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/*
* Copyright (c) 2014 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_CODING_NETEQ_AUDIO_CLASSIFIER_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_CLASSIFIER_H_
extern "C" {
#include "celt.h"
#include "analysis.h"
#include "opus_private.h"
}
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// This class provides a speech/music classification and is a wrapper over the
// Opus classifier. It currently only supports 48 kHz mono or stereo with a
// frame size of 20 ms.
class AudioClassifier {
public:
AudioClassifier();
virtual ~AudioClassifier();
// Classifies one frame of audio data in input,
// input_length : must be channels * 960;
// channels : must be 1 (mono) or 2 (stereo).
bool Analysis(const int16_t* input, int input_length, int channels);
// Gets the current classification : true = music, false = speech.
virtual bool is_music() const;
// Gets the current music probability.
float music_probability() const { return music_probability_; }
private:
AnalysisInfo analysis_info_;
bool is_music_;
float music_probability_;
const CELTMode* celt_mode_;
TonalityAnalysisState analysis_state_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_CLASSIFIER_H_