Adding reduced size RTCP configuration down to the video stream level.

talk/app/webrtc/webrtcsdp.cc

 /*
  * libjingle
  * Copyright 2011 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 104 matching lines @@
 static const char kLineTypeTimeZone = 'z';
 static const char kLineTypeEncryptionKey = 'k';
 static const char kLineTypeMedia = 'm';
 static const char kLineTypeConnection = 'c';
 static const char kLineTypeAttributes = 'a';
 
 // Attributes
 static const char kAttributeGroup[] = "group";
 static const char kAttributeMid[] = "mid";
 static const char kAttributeRtcpMux[] = "rtcp-mux";
+static const char kAttributeRtcpReducedSize[] = "rtcp-rsize";
 static const char kAttributeSsrc[] = "ssrc";
 static const char kSsrcAttributeCname[] = "cname";
 static const char kAttributeExtmap[] = "extmap";
 // draft-alvestrand-mmusic-msid-01
 // a=msid-semantic: WMS
 static const char kAttributeMsidSemantics[] = "msid-semantic";
 static const char kMediaStreamSemantic[] = "WMS";
 static const char kSsrcAttributeMsid[] = "msid";
 static const char kDefaultMsid[] = "default";
 static const char kSsrcAttributeMslabel[] = "mslabel";
@@ skipping 1258 matching lines @@
   }
   AddLine(os.str(), message);
 
   // RFC 5761
   // a=rtcp-mux
   if (media_desc->rtcp_mux()) {
     InitAttrLine(kAttributeRtcpMux, &os);
     AddLine(os.str(), message);
   }
 
+  // RFC 5506
+  // a=rtcp-rsize
+  if (media_desc->rtcp_reduced_size()) {
+    InitAttrLine(kAttributeRtcpReducedSize, &os);
+    AddLine(os.str(), message);
+  }
+
   // RFC 4568
   // a=crypto:<tag> <crypto-suite> <key-params> [<session-params>]
   for (std::vector<CryptoParams>::const_iterator it =
            media_desc->cryptos().begin();
        it != media_desc->cryptos().end(); ++it) {
     InitAttrLine(kAttributeCrypto, &os);
     os << kSdpDelimiterColon << it->tag << " " << it->cipher_suite << " "
        << it->key_params;
     if (!it->session_params.empty()) {
       os << " " << it->session_params;
@@ skipping 1133 matching lines @@
       if (!AddSctpDataCodec(static_cast<DataContentDescription*>(media_desc),
                             sctp_port)) {
         return false;
       }
     } else if (IsRtp(protocol)) {
       //
       // RTP specific attrubtes
       //
       if (HasAttribute(line, kAttributeRtcpMux)) {
         media_desc->set_rtcp_mux(true);
+      } else if (HasAttribute(line, kAttributeRtcpReducedSize)) {
+        media_desc->set_rtcp_reduced_size(true);
       } else if (HasAttribute(line, kAttributeSsrcGroup)) {
         if (!ParseSsrcGroupAttribute(line, &ssrc_groups, error)) {
           return false;
         }
       } else if (HasAttribute(line, kAttributeSsrc)) {
         if (!ParseSsrcAttribute(line, &ssrc_infos, error)) {
           return false;
         }
       } else if (HasAttribute(line, kAttributeCrypto)) {
         if (!ParseCryptoAttribute(line, media_desc, error)) {
@@ skipping 491 matching lines @@
     UpdateCodec<AudioContentDescription, cricket::AudioCodec>(
         media_desc, payload_type, feedback_param);
   } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
     UpdateCodec<VideoContentDescription, cricket::VideoCodec>(
         media_desc, payload_type, feedback_param);
   }
   return true;
 }
 
 }  // namespace webrtc