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 "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
  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   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_