Index: webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc |
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc |
index 7770e65a269ec8e246d982e647f9ad3bc2d23f23..e0af3362b8416477a34125cafd399392365e861e 100644 |
--- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc |
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc |
@@ -94,6 +94,11 @@ void DebugDumpWriterImpl::DumpNetworkMetrics( |
if (metrics.rtt_ms) |
dump_metrics->set_rtt_ms(*metrics.rtt_ms); |
+ if (metrics.uplink_recoverable_packet_loss_fraction) { |
+ dump_metrics->set_uplink_recoverable_packet_loss_fraction( |
+ *metrics.uplink_recoverable_packet_loss_fraction); |
+ } |
+ |
DumpEventToFile(event, dump_file_.get()); |
#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP |
} |