OLD | NEW |
1 // Copyright 2014 The Chromium Authors. All rights reserved. | 1 // Copyright 2014 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ | 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ |
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ | 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ |
7 | 7 |
8 #include <vector> | 8 #include <vector> |
9 | 9 |
10 #include "base/memory/ref_counted.h" | 10 #include "base/memory/ref_counted.h" |
11 #include "base/memory/scoped_vector.h" | 11 #include "base/memory/scoped_vector.h" |
12 #include "base/single_thread_task_runner.h" | 12 #include "base/single_thread_task_runner.h" |
13 #include "base/synchronization/lock.h" | 13 #include "base/synchronization/lock.h" |
14 #include "base/threading/thread_checker.h" | |
15 #include "content/common/content_export.h" | 14 #include "content/common/content_export.h" |
| 15 #include "content/renderer/media/media_stream_audio_level_calculator.h" |
16 #include "third_party/webrtc/api/mediastreamtrack.h" | 16 #include "third_party/webrtc/api/mediastreamtrack.h" |
17 #include "third_party/webrtc/media/base/audiorenderer.h" | 17 #include "third_party/webrtc/media/base/audiorenderer.h" |
18 | 18 |
19 namespace cricket { | 19 namespace cricket { |
20 class AudioRenderer; | 20 class AudioRenderer; |
21 } | 21 } |
22 | 22 |
23 namespace webrtc { | 23 namespace webrtc { |
24 class AudioSourceInterface; | 24 class AudioSourceInterface; |
25 class AudioProcessorInterface; | 25 class AudioProcessorInterface; |
26 } | 26 } |
27 | 27 |
28 namespace content { | 28 namespace content { |
29 | 29 |
30 class MediaStreamAudioProcessor; | 30 class MediaStreamAudioProcessor; |
31 class WebRtcAudioSinkAdapter; | 31 class WebRtcAudioSinkAdapter; |
32 class WebRtcLocalAudioTrack; | 32 class WebRtcLocalAudioTrack; |
33 | 33 |
| 34 // Provides an implementation of the webrtc::AudioTrackInterface that can be |
| 35 // bound/unbound to/from a MediaStreamAudioTrack. In other words, this is an |
| 36 // adapter that sits between the media stream object graph and WebRtc's object |
| 37 // graph and proxies between the two. |
34 class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter | 38 class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter |
35 : NON_EXPORTED_BASE( | 39 : NON_EXPORTED_BASE( |
36 public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) { | 40 public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) { |
37 public: | 41 public: |
38 static scoped_refptr<WebRtcLocalAudioTrackAdapter> Create( | 42 static scoped_refptr<WebRtcLocalAudioTrackAdapter> Create( |
39 const std::string& label, | 43 const std::string& label, |
40 webrtc::AudioSourceInterface* track_source); | 44 webrtc::AudioSourceInterface* track_source); |
41 | 45 |
42 WebRtcLocalAudioTrackAdapter( | 46 WebRtcLocalAudioTrackAdapter( |
43 const std::string& label, | 47 const std::string& label, |
44 webrtc::AudioSourceInterface* track_source, | 48 webrtc::AudioSourceInterface* track_source, |
45 const scoped_refptr<base::SingleThreadTaskRunner>& signaling_thread); | 49 const scoped_refptr<base::SingleThreadTaskRunner>& signaling_task_runner); |
46 | 50 |
47 ~WebRtcLocalAudioTrackAdapter() override; | 51 ~WebRtcLocalAudioTrackAdapter() override; |
48 | 52 |
49 void Initialize(WebRtcLocalAudioTrack* owner); | 53 void Initialize(WebRtcLocalAudioTrack* owner); |
50 | 54 |
51 // Called on the audio thread by the WebRtcLocalAudioTrack to set the signal | 55 // Set the object that provides shared access to the current audio signal |
52 // level of the audio data. | 56 // level. This method may only be called once, before the audio data flow |
53 void SetSignalLevel(int signal_level); | 57 // starts. |
| 58 void SetLevel(scoped_refptr<MediaStreamAudioLevelCalculator::Level> level); |
54 | 59 |
55 // Method called by the WebRtcLocalAudioTrack to set the processor that | 60 // Method called by the WebRtcLocalAudioTrack to set the processor that |
56 // applies signal processing on the data of the track. | 61 // applies signal processing on the data of the track. |
57 // This class will keep a reference of the |processor|. | 62 // This class will keep a reference of the |processor|. |
58 // Called on the main render thread. | 63 // Called on the main render thread. |
| 64 // This method may only be called once, before the audio data flow starts. |
59 void SetAudioProcessor( | 65 void SetAudioProcessor( |
60 const scoped_refptr<MediaStreamAudioProcessor>& processor); | 66 const scoped_refptr<MediaStreamAudioProcessor>& processor); |
61 | 67 |
62 // webrtc::MediaStreamTrack implementation. | 68 // webrtc::MediaStreamTrack implementation. |
63 std::string kind() const override; | 69 std::string kind() const override; |
64 bool set_enabled(bool enable) override; | 70 bool set_enabled(bool enable) override; |
65 | 71 |
66 private: | 72 private: |
67 // webrtc::AudioTrackInterface implementation. | 73 // webrtc::AudioTrackInterface implementation. |
68 void AddSink(webrtc::AudioTrackSinkInterface* sink) override; | 74 void AddSink(webrtc::AudioTrackSinkInterface* sink) override; |
69 void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override; | 75 void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override; |
70 bool GetSignalLevel(int* level) override; | 76 bool GetSignalLevel(int* level) override; |
71 rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor() | 77 rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor() |
72 override; | 78 override; |
73 webrtc::AudioSourceInterface* GetSource() const override; | 79 webrtc::AudioSourceInterface* GetSource() const override; |
74 | 80 |
75 // Weak reference. | 81 // Weak reference. |
76 WebRtcLocalAudioTrack* owner_; | 82 WebRtcLocalAudioTrack* owner_; |
77 | 83 |
78 // The source of the audio track which handles the audio constraints. | 84 // The source of the audio track which handles the audio constraints. |
79 // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack. | 85 // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack. |
80 rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_; | 86 rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_; |
81 | 87 |
82 // Libjingle's signaling thread. | 88 // Libjingle's signaling thread. |
83 const scoped_refptr<base::SingleThreadTaskRunner> signaling_thread_; | 89 const scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner_; |
84 | 90 |
85 // The audio processsor that applies audio processing on the data of audio | 91 // The audio processsor that applies audio processing on the data of audio |
86 // track. | 92 // track. |
87 scoped_refptr<MediaStreamAudioProcessor> audio_processor_; | 93 scoped_refptr<MediaStreamAudioProcessor> audio_processor_; |
88 | 94 |
89 // A vector of WebRtc VoE channels that the capturer sends data to. | |
90 std::vector<int> voe_channels_; | |
91 | |
92 // A vector of the peer connection sink adapters which receive the audio data | 95 // A vector of the peer connection sink adapters which receive the audio data |
93 // from the audio track. | 96 // from the audio track. |
94 ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_; | 97 ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_; |
95 | 98 |
96 // The amplitude of the signal. | 99 // Thread-safe accessor to current audio signal level. |
97 int signal_level_; | 100 scoped_refptr<MediaStreamAudioLevelCalculator::Level> level_; |
98 | |
99 // Thread checker for libjingle's signaling thread. | |
100 base::ThreadChecker signaling_thread_checker_; | |
101 base::ThreadChecker capture_thread_; | |
102 | |
103 // Protects |voe_channels_|, |audio_processor_| and |signal_level_|. | |
104 mutable base::Lock lock_; | |
105 }; | 101 }; |
106 | 102 |
107 } // namespace content | 103 } // namespace content |
108 | 104 |
109 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ | 105 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ |
OLD | NEW |