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_); |