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1 /*
2  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_
13 
14 #include "webrtc/base/scoped_ptr.h"
15 #include "webrtc/modules/audio_processing/vad/voice_activity_detector.h"
16 #include "webrtc/typedefs.h"
17 
18 namespace webrtc {
19 
20 class AudioFrame;
21 class Histogram;
22 
23 class Agc {
24  public:
25   Agc();
26   virtual ~Agc();
27 
28   // Returns the proportion of samples in the buffer which are at full-scale
29   // (and presumably clipped).
30   virtual float AnalyzePreproc(const int16_t* audio, size_t length);
31   // |audio| must be mono; in a multi-channel stream, provide the first (usually
32   // left) channel.
33   virtual int Process(const int16_t* audio, size_t length, int sample_rate_hz);
34 
35   // Retrieves the difference between the target RMS level and the current
36   // signal RMS level in dB. Returns true if an update is available and false
37   // otherwise, in which case |error| should be ignored and no action taken.
38   virtual bool GetRmsErrorDb(int* error);
39   virtual void Reset();
40 
41   virtual int set_target_level_dbfs(int level);
target_level_dbfs()42   virtual int target_level_dbfs() const { return target_level_dbfs_; }
43 
voice_probability()44   virtual float voice_probability() const {
45     return vad_.last_voice_probability();
46   }
47 
48  private:
49   double target_level_loudness_;
50   int target_level_dbfs_;
51   rtc::scoped_ptr<Histogram> histogram_;
52   rtc::scoped_ptr<Histogram> inactive_histogram_;
53   VoiceActivityDetector vad_;
54 };
55 
56 }  // namespace webrtc
57 
58 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_H_
59