Apply VideoOptions per stream.

talk/media/webrtc/webrtcvideoengine2.h

Index: talk/media/webrtc/webrtcvideoengine2.h
diff --git a/talk/media/webrtc/webrtcvideoengine2.h b/talk/media/webrtc/webrtcvideoengine2.h
index eb2480c4cbde614ff43b6afc8ed6397aebb79787..2be2a13d1fdb22d69c9a9ed862911349d0ac1445 100644
--- a/talk/media/webrtc/webrtcvideoengine2.h
+++ b/talk/media/webrtc/webrtcvideoengine2.h
@@ -202,7 +202,8 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
   bool SetSendRtpHeaderExtensions(
       const std::vector<RtpHeaderExtension>& extensions);
   bool SetMaxSendBandwidth(int bps);
-  bool SetOptions(const VideoOptions& options);
+  void SetSharedOptions(const VideoOptions& options);
+  bool SetOptions(uint32_t ssrc, const VideoOptions& options);
   bool SetRecvCodecs(const std::vector<VideoCodec>& codecs);
   bool SetRecvRtpHeaderExtensions(
       const std::vector<RtpHeaderExtension>& extensions);
@@ -240,7 +241,6 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
         const StreamParams& sp,
         const webrtc::VideoSendStream::Config& config,
         WebRtcVideoEncoderFactory* external_encoder_factory,
-        const VideoOptions& options,
         int max_bitrate_bps,
         const rtc::Optional<VideoCodecSettings>& codec_settings,
         const std::vector<webrtc::RtpExtension>& rtp_extensions,
@@ -466,7 +466,6 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
   };
 
   void Construct(webrtc::Call* call, WebRtcVideoEngine2* engine);
-  void SetDefaultOptions();
 
   bool SendRtp(const uint8_t* data,
                size_t len,
@@ -505,6 +504,7 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
   rtc::CriticalSection capturer_crit_;
   bool signal_cpu_adaptation_ GUARDED_BY(capturer_crit_);
   std::map<uint32_t, VideoCapturer*> capturers_ GUARDED_BY(capturer_crit_);
+  bool disable_prerenderer_smoothing_;
 
   rtc::CriticalSection stream_crit_;
   // Using primary-ssrc (first ssrc) as key.
@@ -523,7 +523,6 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
   std::vector<VideoCodecSettings> recv_codecs_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
   webrtc::Call::Config::BitrateConfig bitrate_config_;
-  VideoOptions options_;
   // TODO(deadbeef): Don't duplicate information between
   // send_params/recv_params, rtp_extensions, options, etc.
   VideoSendParameters send_params_;