Make UBSan warnings fatal and fix the existing ones

webrtc/modules/rtp_rtcp/source/rtp_sender.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 907 matching lines @@
   }
 
   return sent;
 }
 
 void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
   if (!send_side_delay_observer_ || capture_time_ms <= 0)
     return;
 
   uint32_t ssrc;
-  int avg_delay_ms = 0;
+  int64_t avg_delay_ms = 0;
   int max_delay_ms = 0;
   {
     rtc::CritScope lock(&send_critsect_);
     if (!ssrc_)
       return;
     ssrc = *ssrc_;
   }
   {
     rtc::CritScope cs(&statistics_crit_);
     // TODO(holmer): Compute this iteratively instead.
     send_delays_[now_ms] = now_ms - capture_time_ms;
     send_delays_.erase(send_delays_.begin(),
                        send_delays_.lower_bound(now_ms -
                        kSendSideDelayWindowMs));
     int num_delays = 0;
     for (auto it = send_delays_.upper_bound(now_ms - kSendSideDelayWindowMs);
          it != send_delays_.end(); ++it) {
       max_delay_ms = std::max(max_delay_ms, it->second);
       avg_delay_ms += it->second;
       ++num_delays;
     }
     if (num_delays == 0)
       return;
     avg_delay_ms = (avg_delay_ms + num_delays / 2) / num_delays;
   }
-  send_side_delay_observer_->SendSideDelayUpdated(avg_delay_ms, max_delay_ms,
-                                                  ssrc);
+  send_side_delay_observer_->SendSideDelayUpdated(
+      rtc::dchecked_cast<int>(avg_delay_ms), max_delay_ms, ssrc);
 }
 
 void RTPSender::UpdateOnSendPacket(int packet_id,
                                    int64_t capture_time_ms,
                                    uint32_t ssrc) {
   if (!send_packet_observer_ || capture_time_ms <= 0 || packet_id == -1)
     return;
 
   send_packet_observer_->OnSendPacket(packet_id, capture_time_ms, ssrc);
 }
@@ skipping 329 matching lines @@
     rtc::CritScope lock(&send_critsect_);
     packet->SetTimestamp(last_rtp_timestamp_);
     packet->set_capture_time_ms(capture_time_ms_);
   }
   AssignSequenceNumber(packet.get());
   SendToNetwork(std::move(packet), StorageType::kDontRetransmit,
                 RtpPacketSender::Priority::kLowPriority);
 }
 
 }  // namespace webrtc