Enabling rtcp-rsize negotiation and fixing some issues with it.

webrtc/video/video_send_stream.cc

Index: webrtc/video/video_send_stream.cc
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index 58ed646363ebbee899656dbb95731eeeeaa36d2a..401d0a27c72d4508f767275f96c3af658f6001cd 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -68,6 +68,9 @@ std::string VideoSendStream::Config::Rtp::ToString() const {
       ss << ", ";
   }
   ss << ']';
+  ss << ", rtcp_mode: "
+     << (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound"
+                                          : "RtcpMode::kReducedSize");
   ss << ", max_packet_size: " << max_packet_size;
   ss << ", extensions: [";
   for (size_t i = 0; i < extensions.size(); ++i) {