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

webrtc/test/direct_transport.h

 /*
  *  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.
  */
 #ifndef WEBRTC_TEST_DIRECT_TRANSPORT_H_
@@ skipping 28 matching lines @@
 
   virtual void StopSending();
   // TODO(holmer): Look into moving this to the constructor.
   virtual void SetReceiver(PacketReceiver* receiver);
 
   bool SendRtp(const uint8_t* data,
                size_t length,
                const PacketOptions& options) override;
   bool SendRtcp(const uint8_t* data, size_t length) override;
 
+  int GetAverageDelayMs();
+
  private:
   static bool NetworkProcess(void* transport);
   bool SendPackets();
 
   rtc::CriticalSection lock_;
   Call* const send_call_;
   rtc::Event packet_event_;
   rtc::PlatformThread thread_;
   Clock* const clock_;
 
   bool shutting_down_;
 
   FakeNetworkPipe fake_network_;
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_DIRECT_TRANSPORT_H_