Initialize VideoSendStream members in constructor.

webrtc/video/vie_receiver.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 37 matching lines @@
       fec_receiver_(FecReceiver::Create(this)),
       rtp_rtcp_(NULL),
       vcm_(module_vcm),
       remote_bitrate_estimator_(remote_bitrate_estimator),
       ntp_estimator_(new RemoteNtpTimeEstimator(clock_)),
       receiving_(false),
       restored_packet_in_use_(false),
       receiving_ast_enabled_(false),
       receiving_cvo_enabled_(false),
       receiving_tsn_enabled_(false),
-      last_packet_log_ms_(-1) {
-  assert(remote_bitrate_estimator);
-}
+      last_packet_log_ms_(-1) {}
 
 ViEReceiver::~ViEReceiver() {
   UpdateHistograms();
 }
 
 void ViEReceiver::UpdateHistograms() {
   FecPacketCounter counter = fec_receiver_->GetPacketCounter();
   if (counter.num_packets > 0) {
     RTC_HISTOGRAM_PERCENTAGE(
         "WebRTC.Video.ReceivedFecPacketsInPercent",
@@ skipping 168 matching lines @@
     return false;
   }
   header.payload_type_frequency = kVideoPayloadTypeFrequency;
   bool in_order = IsPacketInOrder(header);
   return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
 }
 
 bool ViEReceiver::DeliverRtp(const uint8_t* rtp_packet,
                              size_t rtp_packet_length,
                              const PacketTime& packet_time) {
+  RTC_DCHECK(remote_bitrate_estimator_);

[stefan-webrtc, 2016/02/11 12:28:06]

Why not keep it up there to fail as early as possible?

[pbos-webrtc, 2016/02/11 12:48:13]

nullptr on the sender side, which shouldn't call DeliverRtp.

   {
     rtc::CritScope lock(&receive_cs_);
     if (!receiving_) {
       return false;
     }
   }
 
   RTPHeader header;
   if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length,
                                  &header)) {
@@ skipping 200 matching lines @@
       rtp_receive_statistics_->GetStatistician(header.ssrc);
   if (!statistician)
     return false;
   // Check if this is a retransmission.
   int64_t min_rtt = 0;
   rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
   return !in_order &&
       statistician->IsRetransmitOfOldPacket(header, min_rtt);
 }
 }  // namespace webrtc