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

talk/app/webrtc/webrtcsdp_unittest.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 135 matching lines @@
     "a=candidate:a0+B/3 1 udp 2130706432 74.125.127.126 2345 typ srflx "
     "raddr 192.168.1.5 rport 2346 "
     "generation 2\r\n"
     "a=candidate:a0+B/3 2 udp 2130706432 74.125.127.126 2347 typ srflx "
     "raddr 192.168.1.5 rport 2348 "
     "generation 2\r\n"
     "a=ice-ufrag:ufrag_voice\r\na=ice-pwd:pwd_voice\r\n"
     "a=mid:audio_content_name\r\n"
     "a=sendrecv\r\n"
     "a=rtcp-mux\r\n"
+    "a=rtcp-rsize\r\n"
     "a=crypto:1 AES_CM_128_HMAC_SHA1_32 "
     "inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 "
     "dummy_session_params\r\n"
     "a=rtpmap:111 opus/48000/2\r\n"
     "a=rtpmap:103 ISAC/16000\r\n"
     "a=rtpmap:104 ISAC/32000\r\n"
     "a=ssrc:1 cname:stream_1_cname\r\n"
     "a=ssrc:1 msid:local_stream_1 audio_track_id_1\r\n"
     "a=ssrc:1 mslabel:local_stream_1\r\n"
     "a=ssrc:1 label:audio_track_id_1\r\n"
@@ skipping 47 matching lines @@
     "s=-\r\n"
     "t=0 0\r\n"
     "a=msid-semantic: WMS local_stream_1 local_stream_2\r\n"
     "m=audio 9 RTP/SAVPF 111 103 104\r\n"
     "c=IN IP4 0.0.0.0\r\n"
     "a=rtcp:9 IN IP4 0.0.0.0\r\n"
     "a=ice-ufrag:ufrag_voice\r\na=ice-pwd:pwd_voice\r\n"
     "a=mid:audio_content_name\r\n"
     "a=sendrecv\r\n"
     "a=rtcp-mux\r\n"
+    "a=rtcp-rsize\r\n"
     "a=crypto:1 AES_CM_128_HMAC_SHA1_32 "
     "inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 "
     "dummy_session_params\r\n"
     "a=rtpmap:111 opus/48000/2\r\n"
     "a=rtpmap:103 ISAC/16000\r\n"
     "a=rtpmap:104 ISAC/32000\r\n"
     "a=ssrc:1 cname:stream_1_cname\r\n"
     "a=ssrc:1 msid:local_stream_1 audio_track_id_1\r\n"
     "a=ssrc:1 mslabel:local_stream_1\r\n"
     "a=ssrc:1 label:audio_track_id_1\r\n"
@@ skipping 464 matching lines @@
       JsepIceCandidate jice(mline_id,
                             mline_index,
                             candidates_.at(i));
       jdesc_.AddCandidate(&jice);
     }
   }
 
   AudioContentDescription* CreateAudioContentDescription() {
     AudioContentDescription* audio = new AudioContentDescription();
     audio->set_rtcp_mux(true);
+    audio->set_rtcp_reduced_size(true);
     StreamParams audio_stream1;
     audio_stream1.id = kAudioTrackId1;
     audio_stream1.cname = kStream1Cname;
     audio_stream1.sync_label = kStreamLabel1;
     audio_stream1.ssrcs.push_back(kAudioTrack1Ssrc);
     audio->AddStream(audio_stream1);
     StreamParams audio_stream2;
     audio_stream2.id = kAudioTrackId2;
     audio_stream2.cname = kStream2Cname;
     audio_stream2.sync_label = kStreamLabel2;
@@ skipping 11 matching lines @@
                                       const MCD* cd2) {
     // type
     EXPECT_EQ(cd1->type(), cd1->type());
 
     // content direction
     EXPECT_EQ(cd1->direction(), cd2->direction());
 
     // rtcp_mux
     EXPECT_EQ(cd1->rtcp_mux(), cd2->rtcp_mux());
 
+    // rtcp_reduced_size
+    EXPECT_EQ(cd1->rtcp_reduced_size(), cd2->rtcp_reduced_size());
+
     // cryptos
     EXPECT_EQ(cd1->cryptos().size(), cd2->cryptos().size());
     if (cd1->cryptos().size() != cd2->cryptos().size()) {
       ADD_FAILURE();
       return;
     }
     for (size_t i = 0; i< cd1->cryptos().size(); ++i) {
       const CryptoParams c1 = cd1->cryptos().at(i);
       const CryptoParams c2 = cd2->cryptos().at(i);
       EXPECT_TRUE(c1.Matches(c2));
@@ skipping 1953 matching lines @@
       const cricket::MediaContentDescription* mdesc =
           static_cast<const cricket::MediaContentDescription*>(
               desc->contents()[i].description);
       EXPECT_EQ(media_types[media_content_in_sdp[i]], mdesc->type());
     }
 
     std::string serialized_sdp = webrtc::SdpSerialize(jdesc);
     EXPECT_EQ(sdp_string, serialized_sdp);
   }
 }