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 bool force_report_nonzero_energy); 75 76 // Method called by the capturer to set the audio parameters used by source 77 // of the capture data.. 78 // Called on the capture audio thread. 79 void OnSetFormat(const media::AudioParameters& params); 80 81 // Method called by the capturer to set the processor that applies signal 82 // processing on the data of the track. 83 // Called on the capture audio thread. 84 void SetAudioProcessor( 85 const scoped_refptr<MediaStreamAudioProcessor>& processor); 86 87 private: 88 typedef TaggedList<MediaStreamAudioTrackSink> SinkList; 89 90 // All usage of libjingle is through this adapter. The adapter holds 91 // a reference on this object, but not vice versa. 92 WebRtcLocalAudioTrackAdapter* adapter_; 93 94 // The provider of captured data to render. 95 scoped_refptr<WebRtcAudioCapturer> capturer_; 96 97 // The source of the audio track which is used by WebAudio, which provides 98 // data to the audio track when hooking up with WebAudio. 99 scoped_refptr<WebAudioCapturerSource> webaudio_source_; 100 101 // A tagged list of sinks that the audio data is fed to. Tags 102 // indicate tracks that need to be notified that the audio format 103 // has changed. 104 SinkList sinks_; 105 106 // Used to DCHECK that some methods are called on the main render thread. 107 base::ThreadChecker main_render_thread_checker_; 108 109 // Used to DCHECK that some methods are called on the capture audio thread. 110 base::ThreadChecker capture_thread_checker_; 111 112 // Protects |params_| and |sinks_|. 113 mutable base::Lock lock_; 114 115 // Audio parameters of the audio capture stream. 116 // Accessed on only the audio capture thread. 117 media::AudioParameters audio_parameters_; 118 119 // Used to calculate the signal level that shows in the UI. 120 // Accessed on only the audio thread. 121 scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_; 122 123 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); 124 }; 125 126 } // namespace content 127 128 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ 129