Make the BWE threshold adaptive.

webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc

 /*
  *  Copyright (c) 2013 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 236 matching lines @@
   }
   for (size_t i = 0; i < num_tcp_flows; ++i) {
     tcp_flow_ids.insert(next_flow_id);
     all_flow_ids.insert(next_flow_id);
     ++next_flow_id;
   }
 
   std::vector<VideoSource*> sources;
   std::vector<PacketSender*> senders;
 
-  size_t i = 0;
+  size_t i = 1;
   for (int media_flow : media_flow_ids) {
     // Streams started 20 seconds apart to give them different advantage when
     // competing for the bandwidth.
     const int64_t kFlowStartOffsetMs = i++ * (rand() % 10000);
     sources.push_back(new AdaptiveVideoSource(media_flow, 30, 300, 0,
                                               kFlowStartOffsetMs));
     senders.push_back(new PacedVideoSender(&uplink_, sources.back(), bwe_type));
   }
 
   const int64_t kTcpStartOffsetMs = 5000;
@@ skipping 52 matching lines @@
   for (PacketSender* sender : senders)
     delete sender;
   for (RateCounterFilter* rate_counter : rate_counters)
     delete rate_counter;
   for (PacketReceiver* receiver : receivers)
     delete receiver;
 }
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc