webrtc_m130/webrtc/modules/audio_processing/echo_cancellation_impl.h
peah 0332c2db39 Added support in the AEC for refined filter adaptation.
The following algorithmic functionality was added:
-Add support for an exact regressor power to be computed
 which avoids the issue with the updating of the filter
 sometimes being unstable.
-Lowered the fixed step size of the adaptive filter to 0.05
 which significantly reduces the sensitivity of the
 adaptive filter to near-end noise, nonlinearities,
 doubletalk and the unmodelled echo path tail. It also
 reduces the tracking speed of the adaptive filter but the
 chosen value proved to give a sufficient tradeoff for the
 requirements on the adaptive filter.

To allow the new functionality to be selectively applied the following was done:
-A new Config was added for selectively activating the functionality.
-Functionality was added in the audioprocessing  and echocancellationimpl classes
 for passing the activation of the functionality down to the AEC algorithms.

To make the code for the introduction of the functionality clean,
the following refactoring was done:
-The selection of the step size was moved to a single place.
-The constant for the step size of the adaptive filter in extended filter mode was
 made local.
-The state variable storing the step-size was renamed to a more describing name.

When the new functionality is not activated, the changes
have been tested for bitexactness on Linux.

TBR=minyue@webrtc.org
BUG=webrtc:5778, webrtc:5777

Review URL: https://codereview.webrtc.org/1887003002

Cr-Commit-Position: refs/heads/master@{#12384}
2016-04-15 18:23:36 +00:00

126 lines
4.5 KiB
C++

/*
* 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.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
#include <memory>
#include <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/swap_queue.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
namespace webrtc {
class AudioBuffer;
class EchoCancellationImpl : public EchoCancellation {
public:
EchoCancellationImpl(rtc::CriticalSection* crit_render,
rtc::CriticalSection* crit_capture);
virtual ~EchoCancellationImpl();
int ProcessRenderAudio(const AudioBuffer* audio);
int ProcessCaptureAudio(AudioBuffer* audio, int stream_delay_ms);
// EchoCancellation implementation.
bool is_enabled() const override;
int stream_drift_samples() const override;
SuppressionLevel suppression_level() const override;
bool is_drift_compensation_enabled() const override;
void Initialize(int sample_rate_hz,
size_t num_reverse_channels_,
size_t num_output_channels_,
size_t num_proc_channels_);
void SetExtraOptions(const Config& config);
bool is_delay_agnostic_enabled() const;
bool is_extended_filter_enabled() const;
bool is_aec3_enabled() const;
std::string GetExperimentsDescription();
bool is_refined_adaptive_filter_enabled() const;
// Checks whether the module is enabled. Must only be
// called from the render side of APM as otherwise
// deadlocks may occur.
bool is_enabled_render_side_query() const;
// Reads render side data that has been queued on the render call.
// Called holding the capture lock.
void ReadQueuedRenderData();
// Returns the system delay of the first AEC component.
int GetSystemDelayInSamples() const;
private:
class Canceller;
struct StreamProperties;
// EchoCancellation implementation.
int Enable(bool enable) override;
int enable_drift_compensation(bool enable) override;
void set_stream_drift_samples(int drift) override;
int set_suppression_level(SuppressionLevel level) override;
int enable_metrics(bool enable) override;
bool are_metrics_enabled() const override;
bool stream_has_echo() const override;
int GetMetrics(Metrics* metrics) override;
int enable_delay_logging(bool enable) override;
bool is_delay_logging_enabled() const override;
int GetDelayMetrics(int* median, int* std) override;
int GetDelayMetrics(int* median,
int* std,
float* fraction_poor_delays) override;
struct AecCore* aec_core() const override;
size_t NumCancellersRequired() const;
void AllocateRenderQueue();
int Configure();
rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
rtc::CriticalSection* const crit_capture_;
bool enabled_ = false;
bool drift_compensation_enabled_ GUARDED_BY(crit_capture_);
bool metrics_enabled_ GUARDED_BY(crit_capture_);
SuppressionLevel suppression_level_ GUARDED_BY(crit_capture_);
int stream_drift_samples_ GUARDED_BY(crit_capture_);
bool was_stream_drift_set_ GUARDED_BY(crit_capture_);
bool stream_has_echo_ GUARDED_BY(crit_capture_);
bool delay_logging_enabled_ GUARDED_BY(crit_capture_);
bool extended_filter_enabled_ GUARDED_BY(crit_capture_);
bool delay_agnostic_enabled_ GUARDED_BY(crit_capture_);
bool aec3_enabled_ GUARDED_BY(crit_capture_);
bool refined_adaptive_filter_enabled_ GUARDED_BY(crit_capture_) = false;
size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
GUARDED_BY(crit_capture_);
std::vector<float> render_queue_buffer_ GUARDED_BY(crit_render_);
std::vector<float> capture_queue_buffer_ GUARDED_BY(crit_capture_);
// Lock protection not needed.
std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
render_signal_queue_;
std::vector<std::unique_ptr<Canceller>> cancellers_;
std::unique_ptr<StreamProperties> stream_properties_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EchoCancellationImpl);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_