Chromium Code Reviews| OLD | NEW |
|---|---|
| 1 /* | 1 /* |
| 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
| 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_ |
| 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_ | 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_ |
| 13 | 13 |
| 14 #include <string> | |
| 15 | |
| 16 #include "webrtc/base/constructormagic.h" | 14 #include "webrtc/base/constructormagic.h" |
| 15 #include "webrtc/modules/audio_processing/aec3/block_framer.h" | |
| 16 #include "webrtc/modules/audio_processing/aec3/block_processor.h" | |
| 17 #include "webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.h" | |
| 18 #include "webrtc/modules/audio_processing/aec3/frame_blocker.h" | |
| 19 #include "webrtc/modules/audio_processing/aec3/render_transfer_buffer.h" | |
| 17 #include "webrtc/modules/audio_processing/audio_buffer.h" | 20 #include "webrtc/modules/audio_processing/audio_buffer.h" |
| 21 #include "webrtc/modules/audio_processing/include/audio_processing.h" | |
| 22 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" | |
| 18 | 23 |
| 19 namespace webrtc { | 24 namespace webrtc { |
| 20 | 25 |
| 26 // Main class for the echo canceller3. It does 4 things: | |
| 27 // -Receives 10 ms frames of band-split audio. | |
| 28 // -Optionally applies an anti-hum (high-pass) filter on the | |
| 29 // received signals. | |
| 30 // -Provides the lower level echo canceller functionality with | |
| 31 // blocks of 64 samples of audio data. | |
| 32 // -Partially handles the jitter in the render and capture API | |
| 33 // call sequence. | |
| 34 // The class is supposed to be used in a single-threaded manner | |
|
hlundin-webrtc
2016/12/16 10:04:47
What does single-threaded mean here? Should all me
peah-webrtc
2016/12/20 10:10:25
Good point. I changed the comment to be more clear
hlundin-webrtc
2016/12/20 15:10:34
I propose you use a RaceChecker, similarly to how
peah-webrtc
2016/12/21 23:13:48
I agree!
Done.
| |
| 35 // apart from the AnalyzeRender call which can be performed | |
| 36 // from another thread. | |
| 21 class EchoCanceller3 { | 37 class EchoCanceller3 { |
| 22 public: | 38 public: |
| 23 EchoCanceller3(int sample_rate_hz, bool use_anti_hum_filter); | 39 EchoCanceller3(int sample_rate_hz, bool use_highpass_filter); |
| 24 ~EchoCanceller3(); | 40 ~EchoCanceller3(); |
| 25 // Analyzes and stores an internal copy of the split-band domain render | 41 // Analyzes and stores an internal copy of the split-band domain render |
| 26 // signal. | 42 // signal. |
| 27 bool AnalyzeRender(AudioBuffer* farend); | 43 bool AnalyzeRender(AudioBuffer* farend); |
| 28 // Analyzes the full-band domain capture signal to detect signal saturation. | 44 // Analyzes the full-band domain capture signal to detect signal saturation. |
| 29 void AnalyzeCapture(AudioBuffer* capture); | 45 void AnalyzeCapture(AudioBuffer* capture); |
| 30 // Processes the split-band domain capture signal in order to remove any echo | 46 // Processes the split-band domain capture signal in order to remove any echo |
| 31 // present in the signal. | 47 // present in the signal. |
| 32 void ProcessCapture(AudioBuffer* capture, bool known_echo_path_change); | 48 void ProcessCapture(AudioBuffer* capture, bool known_echo_path_change); |
| 33 | 49 |
| 50 // Signals whether an external detector has detected echo leakage from the | |
| 51 // echo canceller. | |
| 52 // Note that in the case echo leakage has been flagged, it should be unflagged | |
| 53 // once it is no longer occurring. | |
| 54 void ReportEchoLeakage(bool leakage_detected) { | |
| 55 block_processor_.ReportEchoLeakage(leakage_detected); | |
| 56 } | |
| 57 | |
| 34 // Validates a config. | 58 // Validates a config. |
| 35 static bool Validate(const AudioProcessing::Config::EchoCanceller3& config); | 59 static bool Validate(const AudioProcessing::Config::EchoCanceller3& config); |
| 36 // Dumps a config to a string. | 60 // Dumps a config to a string. |
| 37 static std::string ToString( | 61 static std::string ToString( |
| 38 const AudioProcessing::Config::EchoCanceller3& config); | 62 const AudioProcessing::Config::EchoCanceller3& config); |
| 39 | 63 |
| 40 private: | 64 private: |
| 65 class RenderWriterState { | |
|
hlundin-webrtc
2016/12/16 10:04:47
"State" suggests that this is a simple data contai
ivoc
2016/12/19 11:30:22
Since this class is private and EchoCanceller3 onl
peah-webrtc
2016/12/20 10:10:25
Good point!
Done.
peah-webrtc
2016/12/20 10:10:25
Done.
| |
| 66 public: | |
| 67 RenderWriterState( | |
| 68 ApmDataDumper* data_dumper, | |
| 69 RenderTransferBufferWriter* transfer_buffer_writer, | |
| 70 std::unique_ptr<CascadedBiQuadFilter> render_highpass_filter, | |
| 71 int sample_rate_hz, | |
| 72 int frame_length, | |
| 73 int num_bands); | |
| 74 ~RenderWriterState(); | |
| 75 bool Insert(AudioBuffer* render); | |
| 76 | |
| 77 private: | |
| 78 ApmDataDumper* data_dumper_; | |
| 79 const int sample_rate_hz_; | |
| 80 const size_t frame_length_; | |
| 81 const int num_bands_; | |
| 82 RenderTransferBufferWriter* transfer_buffer_writer_; | |
| 83 std::unique_ptr<CascadedBiQuadFilter> render_highpass_filter_; | |
| 84 std::vector<float> render_queue_input_frame_; | |
| 85 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderWriterState); | |
| 86 }; | |
| 87 // State that may be accessed by the render thread. | |
| 88 std::unique_ptr<RenderWriterState> render_writer_; | |
| 89 | |
| 90 // State that may be accessed by the capture thread. | |
| 41 static int instance_count_; | 91 static int instance_count_; |
| 42 size_t frame_length_; | 92 std::unique_ptr<ApmDataDumper> data_dumper_; |
| 93 const int sample_rate_hz_; | |
| 94 const int num_bands_; | |
| 95 const size_t frame_length_; | |
| 96 BlockFramer output_framer_; | |
| 97 FrameBlocker capture_blocker_; | |
| 98 FrameBlocker render_blocker_; | |
| 99 RenderTransferBuffer render_transfer_buffer_; | |
| 100 BlockProcessor block_processor_; | |
| 101 std::vector<float> render_queue_output_frame_; | |
| 102 std::unique_ptr<CascadedBiQuadFilter> capture_highpass_filter_; | |
| 103 bool saturated_microphone_signal_ = false; | |
| 104 bool EmptyRenderQueue(); | |
|
hlundin-webrtc
2016/12/16 10:04:47
Declare methods before data members.
https://googl
peah-webrtc
2016/12/20 10:10:25
Done.
| |
| 43 | 105 |
| 44 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EchoCanceller3); | 106 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EchoCanceller3); |
| 45 }; | 107 }; |
| 46 } // namespace webrtc | 108 } // namespace webrtc |
| 47 | 109 |
| 48 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_ | 110 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_ |
| OLD | NEW |