Remove MediaChannel::GetOptions().

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 202 matching lines @@
   bool MuteStream(uint32 ssrc, bool mute) override;
 
   // Set send/receive RTP header extensions. This must be done before creating
   // streams as it only has effect on future streams.
   bool SetRecvRtpHeaderExtensions(
       const std::vector<RtpHeaderExtension>& extensions) override;
   bool SetSendRtpHeaderExtensions(
       const std::vector<RtpHeaderExtension>& extensions) override;
   bool SetMaxSendBandwidth(int bps) override;
   bool SetOptions(const VideoOptions& options) override;
-  bool GetOptions(VideoOptions* options) const override {
-    *options = options_;
-    return true;
-  }
   void SetInterface(NetworkInterface* iface) override;
   void UpdateAspectRatio(int ratio_w, int ratio_h) override;
 
   void OnMessage(rtc::Message* msg) override;
 
   void OnLoadUpdate(Load load) override;
 
   // Implemented for VideoMediaChannelTest.
   bool sending() const { return sending_; }
   uint32 GetDefaultSendChannelSsrc() { return default_send_ssrc_; }
@@ skipping 306 matching lines @@
   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_