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1 // Copyright 2013 The Chromium Authors. All rights reserved. | 1 // Copyright 2013 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_LOCAL_AUDIO_TRACK_H_ | 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ |
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ | 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ |
7 | 7 |
8 #include <list> | 8 #include <list> |
9 #include <string> | 9 #include <string> |
10 | 10 |
11 #include "base/macros.h" | 11 #include "base/macros.h" |
12 #include "base/memory/ref_counted.h" | 12 #include "base/memory/ref_counted.h" |
| 13 #include "base/memory/scoped_ptr.h" |
13 #include "base/synchronization/lock.h" | 14 #include "base/synchronization/lock.h" |
14 #include "base/threading/thread_checker.h" | 15 #include "base/threading/thread_checker.h" |
15 #include "content/renderer/media/media_stream_audio_track.h" | 16 #include "content/renderer/media/media_stream_audio_track.h" |
16 #include "content/renderer/media/tagged_list.h" | 17 #include "content/renderer/media/tagged_list.h" |
17 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" | |
18 #include "media/audio/audio_parameters.h" | 18 #include "media/audio/audio_parameters.h" |
19 | 19 |
20 namespace media { | 20 namespace media { |
21 class AudioBus; | 21 class AudioBus; |
22 } | 22 } |
23 | 23 |
24 namespace content { | 24 namespace content { |
25 | 25 |
26 class MediaStreamAudioLevelCalculator; | 26 class MediaStreamAudioLevelCalculator; |
27 class MediaStreamAudioProcessor; | 27 class MediaStreamAudioProcessor; |
28 class MediaStreamAudioSink; | 28 class MediaStreamAudioSink; |
29 class MediaStreamAudioSinkOwner; | 29 class MediaStreamAudioSinkOwner; |
30 class MediaStreamAudioTrackSink; | 30 class MediaStreamAudioTrackSink; |
| 31 class WebAudioCapturerSource; |
| 32 class WebRtcAudioCapturer; |
| 33 class WebRtcLocalAudioTrackAdapter; |
31 | 34 |
32 // A WebRtcLocalAudioTrack manages thread-safe connects/disconnects to sinks, | 35 // A WebRtcLocalAudioTrack instance contains the implementations of |
33 // and the delivery of audio data from the source to the sinks. | 36 // MediaStreamTrackExtraData. |
| 37 // When an instance is created, it will register itself as a track to the |
| 38 // WebRtcAudioCapturer to get the captured data, and forward the data to |
| 39 // its |sinks_|. The data flow can be stopped by disabling the audio track. |
| 40 // TODO(tommi): Rename to MediaStreamLocalAudioTrack. |
34 class CONTENT_EXPORT WebRtcLocalAudioTrack | 41 class CONTENT_EXPORT WebRtcLocalAudioTrack |
35 : NON_EXPORTED_BASE(public MediaStreamAudioTrack) { | 42 : NON_EXPORTED_BASE(public MediaStreamAudioTrack) { |
36 public: | 43 public: |
37 explicit WebRtcLocalAudioTrack( | 44 WebRtcLocalAudioTrack(WebRtcLocalAudioTrackAdapter* adapter, |
38 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter); | 45 const scoped_refptr<WebRtcAudioCapturer>& capturer, |
| 46 WebAudioCapturerSource* webaudio_source); |
39 | 47 |
40 ~WebRtcLocalAudioTrack() override; | 48 ~WebRtcLocalAudioTrack() override; |
41 | 49 |
42 // Add a sink to the track. This function will trigger a OnSetFormat() | 50 // Add a sink to the track. This function will trigger a OnSetFormat() |
43 // call on the |sink|. | 51 // call on the |sink|. |
44 // Called on the main render thread. | 52 // Called on the main render thread. |
45 void AddSink(MediaStreamAudioSink* sink) override; | 53 void AddSink(MediaStreamAudioSink* sink) override; |
46 | 54 |
47 // Remove a sink from the track. | 55 // Remove a sink from the track. |
48 // Called on the main render thread. | 56 // Called on the main render thread. |
49 void RemoveSink(MediaStreamAudioSink* sink) override; | 57 void RemoveSink(MediaStreamAudioSink* sink) override; |
50 | 58 |
| 59 // Starts the local audio track. Called on the main render thread and |
| 60 // should be called only once when audio track is created. |
| 61 void Start(); |
| 62 |
51 // Overrides for MediaStreamTrack. | 63 // Overrides for MediaStreamTrack. |
| 64 |
52 void SetEnabled(bool enabled) override; | 65 void SetEnabled(bool enabled) override; |
| 66 |
| 67 // Stops the local audio track. Called on the main render thread and |
| 68 // should be called only once when audio track going away. |
| 69 void Stop() override; |
| 70 |
53 webrtc::AudioTrackInterface* GetAudioAdapter() override; | 71 webrtc::AudioTrackInterface* GetAudioAdapter() override; |
| 72 |
| 73 // Returns the output format of the capture source. May return an invalid |
| 74 // AudioParameters if the format is not yet available. |
| 75 // Called on the main render thread. |
54 media::AudioParameters GetOutputFormat() const override; | 76 media::AudioParameters GetOutputFormat() const override; |
55 | 77 |
56 // Method called by the capturer to deliver the capture data. | 78 // Method called by the capturer to deliver the capture data. |
57 // Called on the capture audio thread. | 79 // Called on the capture audio thread. |
58 void Capture(const media::AudioBus& audio_bus, | 80 void Capture(const media::AudioBus& audio_bus, |
59 base::TimeTicks estimated_capture_time); | 81 base::TimeTicks estimated_capture_time, |
| 82 bool force_report_nonzero_energy); |
60 | 83 |
61 // Method called by the capturer to set the audio parameters used by source | 84 // Method called by the capturer to set the audio parameters used by source |
62 // of the capture data.. | 85 // of the capture data.. |
63 // Called on the capture audio thread. | 86 // Called on the capture audio thread. |
64 void OnSetFormat(const media::AudioParameters& params); | 87 void OnSetFormat(const media::AudioParameters& params); |
65 | 88 |
66 // Called by the capturer before the audio data flow begins to set the object | 89 // Method called by the capturer to set the processor that applies signal |
67 // that provides shared access to the current audio signal level. | 90 // processing on the data of the track. |
68 void SetLevel(scoped_refptr<MediaStreamAudioLevelCalculator::Level> level); | 91 // Called on the capture audio thread. |
69 | 92 void SetAudioProcessor( |
70 // Called by the capturer before the audio data flow begins to provide a | 93 const scoped_refptr<MediaStreamAudioProcessor>& processor); |
71 // reference to the audio processor so that the track can query stats from it. | |
72 void SetAudioProcessor(scoped_refptr<MediaStreamAudioProcessor> processor); | |
73 | 94 |
74 private: | 95 private: |
75 typedef TaggedList<MediaStreamAudioTrackSink> SinkList; | 96 typedef TaggedList<MediaStreamAudioTrackSink> SinkList; |
76 | 97 |
77 // MediaStreamAudioTrack override. | |
78 void OnStop() final; | |
79 | |
80 // All usage of libjingle is through this adapter. The adapter holds | 98 // All usage of libjingle is through this adapter. The adapter holds |
81 // a pointer to this object, but no reference. | 99 // a pointer to this object, but no reference. |
82 const scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_; | 100 const scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_; |
83 | 101 |
| 102 // The provider of captured data to render. |
| 103 scoped_refptr<WebRtcAudioCapturer> capturer_; |
| 104 |
| 105 // The source of the audio track which is used by WebAudio, which provides |
| 106 // data to the audio track when hooking up with WebAudio. |
| 107 scoped_refptr<WebAudioCapturerSource> webaudio_source_; |
| 108 |
84 // A tagged list of sinks that the audio data is fed to. Tags | 109 // A tagged list of sinks that the audio data is fed to. Tags |
85 // indicate tracks that need to be notified that the audio format | 110 // indicate tracks that need to be notified that the audio format |
86 // has changed. | 111 // has changed. |
87 SinkList sinks_; | 112 SinkList sinks_; |
88 | 113 |
89 // Tests that methods are called on libjingle's signaling thread. | 114 // Tests that methods are called on libjingle's signaling thread. |
90 base::ThreadChecker signal_thread_checker_; | 115 base::ThreadChecker signal_thread_checker_; |
91 | 116 |
92 // Used to DCHECK that some methods are called on the capture audio thread. | 117 // Used to DCHECK that some methods are called on the capture audio thread. |
93 base::ThreadChecker capture_thread_checker_; | 118 base::ThreadChecker capture_thread_checker_; |
94 | 119 |
95 // Protects |params_| and |sinks_|. | 120 // Protects |params_| and |sinks_|. |
96 mutable base::Lock lock_; | 121 mutable base::Lock lock_; |
97 | 122 |
98 // Audio parameters of the audio capture stream. | 123 // Audio parameters of the audio capture stream. |
| 124 // Accessed on only the audio capture thread. |
99 media::AudioParameters audio_parameters_; | 125 media::AudioParameters audio_parameters_; |
100 | 126 |
| 127 // Used to calculate the signal level that shows in the UI. |
| 128 // Accessed on only the audio thread. |
| 129 scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_; |
| 130 |
101 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); | 131 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); |
102 }; | 132 }; |
103 | 133 |
104 } // namespace content | 134 } // namespace content |
105 | 135 |
106 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ | 136 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ |
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