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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.
40 lines
1.1 KiB
C++
40 lines
1.1 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_PATH_VARIABILITY_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_VARIABILITY_H_
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namespace webrtc {
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struct EchoPathVariability {
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enum class DelayAdjustment {
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kNone,
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kBufferReadjustment,
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kBufferFlush,
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kDelayReset,
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kNewDetectedDelay
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};
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EchoPathVariability(bool gain_change,
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DelayAdjustment delay_change,
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bool clock_drift);
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bool AudioPathChanged() const {
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return gain_change || delay_change != DelayAdjustment::kNone;
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}
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bool gain_change;
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DelayAdjustment delay_change;
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bool clock_drift;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_VARIABILITY_H_
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