Rebased changes to 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 eff1a612c82a74708e0e17a2ee86655bbf212e50..d76abcecf0c0c9d791b70450ff354b6772895b91 100644
--- a/talk/media/webrtc/webrtcvideoengine2.h
+++ b/talk/media/webrtc/webrtcvideoengine2.h
@@ -230,7 +230,8 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
   bool MuteStream(uint32_t ssrc, bool mute);
 
   void SetMaxSendBandwidth(int bps);
-  void SetOptions(const VideoOptions& options);
+  void SetSharedOptions(const VideoOptions& options);
+  bool SetOptions(uint32_t ssrc, const VideoOptions& options);
 
   void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config,
                             const StreamParams& sp) const;
@@ -254,7 +255,6 @@ 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,
@@ -472,7 +472,6 @@ class WebRtcVideoChannel2 : public rtc::MessageHandler,
   };
 
   void Construct(webrtc::Call* call, WebRtcVideoEngine2* engine);
-  void SetDefaultOptions();
 
   bool SendRtp(const uint8_t* data,
                size_t len,
@@ -511,6 +510,7 @@ 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.
@@ -529,7 +529,6 @@ 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_;