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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.
36 lines
1019 B
C++
36 lines
1019 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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#include "modules/audio_processing/agc/utility.h"
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#include <math.h>
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static const double kLog10 = 2.30258509299;
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static const double kLinear2DbScale = 20.0 / kLog10;
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static const double kLinear2LoudnessScale = 13.4 / kLog10;
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double Loudness2Db(double loudness) {
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return loudness * kLinear2DbScale / kLinear2LoudnessScale;
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}
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double Linear2Loudness(double rms) {
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if (rms == 0)
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return -15;
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return kLinear2LoudnessScale * log(rms);
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}
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double Db2Loudness(double db) {
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return db * kLinear2LoudnessScale / kLinear2DbScale;
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}
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double Dbfs2Loudness(double dbfs) {
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return Db2Loudness(90 + dbfs);
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}
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