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      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   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