Arun Raghavan 753eada3aa Update audio_processing module
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1

Update notes:

 * Pull in third party license file

 * Replace .gypi files with BUILD.gn to keep track of what changes
   upstream

 * Bunch of new filse pulled in as dependencies

 * Won't build yet due to changes needed on top of these
2015-10-15 16:18:45 +05:30

58 lines
1.9 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_PITCH_BASED_VAD_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_PITCH_BASED_VAD_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_processing/vad/common.h"
#include "webrtc/modules/audio_processing/vad/gmm.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioFrame;
class VadCircularBuffer;
// Computes the probability of the input audio frame to be active given
// the corresponding pitch-gain and lag of the frame.
class PitchBasedVad {
public:
PitchBasedVad();
~PitchBasedVad();
// Compute pitch-based voicing probability, given the features.
// features: a structure containing features required for computing voicing
// probabilities.
//
// p_combined: an array which contains the combined activity probabilities
// computed prior to the call of this function. The method,
// then, computes the voicing probabilities and combine them
// with the given values. The result are returned in |p|.
int VoicingProbability(const AudioFeatures& features, double* p_combined);
private:
int UpdatePrior(double p);
// TODO(turajs): maybe defining this at a higher level (maybe enum) so that
// all the code recognize it as "no-error."
static const int kNoError = 0;
GmmParameters noise_gmm_;
GmmParameters voice_gmm_;
double p_prior_;
rtc::scoped_ptr<VadCircularBuffer> circular_buffer_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_PITCH_BASED_VAD_H_