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/*
* 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 <vector>
#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<DelayEstimate>& 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_
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