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.
149 lines
5.6 KiB
C++
149 lines
5.6 KiB
C++
/*
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* Copyright (c) 2016 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 MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_
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#include <stddef.h>
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#include <memory>
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#include <vector>
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#include "api/array_view.h"
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#include "api/audio/echo_canceller3_config.h"
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#include "api/audio/echo_control.h"
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#include "modules/audio_processing/aec3/block_delay_buffer.h"
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#include "modules/audio_processing/aec3/block_framer.h"
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#include "modules/audio_processing/aec3/block_processor.h"
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#include "modules/audio_processing/aec3/cascaded_biquad_filter.h"
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#include "modules/audio_processing/aec3/frame_blocker.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/race_checker.h"
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#include "rtc_base/swap_queue.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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// Functor for verifying the invariance of the frames being put into the render
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// queue.
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class Aec3RenderQueueItemVerifier {
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public:
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explicit Aec3RenderQueueItemVerifier(size_t num_bands, size_t frame_length)
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: num_bands_(num_bands), frame_length_(frame_length) {}
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bool operator()(const std::vector<std::vector<float>>& v) const {
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if (v.size() != num_bands_) {
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return false;
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}
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for (const auto& v_k : v) {
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if (v_k.size() != frame_length_) {
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return false;
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}
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}
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return true;
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}
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private:
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const size_t num_bands_;
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const size_t frame_length_;
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};
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// Main class for the echo canceller3.
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// It does 4 things:
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// -Receives 10 ms frames of band-split audio.
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// -Optionally applies an anti-hum (high-pass) filter on the
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// received signals.
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// -Provides the lower level echo canceller functionality with
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// blocks of 64 samples of audio data.
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// -Partially handles the jitter in the render and capture API
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// call sequence.
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//
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// The class is supposed to be used in a non-concurrent manner apart from the
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// AnalyzeRender call which can be called concurrently with the other methods.
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class EchoCanceller3 : public EchoControl {
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public:
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// Normal c-tor to use.
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EchoCanceller3(const EchoCanceller3Config& config,
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int sample_rate_hz,
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bool use_highpass_filter);
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// Testing c-tor that is used only for testing purposes.
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EchoCanceller3(const EchoCanceller3Config& config,
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int sample_rate_hz,
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bool use_highpass_filter,
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std::unique_ptr<BlockProcessor> block_processor);
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~EchoCanceller3() override;
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// Analyzes and stores an internal copy of the split-band domain render
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// signal.
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void AnalyzeRender(AudioBuffer* farend) override;
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// Analyzes the full-band domain capture signal to detect signal saturation.
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void AnalyzeCapture(AudioBuffer* capture) override;
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// Processes the split-band domain capture signal in order to remove any echo
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// present in the signal.
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void ProcessCapture(AudioBuffer* capture, bool level_change) override;
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// Collect current metrics from the echo canceller.
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Metrics GetMetrics() const override;
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// Provides an optional external estimate of the audio buffer delay.
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void SetAudioBufferDelay(size_t delay_ms) override;
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// Signals whether an external detector has detected echo leakage from the
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// echo canceller.
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// Note that in the case echo leakage has been flagged, it should be unflagged
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// once it is no longer occurring.
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void UpdateEchoLeakageStatus(bool leakage_detected) {
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RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
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block_processor_->UpdateEchoLeakageStatus(leakage_detected);
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}
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private:
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class RenderWriter;
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// Empties the render SwapQueue.
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void EmptyRenderQueue();
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rtc::RaceChecker capture_race_checker_;
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rtc::RaceChecker render_race_checker_;
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// State that is accessed by the AnalyzeRender call.
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std::unique_ptr<RenderWriter> render_writer_
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RTC_GUARDED_BY(render_race_checker_);
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// State that may be accessed by the capture thread.
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static int instance_count_;
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std::unique_ptr<ApmDataDumper> data_dumper_;
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const EchoCanceller3Config config_;
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const int sample_rate_hz_;
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const int num_bands_;
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const size_t frame_length_;
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BlockFramer output_framer_ RTC_GUARDED_BY(capture_race_checker_);
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FrameBlocker capture_blocker_ RTC_GUARDED_BY(capture_race_checker_);
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FrameBlocker render_blocker_ RTC_GUARDED_BY(capture_race_checker_);
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SwapQueue<std::vector<std::vector<float>>, Aec3RenderQueueItemVerifier>
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render_transfer_queue_;
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std::unique_ptr<BlockProcessor> block_processor_
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RTC_GUARDED_BY(capture_race_checker_);
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std::vector<std::vector<float>> render_queue_output_frame_
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RTC_GUARDED_BY(capture_race_checker_);
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std::unique_ptr<CascadedBiQuadFilter> capture_highpass_filter_
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RTC_GUARDED_BY(capture_race_checker_);
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bool saturated_microphone_signal_ RTC_GUARDED_BY(capture_race_checker_) =
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false;
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std::vector<std::vector<float>> block_ RTC_GUARDED_BY(capture_race_checker_);
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std::vector<rtc::ArrayView<float>> sub_frame_view_
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RTC_GUARDED_BY(capture_race_checker_);
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BlockDelayBuffer block_delay_buffer_ RTC_GUARDED_BY(capture_race_checker_);
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EchoCanceller3);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_
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