Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp
Partial implementation: currently only populated when a/v sync is enabled.
Bug: webrtc:7065
Change-Id: I8595cc848d080d7c3bef152462a9becf0e5a2196
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155621
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29581}
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index 0d51af0..9000ff9 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -821,6 +821,9 @@
verifier.TestMemberIsUndefined(inbound_stream.burst_discard_rate);
verifier.TestMemberIsUndefined(inbound_stream.gap_loss_rate);
verifier.TestMemberIsUndefined(inbound_stream.gap_discard_rate);
+ // Test runtime too short to get an estimate (at least two RTCP sender
+ // reports need to be received).
+ verifier.MarkMemberTested(inbound_stream.estimated_playout_timestamp, true);
if (inbound_stream.media_type.is_defined() &&
*inbound_stream.media_type == "video") {
verifier.TestMemberIsDefined(inbound_stream.frames_decoded);