788acd17ad
R=aluebs@webrtc.org, bjornv@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/32769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d
70 lines
2.2 KiB
C++
70 lines
2.2 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_AGC_AGC_H_
|
|
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_
|
|
|
|
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
|
|
#include "webrtc/typedefs.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class AudioFrame;
|
|
class AgcAudioProc;
|
|
class Histogram;
|
|
class PitchBasedVad;
|
|
class Resampler;
|
|
class StandaloneVad;
|
|
|
|
class Agc {
|
|
public:
|
|
Agc();
|
|
virtual ~Agc();
|
|
|
|
// Returns the proportion of samples in the buffer which are at full-scale
|
|
// (and presumably clipped).
|
|
virtual float AnalyzePreproc(const int16_t* audio, int length);
|
|
// |audio| must be mono; in a multi-channel stream, provide the first (usually
|
|
// left) channel.
|
|
virtual int Process(const int16_t* audio, int length, int sample_rate_hz);
|
|
|
|
// Retrieves the difference between the target RMS level and the current
|
|
// signal RMS level in dB. Returns true if an update is available and false
|
|
// otherwise, in which case |error| should be ignored and no action taken.
|
|
virtual bool GetRmsErrorDb(int* error);
|
|
virtual void Reset();
|
|
|
|
virtual int set_target_level_dbfs(int level);
|
|
virtual int target_level_dbfs() const { return target_level_dbfs_; }
|
|
|
|
virtual void EnableStandaloneVad(bool enable);
|
|
virtual bool standalone_vad_enabled() const {
|
|
return standalone_vad_enabled_;
|
|
}
|
|
|
|
virtual double voice_probability() const { return last_voice_probability_; }
|
|
|
|
private:
|
|
double target_level_loudness_;
|
|
double last_voice_probability_;
|
|
int target_level_dbfs_;
|
|
bool standalone_vad_enabled_;
|
|
scoped_ptr<Histogram> histogram_;
|
|
scoped_ptr<Histogram> inactive_histogram_;
|
|
scoped_ptr<AgcAudioProc> audio_processing_;
|
|
scoped_ptr<PitchBasedVad> pitch_based_vad_;
|
|
scoped_ptr<StandaloneVad> standalone_vad_;
|
|
scoped_ptr<Resampler> resampler_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_
|