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      1 // Copyright (c) 2012 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_MEDIA_STREAM_DEPENDENCY_FACTORY_H_
      6 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_DEPENDENCY_FACTORY_H_
      7 
      8 #include <string>
      9 
     10 #include "base/basictypes.h"
     11 #include "base/memory/ref_counted.h"
     12 #include "base/threading/thread.h"
     13 #include "content/common/content_export.h"
     14 #include "content/renderer/media/media_stream_extra_data.h"
     15 #include "content/renderer/p2p/socket_dispatcher.h"
     16 #include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h"
     17 #include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h"
     18 
     19 namespace base {
     20 class WaitableEvent;
     21 }
     22 
     23 namespace talk_base {
     24 class NetworkManager;
     25 class PacketSocketFactory;
     26 class Thread;
     27 }
     28 
     29 namespace webrtc {
     30 class PeerConnection;
     31 }
     32 
     33 namespace WebKit {
     34 class WebFrame;
     35 class WebMediaConstraints;
     36 class WebMediaStream;
     37 class WebRTCPeerConnectionHandler;
     38 class WebRTCPeerConnectionHandlerClient;
     39 }
     40 
     41 namespace content {
     42 
     43 class IpcNetworkManager;
     44 class IpcPacketSocketFactory;
     45 class VideoCaptureImplManager;
     46 class WebRtcAudioCapturer;
     47 class WebRtcAudioDeviceImpl;
     48 class WebRtcLoggingHandlerImpl;
     49 class WebRtcLoggingMessageFilter;
     50 struct StreamDeviceInfo;
     51 
     52 #if defined(GOOGLE_TV)
     53 class RTCVideoDecoderFactoryTv;
     54 #endif
     55 
     56 // Object factory for RTC MediaStreams and RTC PeerConnections.
     57 class CONTENT_EXPORT MediaStreamDependencyFactory
     58     : NON_EXPORTED_BASE(public base::NonThreadSafe) {
     59  public:
     60   // MediaSourcesCreatedCallback is used in CreateNativeMediaSources.
     61   typedef base::Callback<void(WebKit::WebMediaStream* web_stream,
     62                               bool live)> MediaSourcesCreatedCallback;
     63   MediaStreamDependencyFactory(
     64       VideoCaptureImplManager* vc_manager,
     65       P2PSocketDispatcher* p2p_socket_dispatcher);
     66   virtual ~MediaStreamDependencyFactory();
     67 
     68   // Create a RTCPeerConnectionHandler object that implements the
     69   // WebKit WebRTCPeerConnectionHandler interface.
     70   WebKit::WebRTCPeerConnectionHandler* CreateRTCPeerConnectionHandler(
     71       WebKit::WebRTCPeerConnectionHandlerClient* client);
     72 
     73   // CreateNativeMediaSources creates libjingle representations of
     74   // the underlying sources to the tracks in |web_stream|.
     75   // |sources_created| is invoked when the sources have either been created and
     76   // transitioned to a live state or failed.
     77   // The libjingle sources is stored in the extra data field of
     78   // WebMediaStreamSource.
     79   // |audio_constraints| and |video_constraints| set parameters for the sources.
     80   void CreateNativeMediaSources(
     81       int render_view_id,
     82       const WebKit::WebMediaConstraints& audio_constraints,
     83       const WebKit::WebMediaConstraints& video_constraints,
     84       WebKit::WebMediaStream* web_stream,
     85       const MediaSourcesCreatedCallback& sources_created);
     86 
     87   // Creates a libjingle representation of a MediaStream and stores
     88   // it in the extra data field of |web_stream|.
     89   void CreateNativeLocalMediaStream(
     90       WebKit::WebMediaStream* web_stream);
     91 
     92   // Creates a libjingle representation of a MediaStream and stores
     93   // it in the extra data field of |web_stream|.
     94   // |stream_stopped| is a callback that is run when a MediaStream have been
     95   // stopped.
     96   void CreateNativeLocalMediaStream(
     97       WebKit::WebMediaStream* web_stream,
     98       const MediaStreamExtraData::StreamStopCallback& stream_stop);
     99 
    100   // Adds a libjingle representation of a MediaStreamTrack to |stream| based
    101   // on the source of |track|.
    102   bool AddNativeMediaStreamTrack(const WebKit::WebMediaStream& stream,
    103                                  const WebKit::WebMediaStreamTrack& track);
    104 
    105   // Creates and adds libjingle representation of a MediaStreamTrack to |stream|
    106   // based on the desired |track_id| and |capturer|.
    107   bool AddNativeVideoMediaTrack(const std::string& track_id,
    108                                 WebKit::WebMediaStream* stream,
    109                                 cricket::VideoCapturer* capturer);
    110 
    111   bool RemoveNativeMediaStreamTrack(const WebKit::WebMediaStream& stream,
    112                                     const WebKit::WebMediaStreamTrack& track);
    113 
    114   // Asks the libjingle PeerConnection factory to create a libjingle
    115   // PeerConnection object.
    116   // The PeerConnection object is owned by PeerConnectionHandler.
    117   virtual scoped_refptr<webrtc::PeerConnectionInterface>
    118       CreatePeerConnection(
    119           const webrtc::PeerConnectionInterface::IceServers& ice_servers,
    120           const webrtc::MediaConstraintsInterface* constraints,
    121           WebKit::WebFrame* web_frame,
    122           webrtc::PeerConnectionObserver* observer);
    123 
    124   // Creates a libjingle representation of a Session description. Used by a
    125   // RTCPeerConnectionHandler instance.
    126   virtual webrtc::SessionDescriptionInterface* CreateSessionDescription(
    127       const std::string& type,
    128       const std::string& sdp,
    129       webrtc::SdpParseError* error);
    130 
    131   // Creates a libjingle representation of an ice candidate.
    132   virtual webrtc::IceCandidateInterface* CreateIceCandidate(
    133       const std::string& sdp_mid,
    134       int sdp_mline_index,
    135       const std::string& sdp);
    136 
    137   WebRtcAudioDeviceImpl* GetWebRtcAudioDevice();
    138 
    139 #if defined(GOOGLE_TV)
    140   RTCVideoDecoderFactoryTv* decoder_factory_tv() { return decoder_factory_tv_; }
    141 #endif
    142 
    143  protected:
    144   // Asks the PeerConnection factory to create a Local MediaStream object.
    145   virtual scoped_refptr<webrtc::MediaStreamInterface>
    146       CreateLocalMediaStream(const std::string& label);
    147 
    148   // Asks the PeerConnection factory to create a Local Audio Source.
    149   virtual scoped_refptr<webrtc::AudioSourceInterface>
    150       CreateLocalAudioSource(
    151           const webrtc::MediaConstraintsInterface* constraints);
    152 
    153   // Asks the PeerConnection factory to create a Local Video Source.
    154   virtual scoped_refptr<webrtc::VideoSourceInterface>
    155       CreateLocalVideoSource(
    156           int video_session_id,
    157           bool is_screen_cast,
    158           const webrtc::MediaConstraintsInterface* constraints);
    159 
    160   // Creates a media::AudioCapturerSource with an implementation that is
    161   // specific for a WebAudio source. The created WebAudioCapturerSource
    162   // instance will function as audio source instead of the default
    163   // WebRtcAudioCapturer.
    164   virtual scoped_refptr<WebRtcAudioCapturer> CreateWebAudioSource(
    165       WebKit::WebMediaStreamSource* source);
    166 
    167   // Asks the PeerConnection factory to create a Local AudioTrack object.
    168   virtual scoped_refptr<webrtc::AudioTrackInterface>
    169       CreateLocalAudioTrack(const std::string& id,
    170                             const scoped_refptr<WebRtcAudioCapturer>& capturer,
    171                             webrtc::AudioSourceInterface* source);
    172 
    173   // Asks the PeerConnection factory to create a Local VideoTrack object.
    174   virtual scoped_refptr<webrtc::VideoTrackInterface>
    175       CreateLocalVideoTrack(const std::string& id,
    176                             webrtc::VideoSourceInterface* source);
    177 
    178   // Asks the PeerConnection factory to create a Local VideoTrack object with
    179   // the video source using |capturer|.
    180   virtual scoped_refptr<webrtc::VideoTrackInterface>
    181       CreateLocalVideoTrack(const std::string& id,
    182                             cricket::VideoCapturer* capturer);
    183 
    184   virtual bool EnsurePeerConnectionFactory();
    185   virtual bool PeerConnectionFactoryCreated();
    186 
    187   // Returns a new capturer or existing capturer based on the |render_view_id|
    188   // and |device_info|. When the |render_view_id| and |device_info| are valid,
    189   // it reuses existing capture if any; otherwise it creates a new capturer.
    190   virtual scoped_refptr<WebRtcAudioCapturer> MaybeCreateAudioCapturer(
    191       int render_view_id, const StreamDeviceInfo& device_info);
    192 
    193  private:
    194   // Creates and deletes |pc_factory_|, which in turn is used for
    195   // creating PeerConnection objects.
    196   bool CreatePeerConnectionFactory();
    197 
    198   void InitializeWorkerThread(talk_base::Thread** thread,
    199                               base::WaitableEvent* event);
    200 
    201   void CreateIpcNetworkManagerOnWorkerThread(base::WaitableEvent* event);
    202   void DeleteIpcNetworkManager();
    203   void CleanupPeerConnectionFactory();
    204 
    205   void CreateWebRtcLoggingHandler(WebRtcLoggingMessageFilter* filter,
    206                                   const std::string& app_session_id,
    207                                   const std::string& app_url);
    208 
    209   // We own network_manager_, must be deleted on the worker thread.
    210   // The network manager uses |p2p_socket_dispatcher_|.
    211   IpcNetworkManager* network_manager_;
    212   scoped_ptr<IpcPacketSocketFactory> socket_factory_;
    213 
    214   scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
    215 
    216 #if defined(GOOGLE_TV)
    217   // |pc_factory_| will hold the ownership of this object, and |pc_factory_|
    218   // outlives this object. Thus weak pointer is sufficient.
    219   RTCVideoDecoderFactoryTv* decoder_factory_tv_;
    220 #endif
    221 
    222   scoped_refptr<VideoCaptureImplManager> vc_manager_;
    223   scoped_refptr<P2PSocketDispatcher> p2p_socket_dispatcher_;
    224   scoped_refptr<WebRtcAudioDeviceImpl> audio_device_;
    225 
    226   // PeerConnection threads. signaling_thread_ is created from the
    227   // "current" chrome thread.
    228   talk_base::Thread* signaling_thread_;
    229   talk_base::Thread* worker_thread_;
    230   base::Thread chrome_worker_thread_;
    231 
    232   DISALLOW_COPY_AND_ASSIGN(MediaStreamDependencyFactory);
    233 };
    234 
    235 }  // namespace content
    236 
    237 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_DEPENDENCY_FACTORY_H_
    238