| Index: webrtc/modules/audio_processing/aec3/echo_remover.h
|
| diff --git a/webrtc/modules/audio_processing/aec3/echo_remover.h b/webrtc/modules/audio_processing/aec3/echo_remover.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..f7ac50cec7b14584c204a4c1886307a4d343aa9b
|
| --- /dev/null
|
| +++ b/webrtc/modules/audio_processing/aec3/echo_remover.h
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| @@ -0,0 +1,44 @@
|
| +/*
|
| + * 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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_
|
| +#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_
|
| +
|
| +#include <vector>
|
| +
|
| +#include "webrtc/base/optional.h"
|
| +#include "webrtc/modules/audio_processing/aec3/echo_path_variability.h"
|
| +
|
| +namespace webrtc {
|
| +
|
| +// Class for removing the echo from the capture signal.
|
| +class EchoRemover {
|
| + public:
|
| + static EchoRemover* Create(int sample_rate_hz);
|
| + virtual ~EchoRemover() = default;
|
| +
|
| + // 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 ProcessBlock(
|
| + const rtc::Optional<size_t>& echo_path_delay_samples,
|
| + const EchoPathVariability& echo_path_variability,
|
| + bool capture_signal_saturation,
|
| + const std::vector<std::vector<float>>& render,
|
| + std::vector<std::vector<float>>* 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;
|
| +};
|
| +
|
| +} // namespace webrtc
|
| +
|
| +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_
|
|
|