OLD | NEW |
---|---|
1 /* | 1 /* |
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 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_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_ |
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_ | 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_ |
13 | 13 |
14 #include "webrtc/base/constructormagic.h" | 14 #include "webrtc/base/constructormagic.h" |
15 #include "webrtc/base/criticalsection.h" | 15 #include "webrtc/base/criticalsection.h" |
16 #include "webrtc/base/thread_checker.h" | 16 #include "webrtc/base/thread_checker.h" |
17 #include "webrtc/modules/audio_processing/agc/agc_manager_direct.h" | 17 #include "webrtc/modules/audio_processing/agc/agc_manager_direct.h" |
18 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 18 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
19 | 19 |
20 namespace webrtc { | 20 namespace webrtc { |
21 | 21 |
22 class ApmDataDumper; | |
23 | |
22 // This class has two main purposes: | 24 // This class has two main purposes: |
23 // | 25 // |
24 // 1) It is returned instead of the real GainControl after the new AGC has been | 26 // 1) It is returned instead of the real GainControl after the new AGC has been |
25 // enabled in order to prevent an outside user from overriding compression | 27 // enabled in order to prevent an outside user from overriding compression |
26 // settings. It doesn't do anything in its implementation, except for | 28 // settings. It doesn't do anything in its implementation, except for |
27 // delegating the const methods and Enable calls to the real GainControl, so | 29 // delegating the const methods and Enable calls to the real GainControl, so |
28 // AGC can still be disabled. | 30 // AGC can still be disabled. |
29 // | 31 // |
30 // 2) It is injected into AgcManagerDirect and implements volume callbacks for | 32 // 2) It is injected into AgcManagerDirect and implements volume callbacks for |
31 // getting and setting the volume level. It just caches this value to be used | 33 // getting and setting the volume level. It just caches this value to be used |
32 // in VoiceEngine later. | 34 // in VoiceEngine later. |
33 class GainControlForExperimentalAgc : public GainControl, | 35 class GainControlForExperimentalAgc : public GainControl, |
34 public VolumeCallbacks { | 36 public VolumeCallbacks { |
35 public: | 37 public: |
36 explicit GainControlForExperimentalAgc(GainControl* gain_control, | 38 GainControlForExperimentalAgc(GainControl* gain_control, |
37 rtc::CriticalSection* crit_capture); | 39 rtc::CriticalSection* crit_capture); |
40 ~GainControlForExperimentalAgc() override; | |
peah-webrtc
2016/10/28 08:37:50
This was required by the compiler as this CL adds
| |
38 | 41 |
39 // GainControl implementation. | 42 // GainControl implementation. |
40 int Enable(bool enable) override; | 43 int Enable(bool enable) override; |
41 bool is_enabled() const override; | 44 bool is_enabled() const override; |
42 int set_stream_analog_level(int level) override; | 45 int set_stream_analog_level(int level) override; |
43 int stream_analog_level() override; | 46 int stream_analog_level() override; |
44 int set_mode(Mode mode) override; | 47 int set_mode(Mode mode) override; |
45 Mode mode() const override; | 48 Mode mode() const override; |
46 int set_target_level_dbfs(int level) override; | 49 int set_target_level_dbfs(int level) override; |
47 int target_level_dbfs() const override; | 50 int target_level_dbfs() const override; |
48 int set_compression_gain_db(int gain) override; | 51 int set_compression_gain_db(int gain) override; |
49 int compression_gain_db() const override; | 52 int compression_gain_db() const override; |
50 int enable_limiter(bool enable) override; | 53 int enable_limiter(bool enable) override; |
51 bool is_limiter_enabled() const override; | 54 bool is_limiter_enabled() const override; |
52 int set_analog_level_limits(int minimum, int maximum) override; | 55 int set_analog_level_limits(int minimum, int maximum) override; |
53 int analog_level_minimum() const override; | 56 int analog_level_minimum() const override; |
54 int analog_level_maximum() const override; | 57 int analog_level_maximum() const override; |
55 bool stream_is_saturated() const override; | 58 bool stream_is_saturated() const override; |
56 | 59 |
57 // VolumeCallbacks implementation. | 60 // VolumeCallbacks implementation. |
58 void SetMicVolume(int volume) override; | 61 void SetMicVolume(int volume) override; |
59 int GetMicVolume() override; | 62 int GetMicVolume() override; |
60 | 63 |
64 void Initialize(); | |
65 | |
61 private: | 66 private: |
67 std::unique_ptr<ApmDataDumper> data_dumper_; | |
62 GainControl* real_gain_control_; | 68 GainControl* real_gain_control_; |
63 int volume_; | 69 int volume_; |
64 rtc::CriticalSection* crit_capture_; | 70 rtc::CriticalSection* crit_capture_; |
65 RTC_DISALLOW_COPY_AND_ASSIGN(GainControlForExperimentalAgc); | 71 static int instance_counter_; |
72 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlForExperimentalAgc); | |
66 }; | 73 }; |
67 | 74 |
68 } // namespace webrtc | 75 } // namespace webrtc |
69 | 76 |
70 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_ | 77 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_ |
OLD | NEW |