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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.
32 lines
932 B
C++
32 lines
932 B
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 MODULES_AUDIO_PROCESSING_AEC3_DELAY_ESTIMATE_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_DELAY_ESTIMATE_H_
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namespace webrtc {
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// Stores delay_estimates.
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struct DelayEstimate {
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enum class Quality { kCoarse, kRefined };
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DelayEstimate(Quality quality, size_t delay)
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: quality(quality), delay(delay) {}
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Quality quality;
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size_t delay;
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size_t blocks_since_last_change = 0;
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size_t blocks_since_last_update = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_DELAY_ESTIMATE_H_
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