Fix issue where the first audio packets significantly impacts initial BWE negatively.

webrtc/video/rampup_tests.h

 /*
  *  Copyright (c) 2014 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 110 matching lines @@
   static const unsigned int kExpectedHighBitrateBps = 60000;
   static const unsigned int kLowBandwidthLimitBps = 20000;
   static const unsigned int kExpectedLowBitrateBps = 20000;
   enum TestStates { kFirstRampup, kLowRate, kSecondRampup };
 
   Clock* const clock_;
   const size_t number_of_streams_;
   const bool rtx_used_;
   const rtc::scoped_ptr<EventWrapper> test_done_;
   const rtc::scoped_ptr<RtpHeaderParser> rtp_parser_;
+  const rtc::scoped_ptr<RTPPayloadRegistry> payload_registry_;
   rtc::scoped_ptr<RtpRtcp> rtp_rtcp_;
   internal::TransportAdapter feedback_transport_;
   const rtc::scoped_ptr<ReceiveStatistics> receive_stats_;
   rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
 
   rtc::CriticalSection crit_;
   VideoSendStream* send_stream_ GUARDED_BY(crit_);
   FakeNetworkPipe::Config forward_transport_config_ GUARDED_BY(crit_);
   TestStates test_state_ GUARDED_BY(crit_);
   int64_t state_start_ms_ GUARDED_BY(crit_);
   int64_t interval_start_ms_ GUARDED_BY(crit_);
   unsigned int last_remb_bps_ GUARDED_BY(crit_);
   size_t sent_bytes_ GUARDED_BY(crit_);
   size_t total_overuse_bytes_ GUARDED_BY(crit_);
   bool suspended_in_stats_ GUARDED_BY(crit_);
 };
 
 class RampUpTest : public test::CallTest {
  protected:
   void RunRampUpTest(size_t num_streams,
                      unsigned int start_bitrate_bps,
                      const std::string& extension_type,
                      bool rtx,
                      bool red);
 
   void RunRampUpDownUpTest(size_t number_of_streams, bool rtx, bool red);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_VIDEO_RAMPUP_TESTS_H_