Fix two UBSan warnings in NetEq

webrtc/modules/audio_coding/neteq/delay_manager.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.
  */
 
 #include "webrtc/modules/audio_coding/neteq/delay_manager.h"
 
 #include <assert.h>
 #include <math.h>
 
 #include <algorithm>  // max, min
 
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
 DelayManager::DelayManager(size_t max_packets_in_buffer,
                            DelayPeakDetector* peak_detector)
     : first_packet_received_(false),
@@ skipping 58 matching lines @@
   }
 
   // Try calculating packet length from current and previous timestamps.
   int packet_len_ms;
   if (!IsNewerTimestamp(timestamp, last_timestamp_) ||
       !IsNewerSequenceNumber(sequence_number, last_seq_no_)) {
     // Wrong timestamp or sequence order; use stored value.
     packet_len_ms = packet_len_ms_;
   } else {
     // Calculate timestamps per packet and derive packet length in ms.
-    int packet_len_samp =
+    int64_t packet_len_samp =
         static_cast<uint32_t>(timestamp - last_timestamp_) /
         static_cast<uint16_t>(sequence_number - last_seq_no_);
-    packet_len_ms = (1000 * packet_len_samp) / sample_rate_hz;
+    packet_len_ms =
+        rtc::checked_cast<int>(1000 * packet_len_samp / sample_rate_hz);
   }
 
   if (packet_len_ms > 0) {
     // Cannot update statistics unless |packet_len_ms| is valid.
     // Calculate inter-arrival time (IAT) in integer "packet times"
     // (rounding down). This is the value used as index to the histogram
     // vector |iat_vector_|.
     int iat_packets = packet_iat_count_ms_ / packet_len_ms;
 
     if (streaming_mode_) {
@@ skipping 308 matching lines @@
 int DelayManager::base_target_level() const { return base_target_level_; }
 void DelayManager::set_streaming_mode(bool value) { streaming_mode_ = value; }
 int DelayManager::last_pack_cng_or_dtmf() const {
   return last_pack_cng_or_dtmf_;
 }
 
 void DelayManager::set_last_pack_cng_or_dtmf(int value) {
   last_pack_cng_or_dtmf_ = value;
 }
 }  // namespace webrtc