Set the start bitrate to the delay-based BWE.

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 16 matching lines @@
 static const int kTransportSequenceNumberExtensionId = 8;
 static const unsigned int kSingleStreamTargetBps = 1000000;
 
 class Clock;
 
 class RampUpTester : public test::EndToEndTest {
  public:
   RampUpTester(size_t num_video_streams,
                size_t num_audio_streams,
                unsigned int start_bitrate_bps,
+               int64_t min_run_time_ms,
                const std::string& extension_type,
                bool rtx,
-               bool red);
+               bool red,
+               bool report_perf_stats);
   ~RampUpTester() override;
 
   size_t GetNumVideoStreams() const override;
   size_t GetNumAudioStreams() const override;
 
   void PerformTest() override;
 
  protected:
   virtual bool PollStats();
 
@@ skipping 34 matching lines @@
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config) override;
   void ModifyAudioConfigs(
       AudioSendStream::Config* send_config,
       std::vector<AudioReceiveStream::Config>* receive_configs) override;
   void OnCallsCreated(Call* sender_call, Call* receiver_call) override;
 
   static bool BitrateStatsPollingThread(void* obj);
 
   const int start_bitrate_bps_;
-  bool start_bitrate_verified_;
+  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_;
@@ skipping 22 matching lines @@
   int GetHighLinkCapacity() const;
   void EvolveTestState(int bitrate_bps, bool suspended);
 
   TestStates test_state_;
   int64_t state_start_ms_;
   int64_t interval_start_ms_;
   int sent_bytes_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_CALL_RAMPUP_TESTS_H_