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Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp.

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This collects this metric for both audio and video streams.
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp

This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/130479
which calculated this metric. This CL is purely plumbing from
"StreamDataCounters::last_packet_received_timestamp_ms" to
RTCInboundRtpStreamStats.


Bug: webrtc:10449
Change-Id: I757ad19b5b8e84553da5edd4a75efa3e1fe30b56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131397
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27628}
This commit is contained in:
Henrik Boström
2019-04-15 17:32:00 +02:00
committed by Commit Bot
parent 3d11e2f81c
commit 01738c63aa
12 changed files with 69 additions and 8 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
stats/rtcstats_objects.cc
View File

@ -578,6 +578,7 @@ WEBRTC_RTCSTATS_IMPL(
&packets_received,
&bytes_received,
&packets_lost,
&last_packet_received_timestamp,
&jitter,
&fraction_lost,
&round_trip_time,
@ -605,6 +606,7 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(std::string&& id,
packets_received("packetsReceived"),
bytes_received("bytesReceived"),
packets_lost("packetsLost"),
last_packet_received_timestamp("lastPacketReceivedTimestamp"),
jitter("jitter"),
fraction_lost("fractionLost"),
round_trip_time("roundTripTime"),
@ -627,6 +629,7 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
packets_received(other.packets_received),
bytes_received(other.bytes_received),
packets_lost(other.packets_lost),
last_packet_received_timestamp(other.last_packet_received_timestamp),
jitter(other.jitter),
fraction_lost(other.fraction_lost),
round_trip_time(other.round_trip_time),