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functionalities doing sample-rate conversions: -The implicit resampling done in the AudioBuffer CopyTo, CopyFrom, InterleaveTo and DeinterleaveFrom methods. -The multi-band splitting scheme. The selection of rates in these have been difficult and complicated, partly due to that the APM API which allows for activating the APM submodules without notifying the APM. This CL adds functionality that for each capture frame polls all submodules for whether they are active or not and compares this against a cached result. Furthermore, new functionality is added that based on the results of the comparison do a reinitialization of the APM. This has several advantages -The code deciding on whether to analysis and synthesis is needed for the bandsplitting can be much simplified and centralized. -The selection of the processing rate can be done such as to avoid the implicit resampling that was in some cases unnecessarily done. -The optimization for whether an output copy is needed that was done to improve performance due to the implicit resampling is no longer needed, which simplifies the code and makes it less error-prone in the sense that is no longer neccessary to keep track of whether any module has changed the signal. Finally, it should be noted that the polling of the state for all the submodules was done previously as well, but in a less obvious and distributed manner. BUG=webrtc:6181, webrtc:6220, webrtc:5298, webrtc:6296, webrtc:6298, webrtc:6297 Review-Url: https://codereview.webrtc.org/2304123002 Cr-Commit-Position: refs/heads/master@{#14175}
121 lines
4.4 KiB
C++
121 lines
4.4 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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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
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#include <memory>
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#include <vector>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/swap_queue.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
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namespace webrtc {
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class AudioBuffer;
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class EchoCancellationImpl : public EchoCancellation {
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public:
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EchoCancellationImpl(rtc::CriticalSection* crit_render,
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rtc::CriticalSection* crit_capture);
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~EchoCancellationImpl() override;
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int ProcessRenderAudio(const AudioBuffer* audio);
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int ProcessCaptureAudio(AudioBuffer* audio, int stream_delay_ms);
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// EchoCancellation implementation.
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bool is_enabled() const override;
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int stream_drift_samples() const override;
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SuppressionLevel suppression_level() const override;
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bool is_drift_compensation_enabled() const override;
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void Initialize(int sample_rate_hz,
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size_t num_reverse_channels_,
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size_t num_output_channels_,
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size_t num_proc_channels_);
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void SetExtraOptions(const webrtc::Config& config);
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bool is_delay_agnostic_enabled() const;
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bool is_extended_filter_enabled() const;
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bool is_aec3_enabled() const;
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std::string GetExperimentsDescription();
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bool is_refined_adaptive_filter_enabled() const;
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// Reads render side data that has been queued on the render call.
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// Called holding the capture lock.
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void ReadQueuedRenderData();
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// Returns the system delay of the first AEC component.
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int GetSystemDelayInSamples() const;
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private:
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class Canceller;
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struct StreamProperties;
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// EchoCancellation implementation.
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int Enable(bool enable) override;
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int enable_drift_compensation(bool enable) override;
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void set_stream_drift_samples(int drift) override;
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int set_suppression_level(SuppressionLevel level) override;
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int enable_metrics(bool enable) override;
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bool are_metrics_enabled() const override;
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bool stream_has_echo() const override;
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int GetMetrics(Metrics* metrics) override;
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int enable_delay_logging(bool enable) override;
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bool is_delay_logging_enabled() const override;
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int GetDelayMetrics(int* median, int* std) override;
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int GetDelayMetrics(int* median,
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int* std,
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float* fraction_poor_delays) override;
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struct AecCore* aec_core() const override;
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size_t NumCancellersRequired() const;
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void AllocateRenderQueue();
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int Configure();
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rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
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rtc::CriticalSection* const crit_capture_;
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bool enabled_ = false;
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bool drift_compensation_enabled_ GUARDED_BY(crit_capture_);
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bool metrics_enabled_ GUARDED_BY(crit_capture_);
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SuppressionLevel suppression_level_ GUARDED_BY(crit_capture_);
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int stream_drift_samples_ GUARDED_BY(crit_capture_);
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bool was_stream_drift_set_ GUARDED_BY(crit_capture_);
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bool stream_has_echo_ GUARDED_BY(crit_capture_);
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bool delay_logging_enabled_ GUARDED_BY(crit_capture_);
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bool extended_filter_enabled_ GUARDED_BY(crit_capture_);
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bool delay_agnostic_enabled_ GUARDED_BY(crit_capture_);
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bool aec3_enabled_ GUARDED_BY(crit_capture_);
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bool refined_adaptive_filter_enabled_ GUARDED_BY(crit_capture_) = false;
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size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
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GUARDED_BY(crit_capture_);
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std::vector<float> render_queue_buffer_ GUARDED_BY(crit_render_);
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std::vector<float> capture_queue_buffer_ GUARDED_BY(crit_capture_);
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// Lock protection not needed.
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std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
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render_signal_queue_;
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std::vector<std::unique_ptr<Canceller>> cancellers_;
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std::unique_ptr<StreamProperties> stream_properties_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EchoCancellationImpl);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
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