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Generate signed packets_lost in WebRTC-stats

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Bug: webrtc:8626
Change-Id: Ibeca29c5bb01e57c87fbf6a3c8589eb4e03089d5
Reviewed-on: https://webrtc-review.googlesource.com/32660
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21241}
This commit is contained in:
Harald Alvestrand
2017-12-13 12:26:04 +01:00
committed by Commit Bot
parent 7ca9ae2e26
commit 719487ec7a
4 changed files with 7 additions and 5 deletions
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2
api/stats/rtcstats_objects.h
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@ -361,7 +361,7 @@ class RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
RTCStatsMember<uint32_t> packets_received;
RTCStatsMember<uint64_t> bytes_received;
RTCStatsMember<uint32_t> packets_lost;
RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
// 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;