Don't report perf metrics for packet loss ramp-up tests.

webrtc/call/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 53 matching lines @@
 
   webrtc::RtcEventLogNullImpl event_log_;
   rtc::Event stop_event_;
   Clock* const clock_;
   FakeNetworkPipe::Config forward_transport_config_;
   const size_t num_video_streams_;
   const size_t num_audio_streams_;
   const size_t num_flexfec_streams_;
   const bool rtx_;
   const bool red_;
+  const bool report_perf_stats_;
   Call* sender_call_;
   VideoSendStream* send_stream_;
   test::PacketTransport* send_transport_;
 
  private:
   typedef std::map<uint32_t, uint32_t> SsrcMap;
   class VideoStreamFactory;
 
   Call::Config GetSenderCallConfig() override;
   void OnVideoStreamsCreated(
       VideoSendStream* send_stream,
       const std::vector<VideoReceiveStream*>& receive_streams) override;
   test::PacketTransport* CreateSendTransport(Call* sender_call) override;
   void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config) override;
   void ModifyAudioConfigs(
       AudioSendStream::Config* send_config,
       std::vector<AudioReceiveStream::Config>* receive_configs) override;
   void ModifyFlexfecConfigs(
       std::vector<FlexfecReceiveStream::Config>* receive_configs) override;
   void OnCallsCreated(Call* sender_call, Call* receiver_call) override;
 
   static void BitrateStatsPollingThread(void* obj);
 
   const int start_bitrate_bps_;
   const int64_t min_run_time_ms_;
-  const bool report_perf_stats_;
   int expected_bitrate_bps_;
   int64_t test_start_ms_;
   int64_t ramp_up_finished_ms_;
 
   const std::string extension_type_;
   std::vector<uint32_t> video_ssrcs_;
   std::vector<uint32_t> video_rtx_ssrcs_;
   std::vector<uint32_t> audio_ssrcs_;
 
   rtc::PlatformThread poller_thread_;
 };
 
 class RampUpDownUpTester : public RampUpTester {
  public:
   RampUpDownUpTester(size_t num_video_streams,
                      size_t num_audio_streams,
                      size_t num_flexfec_streams,
                      unsigned int start_bitrate_bps,
                      const std::string& extension_type,
                      bool rtx,
                      bool red,
-                     const std::vector<int>& loss_rates);
+                     const std::vector<int>& loss_rates,
+                     bool report_perf_stats);
   ~RampUpDownUpTester() override;
 
  protected:
   void PollStats() override;
 
  private:
   enum TestStates {
     kFirstRampup = 0,
     kLowRate,
     kSecondRampup,
@@ skipping 13 matching lines @@
   const std::vector<int> link_rates_;
   TestStates test_state_;
   TestStates next_state_;
   int64_t state_start_ms_;
   int64_t interval_start_ms_;
   int sent_bytes_;
   std::vector<int> loss_rates_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_CALL_RAMPUP_TESTS_H_