| commit | 01738c63aa9480b8c4e3fcf3feb58ae9b9f94c6f | [log] [tgz] |
|---|---|---|
| author | Henrik Boström <hbos@webrtc.org> | Mon Apr 15 17:32:00 2019 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Apr 15 16:06:01 2019 +0000 |
| tree | d959eae63d2c1894aa15851508bf93e153cd50d8 | |
| parent | 3d11e2f81cccea4a03fb44817afc9724e8723a1e [diff] [blame] |
Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. 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}
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc index 6935769..9b1b6d0 100644 --- a/pc/rtc_stats_integrationtest.cc +++ b/pc/rtc_stats_integrationtest.cc
@@ -719,6 +719,7 @@ // packets_lost is defined as signed, but this should never happen in // this test. See RFC 3550. verifier.TestMemberIsNonNegative<int32_t>(inbound_stream.packets_lost); + verifier.TestMemberIsDefined(inbound_stream.last_packet_received_timestamp); if (inbound_stream.media_type.is_defined() && *inbound_stream.media_type == "video") { verifier.TestMemberIsUndefined(inbound_stream.jitter);