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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.
28 lines
859 B
C++
28 lines
859 B
C++
/*
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* Copyright (c) 2014 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_TRANSIENT_COMMON_H_
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#define MODULES_AUDIO_PROCESSING_TRANSIENT_COMMON_H_
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namespace webrtc {
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namespace ts {
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static const float kPi = 3.14159265358979323846f;
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static const int kChunkSizeMs = 10;
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enum {
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kSampleRate8kHz = 8000,
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kSampleRate16kHz = 16000,
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kSampleRate32kHz = 32000,
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kSampleRate48kHz = 48000
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};
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} // namespace ts
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_TRANSIENT_COMMON_H_
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