Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(265)

Side by Side Diff: webrtc/modules/audio_processing/gain_control_impl.h

Issue 1801003002: Removed the dependency on AudioProcessingImpl in GainControlImpl (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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_GAIN_CONTROL_IMPL_H_ 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
13 13
14 #include <memory> 14 #include <memory>
15 #include <vector> 15 #include <vector>
16 16
17 #include "webrtc/base/constructormagic.h" 17 #include "webrtc/base/constructormagic.h"
18 #include "webrtc/base/criticalsection.h" 18 #include "webrtc/base/criticalsection.h"
19 #include "webrtc/base/thread_annotations.h" 19 #include "webrtc/base/thread_annotations.h"
20 #include "webrtc/common_audio/swap_queue.h" 20 #include "webrtc/common_audio/swap_queue.h"
21 #include "webrtc/modules/audio_processing/include/audio_processing.h" 21 #include "webrtc/modules/audio_processing/include/audio_processing.h"
22 #include "webrtc/modules/audio_processing/render_queue_item_verifier.h" 22 #include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
23 23
24 namespace webrtc { 24 namespace webrtc {
25 25
26 class AudioBuffer; 26 class AudioBuffer;
27 27
28 class GainControlImpl : public GainControl { 28 class GainControlImpl : public GainControl {
29 public: 29 public:
30 GainControlImpl(const AudioProcessing* apm, 30 GainControlImpl(rtc::CriticalSection* crit_render,
31 rtc::CriticalSection* crit_render,
32 rtc::CriticalSection* crit_capture); 31 rtc::CriticalSection* crit_capture);
33 ~GainControlImpl() override; 32 ~GainControlImpl() override;
34 33
35 int ProcessRenderAudio(AudioBuffer* audio); 34 int ProcessRenderAudio(AudioBuffer* audio);
36 int AnalyzeCaptureAudio(AudioBuffer* audio); 35 int AnalyzeCaptureAudio(AudioBuffer* audio);
37 int ProcessCaptureAudio(AudioBuffer* audio); 36 int ProcessCaptureAudio(AudioBuffer* audio, bool stream_has_echo);
38 37
39 void Initialize(); 38 void Initialize(size_t num_proc_channels, int sample_rate_hz);
40 39
41 // GainControl implementation. 40 // GainControl implementation.
42 bool is_enabled() const override; 41 bool is_enabled() const override;
43 int stream_analog_level() override; 42 int stream_analog_level() override;
44 bool is_limiter_enabled() const override; 43 bool is_limiter_enabled() const override;
45 Mode mode() const override; 44 Mode mode() const override;
46 45
47 // Reads render side data that has been queued on the render call. 46 // Reads render side data that has been queued on the render call.
48 void ReadQueuedRenderData(); 47 void ReadQueuedRenderData();
49 48
50 private: 49 private:
51 class GainController; 50 class GainController;
52 51
53 // GainControl implementation. 52 // GainControl implementation.
54 int Enable(bool enable) override; 53 int Enable(bool enable) override;
55 int set_stream_analog_level(int level) override; 54 int set_stream_analog_level(int level) override;
56 int set_mode(Mode mode) override; 55 int set_mode(Mode mode) override;
57 int set_target_level_dbfs(int level) override; 56 int set_target_level_dbfs(int level) override;
58 int target_level_dbfs() const override; 57 int target_level_dbfs() const override;
59 int set_compression_gain_db(int gain) override; 58 int set_compression_gain_db(int gain) override;
60 int compression_gain_db() const override; 59 int compression_gain_db() const override;
61 int enable_limiter(bool enable) override; 60 int enable_limiter(bool enable) override;
62 int set_analog_level_limits(int minimum, int maximum) override; 61 int set_analog_level_limits(int minimum, int maximum) override;
63 int analog_level_minimum() const override; 62 int analog_level_minimum() const override;
64 int analog_level_maximum() const override; 63 int analog_level_maximum() const override;
65 bool stream_is_saturated() const override; 64 bool stream_is_saturated() const override;
66 65
67 size_t num_handles_required() const;
68
69 void AllocateRenderQueue(); 66 void AllocateRenderQueue();
70 int Configure(); 67 int Configure();
71 68
72 // Not guarded as its public API is thread safe.
73 const AudioProcessing* apm_;
74
75 rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_); 69 rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
76 rtc::CriticalSection* const crit_capture_; 70 rtc::CriticalSection* const crit_capture_;
77 71
78 bool enabled_ = false; 72 bool enabled_ = false;
79 73
80 Mode mode_ GUARDED_BY(crit_capture_); 74 Mode mode_ GUARDED_BY(crit_capture_);
81 int minimum_capture_level_ GUARDED_BY(crit_capture_); 75 int minimum_capture_level_ GUARDED_BY(crit_capture_);
82 int maximum_capture_level_ GUARDED_BY(crit_capture_); 76 int maximum_capture_level_ GUARDED_BY(crit_capture_);
83 bool limiter_enabled_ GUARDED_BY(crit_capture_); 77 bool limiter_enabled_ GUARDED_BY(crit_capture_);
84 int target_level_dbfs_ GUARDED_BY(crit_capture_); 78 int target_level_dbfs_ GUARDED_BY(crit_capture_);
85 int compression_gain_db_ GUARDED_BY(crit_capture_); 79 int compression_gain_db_ GUARDED_BY(crit_capture_);
86 int analog_capture_level_ GUARDED_BY(crit_capture_); 80 int analog_capture_level_ GUARDED_BY(crit_capture_);
87 bool was_analog_level_set_ GUARDED_BY(crit_capture_); 81 bool was_analog_level_set_ GUARDED_BY(crit_capture_);
88 bool stream_is_saturated_ GUARDED_BY(crit_capture_); 82 bool stream_is_saturated_ GUARDED_BY(crit_capture_);
89 83
90 size_t render_queue_element_max_size_ GUARDED_BY(crit_render_) 84 size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
91 GUARDED_BY(crit_capture_); 85 GUARDED_BY(crit_capture_);
92 std::vector<int16_t> render_queue_buffer_ GUARDED_BY(crit_render_); 86 std::vector<int16_t> render_queue_buffer_ GUARDED_BY(crit_render_);
93 std::vector<int16_t> capture_queue_buffer_ GUARDED_BY(crit_capture_); 87 std::vector<int16_t> capture_queue_buffer_ GUARDED_BY(crit_capture_);
94 88
95 // Lock protection not needed. 89 // Lock protection not needed.
96 std::unique_ptr< 90 std::unique_ptr<
97 SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>> 91 SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
98 render_signal_queue_; 92 render_signal_queue_;
99 93
100 std::vector<std::unique_ptr<GainController>> gain_controllers_; 94 std::vector<std::unique_ptr<GainController>> gain_controllers_;
101 95
96 rtc::Optional<size_t> num_proc_channels_ GUARDED_BY(crit_capture_);
97 rtc::Optional<int> sample_rate_hz_ GUARDED_BY(crit_capture_);
98
102 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlImpl); 99 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlImpl);
103 }; 100 };
104 } // namespace webrtc 101 } // namespace webrtc
105 102
106 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ 103 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
OLDNEW
« no previous file with comments | « webrtc/modules/audio_processing/audio_processing_impl.cc ('k') | webrtc/modules/audio_processing/gain_control_impl.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698