Make the BWE threshold adaptive.

webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.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 662 matching lines @@
       ++prototype_header_.sequenceNumber;
       uint32_t size = NextPacketSize(frame_size, payload_size);
       MediaPacket* new_packet =
           new MediaPacket(flow_id_, send_time_us, size, prototype_header_);
       new_packets.push_back(new_packet);
       new_packet->SetAbsSendTimeMs(next_frame_ms_);
       new_packet->set_sender_timestamp_us(send_time_us);
       payload_size -= size;
     }
 
-    next_frame_ms_ += frame_period_ms_;
+    next_frame_ms_ += frame_period_ms_;  // + ((rand() % 3) - 2);
   }
   in_out->merge(new_packets, DereferencingComparator<Packet>);
 }
 
 AdaptiveVideoSource::AdaptiveVideoSource(int flow_id,
                                          float fps,
                                          uint32_t kbps,
                                          uint32_t ssrc,
                                          int64_t first_frame_offset_ms)
     : VideoSource(flow_id, fps, kbps, ssrc, first_frame_offset_ms) {
@@ skipping 47 matching lines @@
 uint32_t PeriodicKeyFrameSource::NextPacketSize(uint32_t frame_size,
                                                 uint32_t remaining_payload) {
   uint32_t fragments =
       (frame_size + (kMaxPayloadSizeBytes - 1)) / kMaxPayloadSizeBytes;
   uint32_t avg_size = (frame_size + fragments - 1) / fragments;
   return std::min(avg_size, remaining_payload);
 }
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc