Fix probing breakage with send-side BWE introduced by r11322.

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 163 matching lines @@
              now_ms - first_report_time_ms_ >= kBweConverganceTimeMs) {
     uma_update_state_ = kDone;
     int bitrate_diff_kbps =
         std::max(bitrate_at_2_seconds_kbps_ - bitrate_kbps, 0);
     RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitialVsConvergedDiff",
                                 bitrate_diff_kbps, 0, 2000, 50);
   }
 }
 
 void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) {
-  // We trust the REMB during the first 2 seconds if we haven't had any
-  // packet loss reported, to allow startup bitrate probing.
-  if (last_fraction_loss_ == 0 && IsInStartPhase(now_ms) &&
-      bwe_incoming_ > bitrate_) {
-    bitrate_ = CapBitrateToThresholds(now_ms, bwe_incoming_);
-    min_bitrate_history_.clear();
-    min_bitrate_history_.push_back(std::make_pair(now_ms, bitrate_));
-    return;
+  // We trust the REMB and/or delay-based estimate during the first 2 seconds if
+  // we haven't had any packet loss reported, to allow startup bitrate probing.
+  if (last_fraction_loss_ == 0 && IsInStartPhase(now_ms)) {
+    uint32_t prev_bitrate = bitrate_;
+    if (bwe_incoming_ > bitrate_)
+      bitrate_ = CapBitrateToThresholds(now_ms, bwe_incoming_);
+    if (delay_based_bitrate_bps_ > bitrate_)
+      bitrate_ = CapBitrateToThresholds(now_ms, delay_based_bitrate_bps_);
+    if (bitrate_ != prev_bitrate) {
+      min_bitrate_history_.clear();
+      min_bitrate_history_.push_back(std::make_pair(now_ms, bitrate_));
+      return;
+    }
   }
   UpdateMinHistory(now_ms);
   // Only start updating bitrate when receiving receiver blocks.
   // TODO(pbos): Handle the case when no receiver report is received for a very
   // long time.
   if (time_last_receiver_block_ms_ != -1) {
     if (last_fraction_loss_ <= 5) {
       // Loss < 2%: Increase rate by 8% of the min bitrate in the last
       // kBweIncreaseIntervalMs.
       // Note that by remembering the bitrate over the last second one can
@@ skipping 91 matching lines @@
     bitrate = min_bitrate_configured_;
   }
   return bitrate;
 }
 
 void SendSideBandwidthEstimation::SetEventLog(RtcEventLog* event_log) {
   event_log_ = event_log;
 }
 
 }  // namespace webrtc