Call OnSentPacket for all packets sent in the test framework.

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 45 matching lines @@
                        size_t* padding_sent,
                        size_t* media_sent) const;
 
   void ReportResult(const std::string& measurement,
                     size_t value,
                     const std::string& units) const;
   void TriggerTestDone();
 
   rtc::Event event_;
   Clock* const clock_;
+  FakeNetworkPipe::Config forward_transport_config_;
   const size_t num_streams_;
   const bool rtx_;
   const bool red_;
   VideoSendStream* send_stream_;
 
  private:
   typedef std::map<uint32_t, uint32_t> SsrcMap;
 
   Call::Config GetSenderCallConfig() override;
   void OnStreamsCreated(
@@ skipping 39 matching lines @@
   static const int kExpectedHighBitrateBps = 60000;
   static const int kLowBandwidthLimitBps = 20000;
   static const int kExpectedLowBitrateBps = 20000;
   enum TestStates { kFirstRampup, kLowRate, kSecondRampup };
 
   Call::Config GetReceiverCallConfig() override;
 
   std::string GetModifierString() const;
   void EvolveTestState(int bitrate_bps, bool suspended);
 
-  FakeNetworkPipe::Config forward_transport_config_;
   TestStates test_state_;
   int64_t state_start_ms_;
   int64_t interval_start_ms_;
   int sent_bytes_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_VIDEO_RAMPUP_TESTS_H_