Add send transports to individual webrtc::Call streams.

webrtc/video/rtc_event_log_unittest.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 292 matching lines @@
   config->rtp.extensions.push_back(RtpExtension(extension_name, rand()));
 }
 
 // Test for the RtcEventLog class. Dumps some RTP packets to disk, then reads
 // them back to see if they match.
 void LogSessionAndReadBack(size_t rtp_count, unsigned random_seed) {
   std::vector<std::vector<uint8_t>> rtp_packets;
   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);
 
   // Create rtp_count RTP packets containing random data.
   const size_t rtp_header_size = 20;
   for (size_t i = 0; i < rtp_count; i++) {
     size_t packet_size = 1000 + rand() % 30;
     rtp_packets.push_back(std::vector<uint8_t>());
     rtp_packets[i].reserve(packet_size);
     for (size_t j = 0; j < packet_size; j++) {
@@ skipping 95 matching lines @@
 
 TEST(RtcEventLogTest, LogSessionAndReadBack) {
   LogSessionAndReadBack(5, 321);
   LogSessionAndReadBack(8, 3141592653u);
   LogSessionAndReadBack(9, 2718281828u);
 }
 
 }  // namespace webrtc
 
 #endif  // ENABLE_RTC_EVENT_LOG