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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" | |
| 14 #include "base/synchronization/lock.h" | 13 #include "base/synchronization/lock.h" |
| 15 #include "base/threading/thread_checker.h" | 14 #include "base/threading/thread_checker.h" |
| 15 #include "content/renderer/media/media_stream_audio_level_calculator.h" |
| 16 #include "content/renderer/media/media_stream_audio_track.h" | 16 #include "content/renderer/media/media_stream_audio_track.h" |
| 17 #include "content/renderer/media/tagged_list.h" | 17 #include "content/renderer/media/tagged_list.h" |
| 18 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
| 18 #include "media/audio/audio_parameters.h" | 19 #include "media/audio/audio_parameters.h" |
| 19 | 20 |
| 20 namespace media { | 21 namespace media { |
| 21 class AudioBus; | 22 class AudioBus; |
| 22 } | 23 } |
| 23 | 24 |
| 24 namespace content { | 25 namespace content { |
| 25 | 26 |
| 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; | |
| 34 | 31 |
| 35 // A WebRtcLocalAudioTrack instance contains the implementations of | 32 // A WebRtcLocalAudioTrack manages thread-safe connects/disconnects to sinks, |
| 36 // MediaStreamTrackExtraData. | 33 // and the delivery of audio data from the source to the sinks. |
| 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. | |
| 41 class CONTENT_EXPORT WebRtcLocalAudioTrack | 34 class CONTENT_EXPORT WebRtcLocalAudioTrack |
| 42 : NON_EXPORTED_BASE(public MediaStreamAudioTrack) { | 35 : NON_EXPORTED_BASE(public MediaStreamAudioTrack) { |
| 43 public: | 36 public: |
| 44 WebRtcLocalAudioTrack(WebRtcLocalAudioTrackAdapter* adapter, | 37 explicit WebRtcLocalAudioTrack( |
| 45 const scoped_refptr<WebRtcAudioCapturer>& capturer, | 38 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter); |
| 46 WebAudioCapturerSource* webaudio_source); | |
| 47 | 39 |
| 48 ~WebRtcLocalAudioTrack() override; | 40 ~WebRtcLocalAudioTrack() override; |
| 49 | 41 |
| 50 // Add a sink to the track. This function will trigger a OnSetFormat() | 42 // Add a sink to the track. This function will trigger a OnSetFormat() |
| 51 // call on the |sink|. | 43 // call on the |sink|. |
| 52 // Called on the main render thread. | 44 // Called on the main render thread. |
| 53 void AddSink(MediaStreamAudioSink* sink) override; | 45 void AddSink(MediaStreamAudioSink* sink) override; |
| 54 | 46 |
| 55 // Remove a sink from the track. | 47 // Remove a sink from the track. |
| 56 // Called on the main render thread. | 48 // Called on the main render thread. |
| 57 void RemoveSink(MediaStreamAudioSink* sink) override; | 49 void RemoveSink(MediaStreamAudioSink* sink) override; |
| 58 | 50 |
| 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 | |
| 63 // Overrides for MediaStreamTrack. | 51 // Overrides for MediaStreamTrack. |
| 64 | |
| 65 void SetEnabled(bool enabled) override; | 52 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 | |
| 71 webrtc::AudioTrackInterface* GetAudioAdapter() override; | 53 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. | |
| 76 media::AudioParameters GetOutputFormat() const override; | 54 media::AudioParameters GetOutputFormat() const override; |
| 77 | 55 |
| 78 // Method called by the capturer to deliver the capture data. | 56 // Method called by the capturer to deliver the capture data. |
| 79 // Called on the capture audio thread. | 57 // Called on the capture audio thread. |
| 80 void Capture(const media::AudioBus& audio_bus, | 58 void Capture(const media::AudioBus& audio_bus, |
| 81 base::TimeTicks estimated_capture_time, | 59 base::TimeTicks estimated_capture_time); |
| 82 bool force_report_nonzero_energy); | |
| 83 | 60 |
| 84 // Method called by the capturer to set the audio parameters used by source | 61 // Method called by the capturer to set the audio parameters used by source |
| 85 // of the capture data.. | 62 // of the capture data.. |
| 86 // Called on the capture audio thread. | 63 // Called on the capture audio thread. |
| 87 void OnSetFormat(const media::AudioParameters& params); | 64 void OnSetFormat(const media::AudioParameters& params); |
| 88 | 65 |
| 89 // Method called by the capturer to set the processor that applies signal | 66 // Called by the capturer before the audio data flow begins to set the object |
| 90 // processing on the data of the track. | 67 // that provides shared access to the current audio signal level. |
| 91 // Called on the capture audio thread. | 68 void SetLevel(scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) { |
| 69 adapter_->SetLevel(level); |
| 70 } |
| 71 |
| 72 // Called by the capturer before the audio data flow begins to provide a |
| 73 // reference to the audio processor so that the track can query stats from it. |
| 92 void SetAudioProcessor( | 74 void SetAudioProcessor( |
| 93 const scoped_refptr<MediaStreamAudioProcessor>& processor); | 75 const scoped_refptr<MediaStreamAudioProcessor>& processor) { |
| 76 adapter_->SetAudioProcessor(processor); |
| 77 } |
| 94 | 78 |
| 95 private: | 79 private: |
| 96 typedef TaggedList<MediaStreamAudioTrackSink> SinkList; | 80 typedef TaggedList<MediaStreamAudioTrackSink> SinkList; |
| 97 | 81 |
| 82 // MediaStreamAudioTrack override. |
| 83 void OnStop() final; |
| 84 |
| 98 // All usage of libjingle is through this adapter. The adapter holds | 85 // All usage of libjingle is through this adapter. The adapter holds |
| 99 // a pointer to this object, but no reference. | 86 // a pointer to this object, but no reference. |
| 100 const scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_; | 87 const scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_; |
| 101 | 88 |
| 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 | |
| 109 // A tagged list of sinks that the audio data is fed to. Tags | 89 // A tagged list of sinks that the audio data is fed to. Tags |
| 110 // indicate tracks that need to be notified that the audio format | 90 // indicate tracks that need to be notified that the audio format |
| 111 // has changed. | 91 // has changed. |
| 112 SinkList sinks_; | 92 SinkList sinks_; |
| 113 | 93 |
| 114 // Tests that methods are called on libjingle's signaling thread. | 94 // Tests that methods are called on libjingle's signaling thread. |
| 115 base::ThreadChecker signal_thread_checker_; | 95 base::ThreadChecker signal_thread_checker_; |
| 116 | 96 |
| 117 // Used to DCHECK that some methods are called on the capture audio thread. | 97 // Used to DCHECK that some methods are called on the capture audio thread. |
| 118 base::ThreadChecker capture_thread_checker_; | 98 base::ThreadChecker capture_thread_checker_; |
| 119 | 99 |
| 120 // Protects |params_| and |sinks_|. | 100 // Protects |params_| and |sinks_|. |
| 121 mutable base::Lock lock_; | 101 mutable base::Lock lock_; |
| 122 | 102 |
| 123 // Audio parameters of the audio capture stream. | 103 // Audio parameters of the audio capture stream. |
| 124 // Accessed on only the audio capture thread. | |
| 125 media::AudioParameters audio_parameters_; | 104 media::AudioParameters audio_parameters_; |
| 126 | 105 |
| 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 | |
| 131 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); | 106 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); |
| 132 }; | 107 }; |
| 133 | 108 |
| 134 } // namespace content | 109 } // namespace content |
| 135 | 110 |
| 136 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ | 111 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ |
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