MediaController/Call instantiation.

talk/media/webrtc/webrtcvideoengine2.h

 /*
  * libjingle
  * Copyright 2014 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 85 matching lines @@
                            uint32_t ssrc) override;
 
   VideoRenderer* GetDefaultRenderer() const;
   void SetDefaultRenderer(VideoMediaChannel* channel, VideoRenderer* renderer);
 
  private:
   uint32_t default_recv_ssrc_;
   VideoRenderer* default_renderer_;
 };
 
-// CallFactory, overridden for testing to verify that webrtc::Call is configured
-// properly.
-class WebRtcCallFactory {
+// WebRtcVideoEngine2 is used for the new native WebRTC Video API (webrtc:1667).
+class WebRtcVideoEngine2 {
  public:
-  virtual ~WebRtcCallFactory();
-  virtual webrtc::Call* CreateCall(const webrtc::Call::Config& config);
-};
-
-// WebRtcVideoEngine2 is used for the new native WebRTC Video API (webrtc:1667).
-class WebRtcVideoEngine2 : public sigslot::has_slots<> {
- public:
-  explicit WebRtcVideoEngine2(WebRtcVoiceEngine* voice_engine);
-  virtual ~WebRtcVideoEngine2();
-
-  // Used for testing to be able to check and use the webrtc::Call config.
-  void SetCallFactory(WebRtcCallFactory* call_factory);
+  WebRtcVideoEngine2();
+  ~WebRtcVideoEngine2();
 
   // Basic video engine implementation.
   void Init();
 
   int GetCapabilities();
   bool SetDefaultEncoderConfig(const VideoEncoderConfig& config);
 
-  WebRtcVideoChannel2* CreateChannel(const VideoOptions& options,
-                                     VoiceMediaChannel* voice_channel);
+  WebRtcVideoChannel2* CreateChannel(webrtc::Call* call,
+                                     const VideoOptions& options);
 
   const std::vector<VideoCodec>& codecs() const;
   const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
   void SetLogging(int min_sev, const char* filter);
 
   // Set a WebRtcVideoDecoderFactory for external decoding. Video engine does
   // not take the ownership of |decoder_factory|. The caller needs to make sure
   // that |decoder_factory| outlives the video engine.
   void SetExternalDecoderFactory(WebRtcVideoDecoderFactory* decoder_factory);
   // Set a WebRtcVideoEncoderFactory for external encoding. Video engine does
   // not take the ownership of |encoder_factory|. The caller needs to make sure
   // that |encoder_factory| outlives the video engine.
   virtual void SetExternalEncoderFactory(
       WebRtcVideoEncoderFactory* encoder_factory);
 
   bool EnableTimedRender();
 
   bool FindCodec(const VideoCodec& in);
   bool CanSendCodec(const VideoCodec& in,
                     const VideoCodec& current,
                     VideoCodec* out);
   // Check whether the supplied trace should be ignored.
   bool ShouldIgnoreTrace(const std::string& trace);
 
  private:
   std::vector<VideoCodec> GetSupportedCodecs() const;
 
-  WebRtcVoiceEngine* voice_engine_;
   std::vector<VideoCodec> video_codecs_;
   std::vector<RtpHeaderExtension> rtp_header_extensions_;
 
   bool initialized_;
 
-  WebRtcCallFactory default_call_factory_;
-  WebRtcCallFactory* call_factory_;
-
   WebRtcVideoDecoderFactory* external_decoder_factory_;
   WebRtcVideoEncoderFactory* external_encoder_factory_;
   rtc::scoped_ptr<WebRtcVideoEncoderFactory> simulcast_encoder_factory_;
 };
 
 class WebRtcVideoChannel2 : public rtc::MessageHandler,
                             public VideoMediaChannel,
                             public webrtc::newapi::Transport,
                             public webrtc::LoadObserver {
  public:
-  WebRtcVideoChannel2(WebRtcCallFactory* call_factory,
-                      WebRtcVoiceEngine* voice_engine,
-                      WebRtcVoiceMediaChannel* voice_channel,
+  WebRtcVideoChannel2(webrtc::Call* call,
                       const VideoOptions& options,
                       WebRtcVideoEncoderFactory* external_encoder_factory,
                       WebRtcVideoDecoderFactory* external_decoder_factory);
-  ~WebRtcVideoChannel2();
-  bool Init();
+  ~WebRtcVideoChannel2() override;
 
   // VideoMediaChannel implementation
-  void DetachVoiceChannel() override;
   bool SetSendParameters(const VideoSendParameters& params) override;
   bool SetRecvParameters(const VideoRecvParameters& params) override;
   bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) override;
   bool SetSendCodecs(const std::vector<VideoCodec>& codecs) override;
   bool GetSendCodec(VideoCodec* send_codec) override;
   bool SetSendStreamFormat(uint32 ssrc, const VideoFormat& format) override;
   bool SetRender(bool render) override;
   bool SetSend(bool send) override;
 
   bool AddSendStream(const StreamParams& sp) override;
@@ skipping 305 matching lines @@
 
   void FillSenderStats(VideoMediaInfo* info);
   void FillReceiverStats(VideoMediaInfo* info);
   void FillBandwidthEstimationStats(const webrtc::Call::Stats& stats,
                                     VideoMediaInfo* info);
 
   rtc::ThreadChecker thread_checker_;
 
   uint32_t rtcp_receiver_report_ssrc_;
   bool sending_;
-  rtc::scoped_ptr<webrtc::Call> call_;
-  WebRtcCallFactory* call_factory_;
+  webrtc::Call* call_;

[pbos-webrtc, 2015/09/08 14:47:41]

Call* const

[the sun, 2015/09/08 15:21:35]

Do you mean I should use a reference (&) instead?

 
   uint32_t default_send_ssrc_;
 
   DefaultUnsignalledSsrcHandler default_unsignalled_ssrc_handler_;
   UnsignalledSsrcHandler* const unsignalled_ssrc_handler_;
 
   // Separate list of set capturers used to signal CPU adaptation. These should
   // not be locked while calling methods that take other locks to prevent
   // lock-order inversions.
   rtc::CriticalSection capturer_crit_;
   bool signal_cpu_adaptation_ GUARDED_BY(capturer_crit_);
   std::map<uint32, VideoCapturer*> capturers_ GUARDED_BY(capturer_crit_);
 
   rtc::CriticalSection stream_crit_;
   // Using primary-ssrc (first ssrc) as key.
   std::map<uint32, WebRtcVideoSendStream*> send_streams_
       GUARDED_BY(stream_crit_);
   std::map<uint32, WebRtcVideoReceiveStream*> receive_streams_
       GUARDED_BY(stream_crit_);
   std::set<uint32> send_ssrcs_ GUARDED_BY(stream_crit_);
   std::set<uint32> receive_ssrcs_ GUARDED_BY(stream_crit_);
 
   Settable<VideoCodecSettings> send_codec_;
   std::vector<webrtc::RtpExtension> send_rtp_extensions_;
 
-  WebRtcVoiceMediaChannel* voice_channel_;
-  const int voice_channel_id_;
   WebRtcVideoEncoderFactory* const external_encoder_factory_;
   WebRtcVideoDecoderFactory* const external_decoder_factory_;
   std::vector<VideoCodecSettings> recv_codecs_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
   webrtc::Call::Config::BitrateConfig bitrate_config_;
   VideoOptions options_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_H_