Index: webrtc/modules/audio_coding/neteq/statistics_calculator.cc |
diff --git a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc |
index e9bceb72d948702fa8f5355379b96910c82e7396..b47ca127dbe7a2107b4315a7bd4eece1ef0d19fa 100644 |
--- a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc |
+++ b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc |
@@ -218,7 +218,7 @@ void StatisticsCalculator::GetNetworkStatistics( |
stats->added_zero_samples = added_zero_samples_; |
stats->current_buffer_size_ms = |
static_cast<uint16_t>(num_samples_in_buffers * 1000 / fs_hz); |
- const int ms_per_packet = rtc::checked_cast<int>( |
+ const int ms_per_packet = rtc::dchecked_cast<int>( |
decision_logic.packet_length_samples() / (fs_hz / 1000)); |
stats->preferred_buffer_size_ms = (delay_manager.TargetLevel() >> 8) * |
ms_per_packet; |