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/synchronization/lock.h"
  12 #include "base/threading/thread_checker.h"
  13 #include "content/renderer/media/media_stream_audio_track_sink.h"
  14 #include "content/renderer/media/tagged_list.h"
  15 #include "content/renderer/media/webrtc_audio_device_impl.h"
  16 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
  17 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
  18 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
  19 #include "third_party/libjingle/source/talk/app/webrtc/mediastreamtrack.h"
  20 #include "third_party/libjingle/source/talk/media/base/audiorenderer.h"
  21
  22 namespace cricket {
  23 class AudioRenderer;
  24 }  // namespace cricket
  25
  26 namespace media {
  27 class AudioBus;
  28 }  // namespace media
  29
  30 namespace content {
  31
  32 class MediaStreamAudioSink;
  33 class MediaStreamAudioSinkOwner;
  34 class PeerConnectionAudioSink;
  35 class WebAudioCapturerSource;
  36 class WebRtcAudioCapturer;
  37
  38 // A WebRtcLocalAudioTrack instance contains the implementations of
  39 // MediaStreamTrack and MediaStreamAudioSink.
  40 // When an instance is created, it will register itself as a track to the
  41 // WebRtcAudioCapturer to get the captured data, and forward the data to
  42 // its |sinks_|. The data flow can be stopped by disabling the audio track.
  43 class CONTENT_EXPORT WebRtcLocalAudioTrack
  44     : NON_EXPORTED_BASE(public cricket::AudioRenderer),
  45       NON_EXPORTED_BASE(
  46           public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) {
  47  public:
  48   static scoped_refptr<WebRtcLocalAudioTrack> Create(
  49       const std::string& id,
  50       const scoped_refptr<WebRtcAudioCapturer>& capturer,
  51       WebAudioCapturerSource* webaudio_source,
  52       webrtc::AudioSourceInterface* track_source,
  53       const webrtc::MediaConstraintsInterface* constraints);
  54
  55   // Add a sink to the track. This function will trigger a OnSetFormat()
  56   // call on the |sink|.
  57   // Called on the main render thread.
  58   void AddSink(MediaStreamAudioSink* sink);
  59
  60   // Remove a sink from the track.
  61   // Called on the main render thread.
  62   void RemoveSink(MediaStreamAudioSink* sink);
  63
  64   // Add/remove PeerConnection sink to/from the track.
  65   // TODO(xians): Remove these two methods after PeerConnection can use the
  66   // same sink interface as MediaStreamAudioSink.
  67   void AddSink(PeerConnectionAudioSink* sink);
  68   void RemoveSink(PeerConnectionAudioSink* sink);
  69
  70   // Starts the local audio track. Called on the main render thread and
  71   // should be called only once when audio track is created.
  72   void Start();
  73
  74   // Stops the local audio track. Called on the main render thread and
  75   // should be called only once when audio track going away.
  76   void Stop();
  77
  78   // Method called by the capturer to deliver the capture data.
  79   // Call on the capture audio thread.
  80   void Capture(media::AudioBus* audio_source,
  81                int audio_delay_milliseconds,
  82                int volume,
  83                bool key_pressed);
  84
  85   // Method called by the capturer to set the audio parameters used by source
  86   // of the capture data..
  87   // Call on the capture audio thread.
  88   void OnSetFormat(const media::AudioParameters& params);
  89
  90   blink::WebAudioSourceProvider* audio_source_provider() const {
  91     return source_provider_.get();
  92   }
  93
  94  protected:
  95   WebRtcLocalAudioTrack(
  96       const std::string& label,
  97       const scoped_refptr<WebRtcAudioCapturer>& capturer,
  98       WebAudioCapturerSource* webaudio_source,
  99       webrtc::AudioSourceInterface* track_source,
 100       const webrtc::MediaConstraintsInterface* constraints);
 101
 102   virtual ~WebRtcLocalAudioTrack();
 103
 104  private:
 105   typedef TaggedList<MediaStreamAudioTrackSink> SinkList;
 106
 107   // cricket::AudioCapturer implementation.
 108   virtual void AddChannel(int channel_id) OVERRIDE;
 109   virtual void RemoveChannel(int channel_id) OVERRIDE;
 110
 111   // webrtc::AudioTrackInterface implementation.
 112   virtual webrtc::AudioSourceInterface* GetSource() const OVERRIDE;
 113   virtual cricket::AudioRenderer* GetRenderer() OVERRIDE;
 114
 115   // webrtc::MediaStreamTrack implementation.
 116   virtual std::string kind() const OVERRIDE;
 117
 118   // The provider of captured data to render.
 119   // The WebRtcAudioCapturer is today created by WebRtcAudioDeviceImpl.
 120   scoped_refptr<WebRtcAudioCapturer> capturer_;
 121
 122   // The source of the audio track which is used by WebAudio, which provides
 123   // data to the audio track when hooking up with WebAudio.
 124   scoped_refptr<WebAudioCapturerSource> webaudio_source_;
 125
 126   // The source of the audio track which handles the audio constraints.
 127   // TODO(xians): merge |track_source_| to |capturer_|.
 128   talk_base::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
 129
 130   // A tagged list of sinks that the audio data is fed to. Tags
 131   // indicate tracks that need to be notified that the audio format
 132   // has changed.
 133   SinkList sinks_;
 134
 135   // Used to DCHECK that some methods are called on the main render thread.
 136   base::ThreadChecker main_render_thread_checker_;
 137
 138   // Used to DCHECK that some methods are called on the capture audio thread.
 139   base::ThreadChecker capture_thread_checker_;
 140
 141   // Protects |params_| and |sinks_|.
 142   mutable base::Lock lock_;
 143
 144   // A vector of WebRtc VoE channels that the capturer sends data to.
 145   std::vector<int> voe_channels_;
 146
 147   bool need_audio_processing_;
 148
 149   // Buffers used for temporary storage during capture callbacks.
 150   // Allocated and accessed only on the capture audio thread.
 151   class ConfiguredBuffer;
 152   scoped_ptr<ConfiguredBuffer> buffer_;
 153
 154   // The source provider to feed the track data to other clients like
 155   // WebAudio.
 156   scoped_ptr<WebRtcLocalAudioSourceProvider> source_provider_;
 157
 158   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
 159 };
 160
 161 }  // namespace content
 162
 163 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_