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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
1.4 KiB
C
40 lines
1.4 KiB
C
/*
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* Copyright (c) 2011 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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// Gaussian probability calculations internally used in vad_core.c.
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#ifndef COMMON_AUDIO_VAD_VAD_GMM_H_
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#define COMMON_AUDIO_VAD_VAD_GMM_H_
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#include <stdint.h>
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// Calculates the probability for |input|, given that |input| comes from a
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// normal distribution with mean and standard deviation (|mean|, |std|).
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//
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// Inputs:
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// - input : input sample in Q4.
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// - mean : mean input in the statistical model, Q7.
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// - std : standard deviation, Q7.
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//
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// Output:
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//
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// - delta : input used when updating the model, Q11.
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// |delta| = (|input| - |mean|) / |std|^2.
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//
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// Return:
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// (probability for |input|) =
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// 1 / |std| * exp(-(|input| - |mean|)^2 / (2 * |std|^2));
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int32_t WebRtcVad_GaussianProbability(int16_t input,
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int16_t mean,
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int16_t std,
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int16_t* delta);
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#endif // COMMON_AUDIO_VAD_VAD_GMM_H_
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