| /* |
| * 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 MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_ |
| #define MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_ |
| |
| #include <vector> |
| |
| #include "api/array_view.h" |
| #include "modules/audio_processing/echo_detector/circular_buffer.h" |
| #include "modules/audio_processing/echo_detector/mean_variance_estimator.h" |
| #include "modules/audio_processing/echo_detector/moving_max.h" |
| #include "modules/audio_processing/echo_detector/normalized_covariance_estimator.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| |
| namespace webrtc { |
| |
| class ApmDataDumper; |
| class AudioBuffer; |
| |
| class ResidualEchoDetector : public EchoDetector { |
| public: |
| ResidualEchoDetector(); |
| ~ResidualEchoDetector() override; |
| |
| // This function should be called while holding the render lock. |
| void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio) override; |
| |
| // This function should be called while holding the capture lock. |
| void AnalyzeCaptureAudio(rtc::ArrayView<const float> capture_audio) override; |
| |
| // This function should be called while holding the capture lock. |
| void Initialize(int capture_sample_rate_hz, |
| int num_capture_channels, |
| int render_sample_rate_hz, |
| int num_render_channels) override; |
| |
| // This function is for testing purposes only. |
| void SetReliabilityForTest(float value) { reliability_ = value; } |
| |
| // This function should be called while holding the capture lock. |
| EchoDetector::Metrics GetMetrics() const override; |
| |
| 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 // MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_ |