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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.
24 lines
751 B
C
24 lines
751 B
C
/*
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* Copyright (c) 2012 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_AGC_UTILITY_H_
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#define MODULES_AUDIO_PROCESSING_AGC_UTILITY_H_
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// TODO(turajs): Add description of function.
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double Loudness2Db(double loudness);
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double Linear2Loudness(double rms);
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double Db2Loudness(double db);
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double Dbfs2Loudness(double dbfs);
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#endif // MODULES_AUDIO_PROCESSING_AGC_UTILITY_H_
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