NetEq: Fixing a bug that caused rtc::checked_cast to trigger

webrtc/modules/audio_coding/neteq/neteq_impl.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 581 matching lines @@
     assert(decoder);  // Should always get a valid object, since we have
                       // already checked that the payload types are known.
     decoder->IncomingPacket(packet_list.front()->payload,
                             packet_list.front()->payload_length,
                             packet_list.front()->header.sequenceNumber,
                             packet_list.front()->header.timestamp,
                             receive_timestamp);
   }
 
   // Insert packets in buffer.
-  size_t temp_bufsize = packet_buffer_->NumPacketsInBuffer();
+  const size_t buffer_length_before_insert =
+      packet_buffer_->NumPacketsInBuffer();
   ret = packet_buffer_->InsertPacketList(
       &packet_list,
       *decoder_database_,
       &current_rtp_payload_type_,
       &current_cng_rtp_payload_type_);
   if (ret == PacketBuffer::kFlushed) {
     // Reset DSP timestamp etc. if packet buffer flushed.
     new_codec_ = true;
     update_sample_rate_and_channels = true;
   } else if (ret != PacketBuffer::kOK) {
@@ skipping 35 matching lines @@
       SetSampleRateAndChannels(decoder_info->fs_hz, decoder->Channels());
   }
 
   // TODO(hlundin): Move this code to DelayManager class.
   const DecoderDatabase::DecoderInfo* dec_info =
           decoder_database_->GetDecoderInfo(main_header.payloadType);
   assert(dec_info);  // Already checked that the payload type is known.
   delay_manager_->LastDecoderType(dec_info->codec_type);
   if (delay_manager_->last_pack_cng_or_dtmf() == 0) {
     // Calculate the total speech length carried in each packet.
-    temp_bufsize = packet_buffer_->NumPacketsInBuffer() - temp_bufsize;
-    temp_bufsize *= decoder_frame_length_;
+    const size_t buffer_length_after_insert =
+        packet_buffer_->NumPacketsInBuffer();
 
-    if ((temp_bufsize > 0) &&
-        (temp_bufsize != decision_logic_->packet_length_samples())) {
-      decision_logic_->set_packet_length_samples(temp_bufsize);
-      delay_manager_->SetPacketAudioLength(
-          static_cast<int>((1000 * temp_bufsize) / fs_hz_));
+    if (buffer_length_after_insert > buffer_length_before_insert) {
+      const size_t packet_length_samples =
+          (buffer_length_after_insert - buffer_length_before_insert) *
+          decoder_frame_length_;
+      if (packet_length_samples != decision_logic_->packet_length_samples()) {
+        decision_logic_->set_packet_length_samples(packet_length_samples);
+        delay_manager_->SetPacketAudioLength(
+            rtc::checked_cast<int>((1000 * packet_length_samples) / fs_hz_));
+      }
     }
 
     // Update statistics.
     if ((int32_t) (main_header.timestamp - timestamp_) >= 0 &&
         !new_codec_) {
       // Only update statistics if incoming packet is not older than last played
       // out packet, and if new codec flag is not set.
       delay_manager_->Update(main_header.sequenceNumber, main_header.timestamp,
                              fs_hz_);
     }
@@ skipping 1294 matching lines @@
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
                                               playout_mode_,
                                               decoder_database_.get(),
                                               *packet_buffer_.get(),
                                               delay_manager_.get(),
                                               buffer_level_filter_.get()));
 }
 }  // namespace webrtc