Add ramp-up tests for transport sequence number with and w/o audio.

webrtc/test/direct_transport.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include "webrtc/test/direct_transport.h"
@@ skipping 52 matching lines @@
   packet_event_.Set();
   return true;
 }
 
 bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) {
   fake_network_.SendPacket(data, length);
   packet_event_.Set();
   return true;
 }
 
+int DirectTransport::GetAverageDelayMs() {
+  return fake_network_.AverageDelay();
+}
+
 bool DirectTransport::NetworkProcess(void* transport) {
   return static_cast<DirectTransport*>(transport)->SendPackets();
 }
 
 bool DirectTransport::SendPackets() {
   fake_network_.Process();
   int64_t wait_time_ms = fake_network_.TimeUntilNextProcess();
   if (wait_time_ms > 0) {
     packet_event_.Wait(static_cast<int>(wait_time_ms));
   }
   rtc::CritScope crit(&lock_);
   return shutting_down_ ? false : true;
 }
 }  // namespace test
 }  // namespace webrtc