content/renderer/media/webrtc_local_audio_track.h

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