/* * Copyright (c) 2017 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_AEC3_ECHO_REMOVER_H_ #define MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_ #include #include "absl/types/optional.h" #include "api/audio/echo_canceller3_config.h" #include "api/audio/echo_control.h" #include "modules/audio_processing/aec3/block.h" #include "modules/audio_processing/aec3/delay_estimate.h" #include "modules/audio_processing/aec3/echo_path_variability.h" #include "modules/audio_processing/aec3/render_buffer.h" namespace webrtc { // Class for removing the echo from the capture signal. class EchoRemover { public: static EchoRemover* Create(const EchoCanceller3Config& config, int sample_rate_hz, size_t num_render_channels, size_t num_capture_channels); virtual ~EchoRemover() = default; // Get current metrics. virtual void GetMetrics(EchoControl::Metrics* metrics) const = 0; // Removes the echo from a block of samples from the capture signal. The // supplied render signal is assumed to be pre-aligned with the capture // signal. virtual void ProcessCapture( EchoPathVariability echo_path_variability, bool capture_signal_saturation, const absl::optional& external_delay, RenderBuffer* render_buffer, Block* linear_output, Block* capture) = 0; // Updates the status on whether echo leakage is detected in the output of the // echo remover. virtual void UpdateEchoLeakageStatus(bool leakage_detected) = 0; // Specifies whether the capture output will be used. The purpose of this is // to allow the echo remover to deactivate some of the processing when the // resulting output is anyway not used, for instance when the endpoint is // muted. virtual void SetCaptureOutputUsage(bool capture_output_used) = 0; }; } // namespace webrtc #endif // MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_