Add send transports to individual webrtc::Call streams.

webrtc/video/rtc_event_log_unittest.cc

Index: webrtc/video/rtc_event_log_unittest.cc
diff --git a/webrtc/video/rtc_event_log_unittest.cc b/webrtc/video/rtc_event_log_unittest.cc
index 0c18e750cc79cbd5938890211b8b64d43ee45135..8a28e2fa4174b4eb6becc17ba235dd4d0bc86d2b 100644
--- a/webrtc/video/rtc_event_log_unittest.cc
+++ b/webrtc/video/rtc_event_log_unittest.cc
@@ -310,8 +310,8 @@ void LogSessionAndReadBack(size_t rtp_count, unsigned random_seed) {
   std::vector<uint8_t> incoming_rtcp_packet;
   std::vector<uint8_t> outgoing_rtcp_packet;
 
-  VideoReceiveStream::Config receiver_config;
-  VideoSendStream::Config sender_config;
+  VideoReceiveStream::Config receiver_config(nullptr);
+  VideoSendStream::Config sender_config(nullptr);
 
   srand(random_seed);