Fix for bwe with overhead on audio only calls.

webrtc/modules/congestion_controller/transport_feedback_adapter.cc

 /*
  *  Copyright (c) 2015 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 25 matching lines @@
       return lhs.arrival_time_ms < rhs.arrival_time_ms;
     if (lhs.send_time_ms != rhs.send_time_ms)
       return lhs.send_time_ms < rhs.send_time_ms;
     return lhs.sequence_number < rhs.sequence_number;
   }
 };
 
 TransportFeedbackAdapter::TransportFeedbackAdapter(
     Clock* clock,
     BitrateController* bitrate_controller)
-    : send_time_history_(clock, kSendTimeHistoryWindowMs),
+    : transport_overhead_bytes_per_packet_(0),
+      send_time_history_(clock, kSendTimeHistoryWindowMs),
       clock_(clock),
       current_offset_ms_(kNoTimestamp),
       last_timestamp_us_(kNoTimestamp),
       bitrate_controller_(bitrate_controller) {}
 
 TransportFeedbackAdapter::~TransportFeedbackAdapter() {}
 
 void TransportFeedbackAdapter::InitBwe() {
   rtc::CritScope cs(&bwe_lock_);
   delay_based_bwe_.reset(new DelayBasedBwe(clock_));
@@ skipping 108 matching lines @@
   return last_packet_feedback_vector_;
 }
 
 void TransportFeedbackAdapter::OnRttUpdate(int64_t avg_rtt_ms,
                                            int64_t max_rtt_ms) {
   rtc::CritScope cs(&bwe_lock_);
   delay_based_bwe_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
 }
 
 }  // namespace webrtc