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

webrtc/test/fake_network_pipe.cc

 /*
  *  Copyright (c) 2012 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 128 matching lines @@
 
   return static_cast<float>(dropped_packets_) /
       (sent_packets_ + dropped_packets_);
 }
 
 int FakeNetworkPipe::AverageDelay() {
   rtc::CritScope crit(&lock_);
   if (sent_packets_ == 0)
     return 0;
 
-  return total_packet_delay_ / static_cast<int>(sent_packets_);
+  return static_cast<int>(total_packet_delay_ /
+                          static_cast<int64_t>(sent_packets_));
 }
 
 void FakeNetworkPipe::Process() {
   int64_t time_now = clock_->TimeInMilliseconds();
   std::queue<NetworkPacket*> packets_to_deliver;
   {
     rtc::CritScope crit(&lock_);
     // Check the capacity link first.
     while (capacity_link_.size() > 0 &&
            time_now >= capacity_link_.front()->arrival_time()) {
@@ skipping 49 matching lines @@
 int64_t FakeNetworkPipe::TimeUntilNextProcess() const {
   rtc::CritScope crit(&lock_);
   const int64_t kDefaultProcessIntervalMs = 30;
   if (capacity_link_.size() == 0 || delay_link_.size() == 0)
     return kDefaultProcessIntervalMs;
   return std::max<int64_t>(next_process_time_ - clock_->TimeInMilliseconds(),
                            0);
 }
 
 }  // namespace webrtc