De-flake VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly

webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc

 /*
  *  Copyright (c) 2012 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 49 matching lines @@
       time_last_decrease_ms_(0),
       first_report_time_ms_(-1),
       initially_lost_packets_(0),
       bitrate_at_2_seconds_kbps_(0),
       uma_update_state_(kNoUpdate),
       rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),
       event_log_(nullptr) {}
 
 SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
 
+void SendSideBandwidthEstimation::SetBitrates(int send_bitrate,
+                                              int min_bitrate,
+                                              int max_bitrate) {
+  if (send_bitrate > 0)
+    SetSendBitrate(send_bitrate);
+  SetMinMaxBitrate(min_bitrate, max_bitrate);
+}
+
 void SendSideBandwidthEstimation::SetSendBitrate(int bitrate) {
   RTC_DCHECK_GT(bitrate, 0);
   bitrate_ = bitrate;
 
   // Clear last sent bitrate history so the new value can be used directly
   // and not capped.
   min_bitrate_history_.clear();
 }
 
 void SendSideBandwidthEstimation::SetMinMaxBitrate(int min_bitrate,
@@ skipping 219 matching lines @@
     bitrate = min_bitrate_configured_;
   }
   return bitrate;
 }
 
 void SendSideBandwidthEstimation::SetEventLog(RtcEventLog* event_log) {
   event_log_ = event_log;
 }
 
 }  // namespace webrtc