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