MediaController/Call instantiation.

talk/media/webrtc/webrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2004 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 82 matching lines @@
 
  public:
   WebRtcVoiceEngine();
   // Dependency injection for testing.
   WebRtcVoiceEngine(VoEWrapper* voe_wrapper, VoETraceWrapper* tracing);
   ~WebRtcVoiceEngine();
   bool Init(rtc::Thread* worker_thread);
   void Terminate();
 
   int GetCapabilities();
-  VoiceMediaChannel* CreateChannel(const AudioOptions& options);
+  webrtc::VoiceEngine* GetVoE() { return voe()->engine(); }
+  VoiceMediaChannel* CreateChannel(webrtc::Call* call,
+                                   const AudioOptions& options);
 
   AudioOptions GetOptions() const { return options_; }
   bool SetOptions(const AudioOptions& options);
   bool SetDelayOffset(int offset);
   bool SetDevices(const Device* in_device, const Device* out_device);
   bool GetOutputVolume(int* level);
   bool SetOutputVolume(int level);
   int GetInputLevel();
   bool SetLocalMonitor(bool enable);
 
@@ skipping 159 matching lines @@
   Settable<bool> extended_filter_aec_;
   Settable<bool> delay_agnostic_aec_;
   Settable<bool> experimental_ns_;
 };
 
 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
 // WebRtc Voice Engine.
 class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
                                 public webrtc::Transport {
  public:
-  explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine);
+  explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
+                                   webrtc::Call* call);
   ~WebRtcVoiceMediaChannel() override;
 
   int voe_channel() const { return voe_channel_; }
   bool valid() const { return voe_channel_ != -1; }
   const AudioOptions& options() const { return options_; }
 
   bool SetSendParameters(const AudioSendParameters& params) override;
   bool SetRecvParameters(const AudioRecvParameters& params) override;
   bool SetOptions(const AudioOptions& options) override;
   bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) override;
@@ skipping 55 matching lines @@
     return VoiceMediaChannel::SendRtcp(&packet) ? static_cast<int>(len) : -1;
   }
 
   bool FindSsrc(int channel_num, uint32* ssrc);
   void OnError(uint32 ssrc, int error);
 
   bool sending() const { return send_ != SEND_NOTHING; }
   int GetReceiveChannelNum(uint32 ssrc) const;
   int GetSendChannelNum(uint32 ssrc) const;
 
-  void SetCall(webrtc::Call* call);
-
  private:
   bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer);
   bool MuteStream(uint32 ssrc, bool mute);
   WebRtcVoiceEngine* engine() { return engine_; }
   int GetLastEngineError() { return engine()->GetLastEngineError(); }
   int GetOutputLevel(int channel);
   bool GetRedSendCodec(const AudioCodec& red_codec,
                        const std::vector<AudioCodec>& all_codecs,
                        webrtc::CodecInst* send_codec);
   bool EnableRtcp(int channel);
@@ skipping 24 matching lines @@
     return options_.conference_mode.GetWithDefaultIfUnset(false);
   }
   bool IsDefaultChannel(int channel_id) const {
     return channel_id == voe_channel();
   }
   bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
   bool SetSendBitrateInternal(int bps);
 
   bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
                           const RtpHeaderExtension* extension);
-  void TryAddAudioRecvStream(uint32 ssrc);
-  void TryRemoveAudioRecvStream(uint32 ssrc);
+  void RecreateAudioReceiveStreams();
+  void AddAudioReceiveStream(uint32 ssrc);
+  void RemoveAudioReceiveStream(uint32 ssrc);
   bool SetRecvCodecsInternal(const std::vector<AudioCodec>& new_codecs);
 
   bool SetChannelRecvRtpHeaderExtensions(
     int channel_id,
     const std::vector<RtpHeaderExtension>& extensions);
   bool SetChannelSendRtpHeaderExtensions(
     int channel_id,
     const std::vector<RtpHeaderExtension>& extensions);
 
   rtc::ThreadChecker thread_checker_;
 
-  WebRtcVoiceEngine* engine_;
+  WebRtcVoiceEngine* const engine_;
   const int voe_channel_;
   rtc::scoped_ptr<WebRtcSoundclipStream> ringback_tone_;
   std::set<int> ringback_channels_;  // channels playing ringback
   std::vector<AudioCodec> recv_codecs_;
   std::vector<AudioCodec> send_codecs_;
   rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
   bool send_bitrate_setting_;
   int send_bitrate_bps_;
   AudioOptions options_;
   bool dtmf_allowed_;
   bool desired_playout_;
   bool nack_enabled_;
   bool playout_;
   bool typing_noise_detected_;
   SendFlags desired_send_;
   SendFlags send_;
-  webrtc::Call* call_;
+  webrtc::Call* const call_;
 
   // send_channels_ contains the channels which are being used for sending.
   // When the default channel (voe_channel) is used for sending, it is
   // contained in send_channels_, otherwise not.
   ChannelMap send_channels_;
   std::vector<RtpHeaderExtension> send_extensions_;
   uint32 default_receive_ssrc_;
   // Note the default channel (voe_channel()) can reside in both
   // receive_channels_ and send_channels_ in non-conference mode and in that
   // case it will only be there if a non-zero default_receive_ssrc_ is set.
   ChannelMap receive_channels_;  // for multiple sources
   std::map<uint32, webrtc::AudioReceiveStream*> receive_streams_;
   std::map<uint32, StreamParams> receive_stream_params_;
   // receive_channels_ can be read from WebRtc callback thread.  Access from
   // the WebRtc thread must be synchronized with edits on the worker thread.
   // Reads on the worker thread are ok.
   std::vector<RtpHeaderExtension> receive_extensions_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   // Do not lock this on the VoE media processor thread; potential for deadlock
   // exists.
   mutable rtc::CriticalSection receive_channels_cs_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_