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 3 // found in the LICENSE file. 4 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ 7 8 #include <list> 9 #include <string> 10 11 #include "base/memory/ref_counted.h" 12 #include "base/synchronization/lock.h" 13 #include "base/threading/thread_checker.h" 14 #include "content/renderer/media/media_stream_track.h" 15 #include "content/renderer/media/tagged_list.h" 16 #include "content/renderer/media/webrtc_audio_device_impl.h" 17 18 namespace content { 19 20 class MediaStreamAudioLevelCalculator; 21 class MediaStreamAudioProcessor; 22 class MediaStreamAudioSink; 23 class MediaStreamAudioSinkOwner; 24 class MediaStreamAudioTrackSink; 25 class PeerConnectionAudioSink; 26 class WebAudioCapturerSource; 27 class WebRtcAudioCapturer; 28 class WebRtcLocalAudioTrackAdapter; 29 30 // A WebRtcLocalAudioTrack instance contains the implementations of 31 // MediaStreamTrackExtraData. 32 // When an instance is created, it will register itself as a track to the 33 // WebRtcAudioCapturer to get the captured data, and forward the data to 34 // its |sinks_|. The data flow can be stopped by disabling the audio track. 35 class CONTENT_EXPORT WebRtcLocalAudioTrack NON_EXPORTED_BASE(public MediaStreamTrack)36 : NON_EXPORTED_BASE(public MediaStreamTrack) { 37 public: 38 WebRtcLocalAudioTrack(WebRtcLocalAudioTrackAdapter* adapter, 39 const scoped_refptr<WebRtcAudioCapturer>& capturer, 40 WebAudioCapturerSource* webaudio_source); 41 42 virtual ~WebRtcLocalAudioTrack(); 43 44 // Add a sink to the track. This function will trigger a OnSetFormat() 45 // call on the |sink|. 46 // Called on the main render thread. 47 void AddSink(MediaStreamAudioSink* sink); 48 49 // Remove a sink from the track. 50 // Called on the main render thread. 51 void RemoveSink(MediaStreamAudioSink* sink); 52 53 // Add/remove PeerConnection sink to/from the track. 54 // TODO(xians): Remove these two methods after PeerConnection can use the 55 // same sink interface as MediaStreamAudioSink. 56 void AddSink(PeerConnectionAudioSink* sink); 57 void RemoveSink(PeerConnectionAudioSink* sink); 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 63 // Stops the local audio track. Called on the main render thread and 64 // should be called only once when audio track going away. 65 virtual void Stop() OVERRIDE; 66 67 // Method called by the capturer to deliver the capture data. 68 // Called on the capture audio thread. 69 void Capture(const int16* audio_data, 70 base::TimeDelta delay, 71 int volume, 72 bool key_pressed, 73 bool need_audio_processing); 74 75 // Method called by the capturer to set the audio parameters used by source 76 // of the capture data.. 77 // Called on the capture audio thread. 78 void OnSetFormat(const media::AudioParameters& params); 79 80 // Method called by the capturer to set the processor that applies signal 81 // processing on the data of the track. 82 // Called on the capture audio thread. 83 void SetAudioProcessor( 84 const scoped_refptr<MediaStreamAudioProcessor>& processor); 85 86 private: 87 typedef TaggedList<MediaStreamAudioTrackSink> SinkList; 88 89 // All usage of libjingle is through this adapter. The adapter holds 90 // a reference on this object, but not vice versa. 91 WebRtcLocalAudioTrackAdapter* adapter_; 92 93 // The provider of captured data to render. 94 scoped_refptr<WebRtcAudioCapturer> capturer_; 95 96 // The source of the audio track which is used by WebAudio, which provides 97 // data to the audio track when hooking up with WebAudio. 98 scoped_refptr<WebAudioCapturerSource> webaudio_source_; 99 100 // A tagged list of sinks that the audio data is fed to. Tags 101 // indicate tracks that need to be notified that the audio format 102 // has changed. 103 SinkList sinks_; 104 105 // Used to DCHECK that some methods are called on the main render thread. 106 base::ThreadChecker main_render_thread_checker_; 107 108 // Used to DCHECK that some methods are called on the capture audio thread. 109 base::ThreadChecker capture_thread_checker_; 110 111 // Protects |params_| and |sinks_|. 112 mutable base::Lock lock_; 113 114 // Audio parameters of the audio capture stream. 115 // Accessed on only the audio capture thread. 116 media::AudioParameters audio_parameters_; 117 118 // Used to calculate the signal level that shows in the UI. 119 // Accessed on only the audio thread. 120 scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_; 121 122 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); 123 }; 124 125 } // namespace content 126 127 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ 128