| /* |
| * 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. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_ |
| #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_ |
| |
| #include "webrtc/modules/audio_processing/aec3/block_framer.h" |
| #include "webrtc/modules/audio_processing/aec3/block_processor.h" |
| #include "webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.h" |
| #include "webrtc/modules/audio_processing/aec3/frame_blocker.h" |
| #include "webrtc/modules/audio_processing/audio_buffer.h" |
| #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" |
| #include "webrtc/rtc_base/constructormagic.h" |
| #include "webrtc/rtc_base/race_checker.h" |
| #include "webrtc/rtc_base/swap_queue.h" |
| |
| namespace webrtc { |
| |
| // Functor for verifying the invariance of the frames being put into the render |
| // queue. |
| class Aec3RenderQueueItemVerifier { |
| public: |
| explicit Aec3RenderQueueItemVerifier(size_t num_bands, size_t frame_length) |
| : num_bands_(num_bands), frame_length_(frame_length) {} |
| |
| bool operator()(const std::vector<std::vector<float>>& v) const { |
| if (v.size() != num_bands_) { |
| return false; |
| } |
| for (const auto& v_k : v) { |
| if (v_k.size() != frame_length_) { |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| private: |
| const size_t num_bands_; |
| const size_t frame_length_; |
| }; |
| |
| // Main class for the echo canceller3. |
| // It does 4 things: |
| // -Receives 10 ms frames of band-split audio. |
| // -Optionally applies an anti-hum (high-pass) filter on the |
| // received signals. |
| // -Provides the lower level echo canceller functionality with |
| // blocks of 64 samples of audio data. |
| // -Partially handles the jitter in the render and capture API |
| // call sequence. |
| // |
| // The class is supposed to be used in a non-concurrent manner apart from the |
| // AnalyzeRender call which can be called concurrently with the other methods. |
| class EchoCanceller3 { |
| public: |
| // Normal c-tor to use. |
| EchoCanceller3(const AudioProcessing::Config::EchoCanceller3& config, |
| int sample_rate_hz, |
| bool use_highpass_filter); |
| // Testing c-tor that is used only for testing purposes. |
| EchoCanceller3(int sample_rate_hz, |
| bool use_highpass_filter, |
| std::unique_ptr<BlockProcessor> block_processor); |
| ~EchoCanceller3(); |
| // Analyzes and stores an internal copy of the split-band domain render |
| // signal. |
| void AnalyzeRender(AudioBuffer* farend); |
| // Analyzes the full-band domain capture signal to detect signal saturation. |
| void AnalyzeCapture(AudioBuffer* capture); |
| // Processes the split-band domain capture signal in order to remove any echo |
| // present in the signal. |
| void ProcessCapture(AudioBuffer* capture, bool level_change); |
| |
| // Signals whether an external detector has detected echo leakage from the |
| // echo canceller. |
| // Note that in the case echo leakage has been flagged, it should be unflagged |
| // once it is no longer occurring. |
| void UpdateEchoLeakageStatus(bool leakage_detected) { |
| RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_); |
| block_processor_->UpdateEchoLeakageStatus(leakage_detected); |
| } |
| |
| // Validates a config. |
| static bool Validate(const AudioProcessing::Config::EchoCanceller3& config); |
| // Dumps a config to a string. |
| static std::string ToString( |
| const AudioProcessing::Config::EchoCanceller3& config); |
| |
| private: |
| class RenderWriter; |
| |
| // Empties the render SwapQueue. |
| void EmptyRenderQueue(); |
| |
| rtc::RaceChecker capture_race_checker_; |
| rtc::RaceChecker render_race_checker_; |
| |
| // State that is accessed by the AnalyzeRender call. |
| std::unique_ptr<RenderWriter> render_writer_ |
| RTC_GUARDED_BY(render_race_checker_); |
| |
| // State that may be accessed by the capture thread. |
| static int instance_count_; |
| std::unique_ptr<ApmDataDumper> data_dumper_; |
| const int sample_rate_hz_; |
| const int num_bands_; |
| const size_t frame_length_; |
| BlockFramer output_framer_ RTC_GUARDED_BY(capture_race_checker_); |
| FrameBlocker capture_blocker_ RTC_GUARDED_BY(capture_race_checker_); |
| FrameBlocker render_blocker_ RTC_GUARDED_BY(capture_race_checker_); |
| SwapQueue<std::vector<std::vector<float>>, Aec3RenderQueueItemVerifier> |
| render_transfer_queue_; |
| std::unique_ptr<BlockProcessor> block_processor_ |
| RTC_GUARDED_BY(capture_race_checker_); |
| std::vector<std::vector<float>> render_queue_output_frame_ |
| RTC_GUARDED_BY(capture_race_checker_); |
| std::unique_ptr<CascadedBiQuadFilter> capture_highpass_filter_ |
| RTC_GUARDED_BY(capture_race_checker_); |
| bool saturated_microphone_signal_ RTC_GUARDED_BY(capture_race_checker_) = |
| false; |
| std::vector<std::vector<float>> block_ RTC_GUARDED_BY(capture_race_checker_); |
| std::vector<rtc::ArrayView<float>> sub_frame_view_ |
| RTC_GUARDED_BY(capture_race_checker_); |
| |
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EchoCanceller3); |
| }; |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_ |