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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
56 lines
1.6 KiB
C++
56 lines
1.6 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_AUDIO_ECHO_CONTROL_H_
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#define API_AUDIO_ECHO_CONTROL_H_
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#include <memory>
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namespace webrtc {
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class AudioBuffer;
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// Interface for an acoustic echo cancellation (AEC) submodule.
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class EchoControl {
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public:
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// Analysis (not changing) of the render signal.
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virtual void AnalyzeRender(AudioBuffer* render) = 0;
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// Analysis (not changing) of the capture signal.
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virtual void AnalyzeCapture(AudioBuffer* capture) = 0;
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// Processes the capture signal in order to remove the echo.
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virtual void ProcessCapture(AudioBuffer* capture, bool echo_path_change) = 0;
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struct Metrics {
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double echo_return_loss;
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double echo_return_loss_enhancement;
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int delay_ms;
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};
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// Collect current metrics from the echo controller.
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virtual Metrics GetMetrics() const = 0;
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// Provides an optional external estimate of the audio buffer delay.
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virtual void SetAudioBufferDelay(size_t delay_ms) = 0;
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virtual ~EchoControl() {}
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};
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// Interface for a factory that creates EchoControllers.
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class EchoControlFactory {
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public:
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virtual std::unique_ptr<EchoControl> Create(int sample_rate_hz) = 0;
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virtual ~EchoControlFactory() = default;
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};
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} // namespace webrtc
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#endif // API_AUDIO_ECHO_CONTROL_H_
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