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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
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1
api/stats/rtcstats_objects.h
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@ -397,6 +397,7 @@ class RTC_EXPORT RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
RTCStatsMember<uint32_t> packets_received;
RTCStatsMember<uint64_t> bytes_received;
RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
RTCStatsMember<double> last_packet_received_timestamp;
// TODO(hbos): Collect and populate this value for both "audio" and "video",
// currently not collected for "video". https://bugs.webrtc.org/7065
RTCStatsMember<double> jitter;