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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
1.9 KiB
C++
56 lines
1.9 KiB
C++
/*
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* Copyright (c) 2017 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 MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/audio/echo_canceller3_config.h"
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#include "api/audio/echo_control.h"
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#include "modules/audio_processing/aec3/delay_estimate.h"
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#include "modules/audio_processing/aec3/echo_path_variability.h"
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#include "modules/audio_processing/aec3/render_buffer.h"
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namespace webrtc {
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// Class for removing the echo from the capture signal.
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class EchoRemover {
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public:
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static EchoRemover* Create(const EchoCanceller3Config& config,
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int sample_rate_hz);
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virtual ~EchoRemover() = default;
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// Get current metrics.
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virtual void GetMetrics(EchoControl::Metrics* metrics) const = 0;
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// Removes the echo from a block of samples from the capture signal. The
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// supplied render signal is assumed to be pre-aligned with the capture
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// signal.
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virtual void ProcessCapture(
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EchoPathVariability echo_path_variability,
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bool capture_signal_saturation,
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const absl::optional<DelayEstimate>& external_delay,
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RenderBuffer* render_buffer,
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std::vector<std::vector<float>>* capture) = 0;
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// Returns the internal delay estimate in blocks.
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virtual absl::optional<int> Delay() const = 0;
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// Updates the status on whether echo leakage is detected in the output of the
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// echo remover.
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virtual void UpdateEchoLeakageStatus(bool leakage_detected) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_
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