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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.
23 lines
813 B
C++
23 lines
813 B
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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#include "modules/audio_processing/aec3/echo_path_variability.h"
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namespace webrtc {
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EchoPathVariability::EchoPathVariability(bool gain_change,
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DelayAdjustment delay_change,
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bool clock_drift)
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: gain_change(gain_change),
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delay_change(delay_change),
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clock_drift(clock_drift) {}
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} // namespace webrtc
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