| /* |
| * 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_RESIDUAL_ECHO_DETECTOR_H_ |
| #define WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_ |
| |
| #include <vector> |
| |
| #include "webrtc/modules/audio_processing/echo_detector/circular_buffer.h" |
| #include "webrtc/modules/audio_processing/echo_detector/mean_variance_estimator.h" |
| #include "webrtc/modules/audio_processing/echo_detector/moving_max.h" |
| #include "webrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.h" |
| #include "webrtc/rtc_base/array_view.h" |
| |
| namespace webrtc { |
| |
| class ApmDataDumper; |
| class AudioBuffer; |
| class EchoDetector; |
| |
| class ResidualEchoDetector { |
| public: |
| ResidualEchoDetector(); |
| ~ResidualEchoDetector(); |
| |
| // This function should be called while holding the render lock. |
| void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio); |
| |
| // This function should be called while holding the capture lock. |
| void AnalyzeCaptureAudio(rtc::ArrayView<const float> capture_audio); |
| |
| // This function should be called while holding the capture lock. |
| void Initialize(); |
| |
| // This function is for testing purposes only. |
| void SetReliabilityForTest(float value) { reliability_ = value; } |
| |
| static void PackRenderAudioBuffer(AudioBuffer* audio, |
| std::vector<float>* packed_buffer); |
| |
| // This function should be called while holding the capture lock. |
| float echo_likelihood() const { return echo_likelihood_; } |
| |
| float echo_likelihood_recent_max() const { |
| return recent_likelihood_max_.max(); |
| } |
| |
| private: |
| static int instance_count_; |
| std::unique_ptr<ApmDataDumper> data_dumper_; |
| // Keep track if the |Process| function has been previously called. |
| bool first_process_call_ = true; |
| // Buffer for storing the power of incoming farend buffers. This is needed for |
| // cases where calls to BufferFarend and Process are jittery. |
| CircularBuffer render_buffer_; |
| // Count how long ago it was that the size of |render_buffer_| was zero. This |
| // value is also reset to zero when clock drift is detected and a value from |
| // the renderbuffer is discarded, even though the buffer is not actually zero |
| // at that point. This is done to avoid repeatedly removing elements in this |
| // situation. |
| size_t frames_since_zero_buffer_size_ = 0; |
| |
| // Circular buffers containing delayed versions of the power, mean and |
| // standard deviation, for calculating the delayed covariance values. |
| std::vector<float> render_power_; |
| std::vector<float> render_power_mean_; |
| std::vector<float> render_power_std_dev_; |
| // Covariance estimates for different delay values. |
| std::vector<NormalizedCovarianceEstimator> covariances_; |
| // Index where next element should be inserted in all of the above circular |
| // buffers. |
| size_t next_insertion_index_ = 0; |
| |
| MeanVarianceEstimator render_statistics_; |
| MeanVarianceEstimator capture_statistics_; |
| // Current echo likelihood. |
| float echo_likelihood_ = 0.f; |
| // Reliability of the current likelihood. |
| float reliability_ = 0.f; |
| MovingMax recent_likelihood_max_; |
| |
| int log_counter_ = 0; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_ |