Add support for negotiating RTCP with reduced size.

talk/media/webrtc/webrtcvideoengine2.cc

Index: talk/media/webrtc/webrtcvideoengine2.cc
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index 05cd6bbcf1a582380d1c35b5aeb3dbdfb98f2b31..a4d680ef564a8b0ad12889e4bba84ed1956a31ad 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -889,20 +889,30 @@ bool WebRtcVideoChannel2::ReceiveCodecsHaveChanged(
   return false;
 }
 
-bool WebRtcVideoChannel2::SetSendParameters(const VideoSendParameters& params) {
+bool WebRtcVideoChannel2::SetSendParameters(
+    const VideoSendParameters& send_params) {
   // TODO(pbos): Refactor this to only recreate the send streams once
   // instead of 4 times.
-  return (SetSendCodecs(params.codecs) &&
-          SetSendRtpHeaderExtensions(params.extensions) &&
-          SetMaxSendBandwidth(params.max_bandwidth_bps) &&
-          SetOptions(params.options));
+  if (!SetSendCodecs(send_params.codecs) ||
+      !SetSendRtpHeaderExtensions(send_params.extensions) ||
+      !SetMaxSendBandwidth(send_params.max_bandwidth_bps) ||
+      !SetOptions(send_params.options)) {
+    return false;
+  }
+  send_params_ = send_params;
+  return true;
 }
 
-bool WebRtcVideoChannel2::SetRecvParameters(const VideoRecvParameters& params) {
+bool WebRtcVideoChannel2::SetRecvParameters(
+    const VideoRecvParameters& recv_params) {
   // TODO(pbos): Refactor this to only recreate the recv streams once
   // instead of twice.
-  return (SetRecvCodecs(params.codecs) &&
-          SetRecvRtpHeaderExtensions(params.extensions));
+  if (!SetRecvCodecs(recv_params.codecs) ||
+      !SetRecvRtpHeaderExtensions(recv_params.extensions)) {
+    return false;
+  }
+  recv_params_ = recv_params;
+  return true;
 }
 
 std::string WebRtcVideoChannel2::CodecSettingsVectorToString(
@@ -1111,14 +1121,10 @@ bool WebRtcVideoChannel2::AddSendStream(const StreamParams& sp) {
   for (uint32 used_ssrc : sp.ssrcs)
     send_ssrcs_.insert(used_ssrc);
 
-  WebRtcVideoSendStream* stream =
-      new WebRtcVideoSendStream(call_.get(),
-                                external_encoder_factory_,
-                                options_,
-                                bitrate_config_.max_bitrate_bps,
-                                send_codec_,
-                                sp,
-                                send_rtp_extensions_);
+  WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
+      call_.get(), external_encoder_factory_, options_,
+      bitrate_config_.max_bitrate_bps, send_codec_, sp, send_rtp_extensions_,
+      send_params_);
 
   uint32 ssrc = sp.first_ssrc();
   DCHECK(ssrc != 0);
@@ -1249,6 +1255,9 @@ void WebRtcVideoChannel2::ConfigureReceiverRtp(
   config->rtp.local_ssrc = rtcp_receiver_report_ssrc_;
 
   config->rtp.extensions = recv_rtp_extensions_;
+  config->rtp.rtcp_mode = recv_params_.rtcp.reduced_size
+                              ? webrtc::newapi::kRtcpReducedSize
+                              : webrtc::newapi::kRtcpCompound;
 
   // TODO(pbos): This protection is against setting the same local ssrc as
   // remote which is not permitted by the lower-level API. RTCP requires a
@@ -1731,7 +1740,10 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
     int max_bitrate_bps,
     const Settable<VideoCodecSettings>& codec_settings,
     const StreamParams& sp,
-    const std::vector<webrtc::RtpExtension>& rtp_extensions)
+    const std::vector<webrtc::RtpExtension>& rtp_extensions,
+    // TODO(pbos): Don't duplicate information between send_params,
+    // rtp_extensions, options, etc.
+    const VideoSendParameters& send_params)
     : ssrcs_(sp.ssrcs),
       ssrc_groups_(sp.ssrc_groups),
       call_(call),
@@ -1755,6 +1767,9 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
                  &parameters_.config.rtp.rtx.ssrcs);
   parameters_.config.rtp.c_name = sp.cname;
   parameters_.config.rtp.extensions = rtp_extensions;
+  // TODO(pbos): Add this to send stream when we support it.
+  // parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size ?
+  //   webrtc::newapi::kRtcpReducedSize : webrtc::newapi::kRtcpCompound;
 
   VideoCodecSettings params;
   if (codec_settings.Get(&params)) {