| 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 d16a11bc63f3edd3c4176943ed6dbe558a87068d..e9bceb72d948702fa8f5355379b96910c82e7396 100644
|
| --- a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
|
| +++ b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
|
| @@ -223,7 +223,8 @@ void StatisticsCalculator::GetNetworkStatistics(
|
| stats->preferred_buffer_size_ms = (delay_manager.TargetLevel() >> 8) *
|
| ms_per_packet;
|
| stats->jitter_peaks_found = delay_manager.PeakFound();
|
| - stats->clockdrift_ppm = delay_manager.AverageIAT();
|
| + stats->clockdrift_ppm =
|
| + rtc::saturated_cast<int32_t>(delay_manager.EstimatedClockDriftPpm());
|
|
|
| stats->packet_loss_rate =
|
| CalculateQ14Ratio(lost_timestamps_, timestamps_since_last_report_);
|
|
|