Add send transports to individual webrtc::Call streams.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1273363005

Cr-Original-Commit-Position: refs/heads/master@{#9807}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 4fbae2b79134572135d9d5fe35a7d1ccdeea3a4d
diff --git a/video_receive_stream.h b/video_receive_stream.h
index 9525336..e613006 100644
--- a/video_receive_stream.h
+++ b/video_receive_stream.h
@@ -86,6 +86,10 @@
   };
 
   struct Config {
+    Config() = delete;
+    explicit Config(newapi::Transport* rtcp_send_transport)
+        : rtcp_send_transport(rtcp_send_transport) {}
+
     std::string ToString() const;
 
     // Decoders for every payload that we can receive.
@@ -137,6 +141,9 @@
       std::vector<RtpExtension> extensions;
     } rtp;
 
+    // Transport for outgoing packets (RTCP).
+    newapi::Transport* rtcp_send_transport = nullptr;
+
     // VideoRenderer will be called for each decoded frame. 'nullptr' disables
     // rendering of this stream.
     VideoRenderer* renderer = nullptr;