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Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

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The value is being moved:
https://github.com/w3c/webrtc-stats/pull/167

Stop collecting this value. Our previous value was incorrect, our RTT
value was a smoothed value based on STUN pings but the spec says it
should be based on RTCP timestamps in RTCP Receiver Report (RR) on
inbound streams with isRemote=true (not supported).

Updated some bug references.

BUG=webrtc:7065, webrtc:7066

Review-Url: https://codereview.webrtc.org/2722633005
Cr-Commit-Position: refs/heads/master@{#16931}
This commit is contained in:
hbos
2017-03-01 01:02:45 -08:00
committed by Commit bot
parent deaf6fb071
commit a7a9be159d
5 changed files with 24 additions and 43 deletions
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3
webrtc/pc/rtcstats_integrationtest.cc
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@ -553,6 +553,7 @@ class RTCStatsReportVerifier {
verifier.TestMemberIsNonNegative<double>(inbound_stream.jitter);
}
verifier.TestMemberIsNonNegative<double>(inbound_stream.fraction_lost);
verifier.TestMemberIsUndefined(inbound_stream.round_trip_time);
verifier.TestMemberIsUndefined(inbound_stream.packets_discarded);
verifier.TestMemberIsUndefined(inbound_stream.packets_repaired);
verifier.TestMemberIsUndefined(inbound_stream.burst_packets_lost);
@ -585,8 +586,6 @@ class RTCStatsReportVerifier {
verifier.TestMemberIsNonNegative<uint32_t>(outbound_stream.packets_sent);
verifier.TestMemberIsNonNegative<uint64_t>(outbound_stream.bytes_sent);
verifier.TestMemberIsUndefined(outbound_stream.target_bitrate);
// TODO(hbos): Defined in video but not audio case. Why? crbug.com/669877
verifier.MarkMemberTested(outbound_stream.round_trip_time, true);
if (outbound_stream.media_type.is_defined() &&
*outbound_stream.media_type == "video") {
verifier.TestMemberIsDefined(outbound_stream.frames_encoded);