Fix bug in ramp-up tests stats where rtx was accounted for in the media ssrc.

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 32 matching lines @@
                const std::string& extension_type,
                bool rtx,
                bool red);
   ~RampUpTester() override;
 
   void PerformTest() override;
 
  protected:
   virtual bool PollStats();
 
-  void GetStats(const VideoSendStream::StreamStats& stream,
-                size_t* total_packets_sent,
-                size_t* total_sent,
-                size_t* padding_sent,
-                size_t* media_sent) const;
+  void AccumulateStats(const VideoSendStream::StreamStats& stream,
+                       size_t* total_packets_sent,
+                       size_t* total_sent,
+                       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_;
   const size_t num_streams_;
   const bool rtx_;
@@ skipping 55 matching lines @@
   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_