Implement RTCRemoteInboundRtpStreamStats for both audio and video.
This implements the essentials of RTCRemoteInboundRtpStreamStats. This
includes:
- ssrc
- transportId
- codecId
- packetsLost
- jitter
- localId
- roundTripTime
https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
The following members are not implemented because they require more
work...
- From RTCReceivedRtpStreamStats: packetsReceived, packetsDiscarded,
packetsRepaired, burstPacketsLost, burstPacketsDiscarded,
burstLossCount, burstDiscardCount, burstLossRate, burstDiscardRate,
gapLossRate and gapDiscardRate.
- From RTCRemoteInboundRtpStreamStats: fractionLost.
Bug: webrtc:10455, webrtc:10456
Change-Id: If2ab0da7105d8c93bba58e14aa93bd22ffe57f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138067
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28073}
diff --git a/pc/rtc_stats_collector.h b/pc/rtc_stats_collector.h
index 82501db..cd5ec21 100644
--- a/pc/rtc_stats_collector.h
+++ b/pc/rtc_stats_collector.h
@@ -190,6 +190,8 @@
void ProducePeerConnectionStats_s(int64_t timestamp_us,
RTCStatsReport* report) const;
// Produces |RTCInboundRTPStreamStats| and |RTCOutboundRTPStreamStats|.
+ // This has to be invoked after codecs and transport stats have been created
+ // because some metrics are calculated through lookup of other metrics.
void ProduceRTPStreamStats_n(
int64_t timestamp_us,
const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,