_loss.instant = my_metrics.erl.instant;
	metrics->echo_return_loss.average = my_metrics.erl.average;
	metrics->echo_return_loss.maximum = my_metrics.erl.max;
	metrics->echo_return_loss.minimum = my_metrics.erl.min;

	metrics->echo_return_loss_enhancement.instant = my_metrics.erle.instant;
	metrics->echo_return_loss_enhancement.average = my_metrics.erle.average;
	metrics->echo_return_loss_enhancement.maximum = my_metrics.erle.max;
	metrics->echo_return_loss_enhancement.minimum = my_metrics.erle.min;

	metrics->a_nlp.instant = my_metrics.aNlp.instant;
	metrics->a_nlp.average = my_metrics.aNlp.average;
	metrics->a_nlp.maximum = my_metrics.aNlp.max;
	metrics->a_nlp.minimum = my_metrics.aNlp.min;

	metrics->divergent_filter_fraction = my_metrics.divergent_filter_fraction;
	return AudioProcessing::kNoError;
	}

	bool EchoCancellationImpl::stream_has_echo() const {
	rtc::CritScope cs(crit_capture_);
	return stream_has_echo_;
	}

	int EchoCancellationImpl::enable_delay_logging(bool enable) {
	{
	rtc::CritScope cs(crit_capture_);
	delay_logging_enabled_ = enable;
	}
	return Configure();
	}

	bool EchoCancellationImpl::is_delay_logging_enabled() const {
	rtc::CritScope cs(crit_capture_);
	return delay_logging_enabled_;
	}

	bool EchoCancellationImpl::is_delay_agnostic_enabled() const {
	rtc::CritScope cs(crit_capture_);
	return delay_agnostic_enabled_;
	}

	bool EchoCancellationImpl::is_aec3_enabled() const {
	rtc::CritScope cs(crit_capture_);
	return aec3_enabled_;
	}

	std::string EchoCancellationImpl::GetExperimentsDescription() {
	rtc::CritScope cs(crit_capture_);
	std::string description = (aec3_enabled_ ? "AEC3;" : "");
	if (refined_adaptive_filter_enabled_) {
	description += "RefinedAdaptiveFilter;";
	}
	return description;
	}

	bool EchoCancellationImpl::is_refined_adaptive_filter_enabled() const {
	rtc::CritScope cs(crit_capture_);
	return refined_adaptive_filter_enabled_;
	}

	bool EchoCancellationImpl::is_extended_filter_enabled() const {
	rtc::CritScope cs(crit_capture_);
	return extended_filter_enabled_;
	}

	// TODO(bjornv): How should we handle the multi-channel case?
	int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) {
	rtc::CritScope cs(crit_capture_);
	float fraction_poor_delays = 0;
	return GetDelayMetrics(median, std, &fraction_poor_delays);
	}

	int EchoCancellationImpl::GetDelayMetrics(int* median, int* std,
	float* fraction_poor_delays) {
	rtc::CritScope cs(crit_capture_);
	if (median == NULL) {
	return AudioProcessing::kNullPointerError;
	}
	if (std == NULL) {
	return AudioProcessing::kNullPointerError;
	}

	if (!enabled_ \|\| !delay_logging_enabled_) {
	return AudioProcessing::kNotEnabledError;
	}

	const int err = WebRtcAec_GetDelayMetrics(cancellers_[0]->state(), median,
	std, fraction_poor_delays);
	if (err != AudioProcessing::kNoError) {
	return MapError(err);
	}

	return AudioProcessing::kNoError;
	}

	struct AecCore* EchoCancellationImpl::aec_core() const {
	rtc::CritScope cs(crit_capture_);
	if (!enabled_) {
	return NULL;
	}
	return WebRtcAec_aec_core(cancellers_[0]->state());
	}

	void EchoCancellationImpl::Initialize(int sample_rate_hz,
	size_t num_reverse_channels,
	size_t num_output_channels,
	size_t num_proc_channels) {
	rtc::CritScope cs_render(crit_render_);
	rtc::CritScope cs_capture(crit_capture_);

	stream_properties_.reset(
	new StreamProperties(sample_rate_hz, num_reverse_channels,
	num_output_channels, num_proc_channels));

	if (!enabled_) {
	return;
	}

	if (NumCancellersRequired() > cancellers_.size()) {
	const size_t cancellers_old_size = cancellers_.size();
	cancellers_.resize(NumCancellersRequired());

	for (size_t i = cancellers_old_size; i < cancellers_.size(); ++i) {
	cancellers_[i].reset(new Canceller());
	}
	}

	for (auto& canceller : cancellers_) {
	canceller->Initialize(sample_rate_hz);
	}

	Configure();

	AllocateRenderQueue();
	}

	int EchoCancellationImpl::GetSystemDelayInSamples() const {
	rtc::CritScope cs(crit_capture_);
	RTC_DCHECK(enabled_);
	// Report the delay for the first AEC component.
	return WebRtcAec_system_delay(
	WebRtcAec_aec_core(cancellers_[0]->state()));
	}

	void EchoCancellationImpl::AllocateRenderQueue() {
	const size_t new_render_queue_element_max_size = std::max<size_t>(
	static_cast<size_t>(1),
	kMaxAllowedValuesOfSamplesPerFrame * NumCancellersRequired());

	rtc::CritScope cs_render(crit_render_);
	rtc::CritScope cs_capture(crit_capture_);

	// Reallocate the queue if the queue item size is too small to fit the
	// data to put in the queue.
	if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
	render_queue_element_max_size_ = new_render_queue_element_max_size;

	std::vector<float> template_queue_element(render_queue_element_max_size_);

	render_signal_queue_.reset(
	new SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>(
	kMaxNumFramesToBuffer, template_queue_element,
	RenderQueueItemVerifier<float>(render_queue_element_max_size_)));

	render_queue_buffer_.resize(render_queue_element_max_size_);
	capture_queue_buffer_.resize(render_queue_element_max_size_);
	} else {
	render_signal_queue_->Clear();
	}
	}

	void EchoCancellationImpl::SetExtraOptions(const webrtc::Config& config) {
	{
	rtc::CritScope cs(crit_capture_);
	extended_filter_enabled_ = config.Get<ExtendedFilter>().enabled;
	delay_agnostic_enabled_ = config.Get<DelayAgnostic>().enabled;
	refined_adaptive_filter_enabled_ =
	config.Get<RefinedAdaptiveFilter>().enabled;
	aec3_enabled_ = config.Get<EchoCanceller3>().enabled;
	}
	Configure();
	}

	int EchoCancellationImpl::Configure() {
	rtc::CritScope cs_render(crit_render_);
	rtc::CritScope cs_capture(crit_capture_);
	AecConfig config;
	config.metricsMode = metrics_enabled_;
	config.nlpMode = MapSetting(suppression_level_);
	config.skewMode = drift_compensation_enabled_;
	config.delay_logging = delay_logging_enabled_;

	int error = AudioProcessing::kNoError;
	for (auto& canceller : cancellers_) {
	WebRtcAec_enable_extended_filter(WebRtcAec_aec_core(canceller->state()),
	extended_filter_enabled_ ? 1 : 0);
	WebRtcAec_enable_delay_agnostic(WebRtcAec_aec_core(canceller->state()),
	delay_agnostic_enabled_ ? 1 : 0);
	WebRtcAec_enable_aec3(WebRtcAec_aec_core(canceller->state()),
	aec3_enabled_ ? 1 : 0);
	WebRtcAec_enable_refined_adaptive_filter(
	WebRtcAec_aec_core(canceller->state()),
	refined_adaptive_filter_enabled_);
	const int handle_error = WebRtcAec_set_config(canceller->state(), config);
	if (handle_error != AudioProcessing::kNoError) {
	error = AudioProcessing::kNoError;
	}
	}
	return error;
	}

	size_t EchoCancellationImpl::NumCancellersRequired() const {
	RTC_DCHECK(stream_properties_);
	return stream_properties_->num_output_channels *
	stream_properties_->num_reverse_channels;
	}

	} // namespace webrtc