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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/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_
    164