Implement RTCOutboundRtpStreamStats.totalPacketSendDelay for video.
This is a standardized metric. Spec:
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
It is meant to replace the legacy googBucketDelay. The average
packet delay over any interval can be calculated as the delta
totalPacketSendDelay divided by the delta packetsSent between two
calls to getStats().
Bug: webrtc:10506
Change-Id: I3d6c6d66e5a06937d0ea8d182a82cd255084ad19
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137044
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27979}
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index a751599..ebb2f38 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -793,12 +793,16 @@
verifier.TestMemberIsDefined(outbound_stream.frames_encoded);
verifier.TestMemberIsNonNegative<double>(
outbound_stream.total_encode_time);
+ verifier.TestMemberIsNonNegative<double>(
+ outbound_stream.total_packet_send_delay);
// The integration test is not set up to test screen share; don't require
// this to be present.
verifier.MarkMemberTested(outbound_stream.content_type, true);
} else {
verifier.TestMemberIsUndefined(outbound_stream.frames_encoded);
verifier.TestMemberIsUndefined(outbound_stream.total_encode_time);
+ // TODO(https://crbug.com/webrtc/10635): Implement for audio as well.
+ verifier.TestMemberIsUndefined(outbound_stream.total_packet_send_delay);
verifier.TestMemberIsUndefined(outbound_stream.content_type);
}
return verifier.ExpectAllMembersSuccessfullyTested();