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Issue 2051443002: Change name of files and class in agc/histogram* in order to avoid file-name clash. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Changed name of the Histogram class to reflect what it actually does, and to match the new file name Created 4 years, 6 months ago
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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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