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2020-07-10Reland "Implement packets_(sent | received) for RTCTransportStats"Artem Titov
This is a reland of fb6f975401972635a644c0db06c135b4c0aaef4a. Related issue in chromium is fixed here: https://chromium-review.googlesource.com/c/chromium/src/+/2287294 Original change's description: > Implement packets_(sent | received) for RTCTransportStats > > Bug: webrtc:11756 > Change-Id: Ic0caad6d4675969ef3ae886f50326e4a2e1cbfe7 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178741 > Reviewed-by: Tommi <tommi@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31643} Bug: webrtc:11756 Change-Id: I1e310e3d23248500eb7dabd23d0ce6c4ec4cb8c6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178871 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31700}
2020-07-10Reland "Complete migration from "track" to "inbound-rtp" stats"Eldar Rello
This is a reland of 94fe0d3de5e8162d1a105fd1a3ec4bd2da97f43b with a fix. Original change's description: > Complete migration from "track" to "inbound-rtp" stats > > Bug: webrtc:11683 > Change-Id: I4c4a4fa0a7d6a20976922aca41d57540aa27fd1d > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178611 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Eldar Rello <elrello@microsoft.com> > Cr-Commit-Position: refs/heads/master@{#31683} Bug: webrtc:11683 Change-Id: I173b91625174051c02ff34127aaf6c086d3c5c66 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179060 Commit-Queue: Eldar Rello <elrello@microsoft.com> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31696}
2020-07-10Revert "Complete migration from "track" to "inbound-rtp" stats"Zeke Chin
This reverts commit 94fe0d3de5e8162d1a105fd1a3ec4bd2da97f43b. Reason for revert: Causes an assert in this line during a call: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/sdk/objc/api/peerconnection/RTCStatisticsReport.mm;drc=87a6e5ab4d8f0baf4e2a9b7752b43d825f9c0ce1;l=122?originalUrl=https:%2F%2Fcs.chromium.org%2F where frameWidth appears more than once Original change's description: > Complete migration from "track" to "inbound-rtp" stats > > Bug: webrtc:11683 > Change-Id: I4c4a4fa0a7d6a20976922aca41d57540aa27fd1d > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178611 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Eldar Rello <elrello@microsoft.com> > Cr-Commit-Position: refs/heads/master@{#31683} TBR=hbos@webrtc.org,hta@webrtc.org,elrello@microsoft.com Change-Id: I0ded36a40c8808dac5a777ed41815e52ab9a2573 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11683 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179020 Reviewed-by: Zeke Chin <tkchin@webrtc.org> Commit-Queue: Zeke Chin <tkchin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31692}
2020-07-10Split DataChannel into two separate classes for RTP and SCTP.Taylor Brandstetter
Done in preparation for some threading changes that would be quite messy if implemented with the class as-is. This results in some code duplication, but is preferable to one class having two completely different modes of operation. RTP data channels are in the process of being removed anyway, so the duplicated code won't last forever. Bug: webrtc:9883 Change-Id: Idfd41a669b56a4bb4819572e4a264a4ffaaba9c0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178940 Commit-Queue: Taylor <deadbeef@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31691}
2020-07-09Complete migration from "track" to "inbound-rtp" statsEldar Rello
Bug: webrtc:11683 Change-Id: I4c4a4fa0a7d6a20976922aca41d57540aa27fd1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178611 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Eldar Rello <elrello@microsoft.com> Cr-Commit-Position: refs/heads/master@{#31683}
2020-07-08[Stats] Add more rtc::Thread::ScopedDisallowBlockingCalls to getStats().Henrik Boström
This ensures with DCHECK-crashes that we don't accidentally do more blocking invokes than we think. Remaining blocking invokes FYI: - PrepareTransceiverStatsInfos_s_w() does 1 blocking invoke (regardless of the number of transceivers or channels) to the worker thread. This is because VoiceMediaChannel, VideoMediaChannel and GetParameters() execute on the worker thread, and the result of these operations are needed on the signalling thread. - pc_->GetCallStats() does 1 blocking invoke to the worker thread. These two blocking invokes can be merged, reducing the total number of blocking invokes from 2 to 1, but this CL does not attempt to do that. I filed https://crbug.com/webrtc/11767 for that. Bug: webrtc:11716 Change-Id: Iebc2ab350d253fd037211cdd283825b4e5b2d446 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178867 Reviewed-by: Evan Shrubsole <eshr@google.com> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31670}
2020-07-08[Stats] Optimization: Minimize number of thread-invokes in getStats().Henrik Boström
TrackMediaInfoMap was previously constructed on the signaling thread. It would iterate all the senders and receivers and perform GetParameters(), which internally would invoke on the worker thread. This resulted in as many thread-invokes as number of receivers. With this CL we piggyback on an existing thread-invoke, performing a single blocking invoke for all transceivers. This is good for performance when there is a lot of thread contention. The code is already exercised by unit tests and integration tests. rtc::Thread::ScopedDisallowBlockingCalls is added to DCHECK that we don't accidentally do any other blocking invokes. A couple of unnecessary DCHECKs had to be removed to avoid PROXY invokes back to the signaling thread (deadlock). These DCHECKs won't be missed. Bug: webrtc:11716 Change-Id: I139c7434682ff627bb88351b5752320dd322d9eb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178816 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31666}
2020-07-08Revert "Implement packets_(sent | received) for RTCTransportStats"Mirko Bonadei
This reverts commit fb6f975401972635a644c0db06c135b4c0aaef4a. Reason for revert: Looks like this breaks chromium.webrtc.fyi: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Mac%20Tester/6000 https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Win10%20Tester/6209 https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Win7%20Tester/6177 https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Win8%20Tester/6299 Original change's description: > Implement packets_(sent | received) for RTCTransportStats > > Bug: webrtc:11756 > Change-Id: Ic0caad6d4675969ef3ae886f50326e4a2e1cbfe7 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178741 > Reviewed-by: Tommi <tommi@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31643} TBR=hbos@webrtc.org,tommi@webrtc.org,titovartem@webrtc.org Change-Id: Icbb0974ba29cbddb614f1f37f8a2de1a7c56b571 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178868 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31665}
2020-07-07Implement packets_(sent | received) for RTCTransportStatsArtem Titov
Bug: webrtc:11756 Change-Id: Ic0caad6d4675969ef3ae886f50326e4a2e1cbfe7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178741 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31643}
2020-07-03Fix missing local and remote ids in RtpStreamStatsEldar Rello
Bug: chromium:1098266 Change-Id: I536464541c5971ea173bd7ed83d523fa50b43d0b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178486 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31620}
2020-06-16Add GetSctpStats to PeerConnectionInternal, remove sctp_data_channels()Tomas Gunnarsson
This removes code from DataChannelController that exposes an internal vector of data channels and puts the onus of returning stats for a data channel, on the data channel object itself. This will come in handy as we make threading changes to the data channel object. Change-Id: Ie164cc5823cd5f9782fc5c9a63aa4c76b8229639 Bug: webrtc:11547, webrtc:11687 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177244 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31533}
2020-06-10Remove enable_simulcast_stats config flag as not needed anymoreEldar Rello
Bug: webrtc:9547 Change-Id: Ie50453aa3496d16bfadfc9fdd3e7e6982278cfba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176841 Commit-Queue: Eldar Rello <elrello@microsoft.com> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31492}
2020-06-07Correct name of DC.dataChannelIdentifier stats memberHarald Alvestrand
Bug: webrtc:8787 Change-Id: Ie32b38f0671e89e94017f439de7614142328642f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176509 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31457}
2020-05-05Reland "Improve outbound-rtp statistics for simulcast"Henrik Boström
This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d. Reason for revert: The original CL is updated in PS #2 to fix the googRtt issue which was that when the legacy sender stats were put in "aggregated_senders" we forgot to update rtt_ms the same way that we do it for "senders". Original change's description: > Revert "Improve outbound-rtp statistics for simulcast" > > This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1. > > Reason for revert: Breaks googRtt in legacy getStats API > > Original change's description: > > Improve outbound-rtp statistics for simulcast > > > > Bug: webrtc:9547 > > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120 > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Eldar Rello <elrello@microsoft.com> > > Cr-Commit-Position: refs/heads/master@{#31097} > > TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:9547 > Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443 > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31165} TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com # Not skipping CQ checks because this is a reland. Bug: webrtc:9547 Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05Revert "Improve outbound-rtp statistics for simulcast"Henrik Boström
This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1. Reason for revert: Breaks googRtt in legacy getStats API Original change's description: > Improve outbound-rtp statistics for simulcast > > Bug: webrtc:9547 > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120 > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Eldar Rello <elrello@microsoft.com> > Cr-Commit-Position: refs/heads/master@{#31097} TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:9547 Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31165}
2020-04-17Improve outbound-rtp statistics for simulcastEldar Rello
Bug: webrtc:9547 Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Eldar Rello <elrello@microsoft.com> Cr-Commit-Position: refs/heads/master@{#31097}
2020-04-02Extend rtc::AdapterType with 2g, 3G, 4G & 5G enum values.Jonas Oreland
This patch adds new enum values for different types of cellular connections. The new costs are currently blocked when sending to remote, (so that arbitrary network switches does not starts occurring). The end-game for this series to be able to distinguish between different type of cellular connections in the ice-layer (e.g when selecting/switching connections). BUG: webrtc:11473 Change-Id: I587ac8fdff4f6cdd0f8905f327232f58818db4f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172582 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30970}
2020-03-11Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats APIArtem Titov
Bug: webrtc:11381 Change-Id: I7df3450e50da49d178e1e3a5d9f4970672d91aac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169120 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30758}
2020-02-13Populate sdp_fmtp_line and channels of RTCCodecStatsJohannes Kron
Change RtpCodecCapability::parameters and RtpCodecParameters::parameters to map from unordered_map to get welldefined FMTP lines. Bug: webrtc:7061 Change-Id: Ie61f76bbab915d72369e36e3f40ea11838827940 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168190 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30512}
2020-01-07Implement RTCOutboundRtpStreamStats::remoteId.Henrik Boström
This CL also removes RTCRtpStreamStats::associateStatsId, which is the legacy name for this stat, which was never implemented (existed in C++ but the member always had the value undefined and was thus never exposed in JavaScript). Bug: webrtc:11228 Change-Id: I28c332e4bdf2f55caaedf993482dca58b6b8b9a0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162800 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30171}
2019-11-25Add totalInterFrameDelay to RTCInboundRTPStreamStatsJohannes Kron
Bug: webrtc:11108 Change-Id: I0e0168ba303b127a8db3946d5fa5f97a1c90fb27 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160042 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29894}
2019-10-28Implement crypto stats on DTLS transportHarald Alvestrand
Bug: chromium:1018077 Change-Id: I585d4064f39e5f9d268b408ebf6ae13a056c778a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158403 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29628}
2019-10-23Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.Åsa Persson
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp Partial implementation: currently only populated when a/v sync is enabled. Bug: webrtc:7065 Change-Id: I8595cc848d080d7c3bef152462a9becf0e5a2196 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155621 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29581}
2019-10-15Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"Niels Möller
This is a reland of fbde32e596f06893d6dda13eb7d29f4c251cf08b The chromium problem should be fixed with https://chromium-review.googlesource.com/c/chromium/src/+/1862437 Original change's description: > Fix GetStats bytesSent/Received, wireup headerBytesSent/Received > > Changes the standard GetStats, legacy GetStats unchanged. > > Bug: webrtc:10525 > Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084 > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29462} Tbr: kwiberg@webrtc.org Bug: webrtc:10525 Change-Id: I3b61f9535aa3f1fca2ed84f068233803d4ec9fe2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157045 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29485}
2019-10-15Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"Mirko Bonadei
This reverts commit fbde32e596f06893d6dda13eb7d29f4c251cf08b. Reason for revert: It seems to break WebRTC FYI tests in Chromium. https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/4763 Original change's description: > Fix GetStats bytesSent/Received, wireup headerBytesSent/Received > > Changes the standard GetStats, legacy GetStats unchanged. > > Bug: webrtc:10525 > Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084 > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29462} TBR=kwiberg@webrtc.org,hbos@webrtc.org,nisse@webrtc.org,hta@webrtc.org Change-Id: I6a983ea4d5ff38e49f096a8ff5cd9b426768f955 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10525 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157043 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29478}
2019-10-14Fix GetStats bytesSent/Received, wireup headerBytesSent/ReceivedNiels Möller
Changes the standard GetStats, legacy GetStats unchanged. Bug: webrtc:10525 Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29462}
2019-09-17Use std::make_unique instead of absl::make_unique.Mirko Bonadei
WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-09Add qualityLimitationResolutionChanges statEvan Shrubsole
Implements the stat qualityLimitationResolutionChanges [1]. [1] https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges Bug: webrtc:10935 Change-Id: I391f2be5958a96b442e32c40ab7043752f3f71dd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150882 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@google.com> Cr-Commit-Position: refs/heads/master@{#29113}
2019-08-28Add support for RTCTransportStats.selectedCandidatePairChangesJonas Oreland
This patch adds accounting and reporting needed for newly added RTCTransportStats.selectedCandidatePairChanges, https://w3c.github.io/webrtc-stats/#dom-rtctransportstats-selectedcandidatepairchanges a) P2PTransportChannel counts everytime selected_connection_ is modified and reports this counter in the GetStats()-call. b) RTCStatsCollector puts the counter into the standardized stats object. Bug: webrtc:10900 Change-Id: Ibaeca18706b8edcbcb44b0c6f2754854bcb545ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149830 Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28987}
2019-08-27Fix frames dropped statisticsJohannes Kron
The |frames_dropped| statistics contain not only frames that are dropped but also frames that are in internal queues. This CL changes that so that |frames_dropped| only contains frames that are dropped. Bug: chromium:990317 Change-Id: If222568501b277a75bc514661c4f8f861b56aaed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150111 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28968}
2019-08-19Reland "[GetStats] Expose video codec implementation in standardized metrics."Henrik Boström
This is a reland of 2b9fa09fa3e3379fd8e76490c394f25670352ef2. It got reverted because I forgot to whitelist the new metrics in chromium, which has now been done: https://chromium-review.googlesource.com/c/chromium/src/+/1760209 Relanding requires no changes to the CL. Original change's description: > [GetStats] Expose video codec implementation in standardized metrics. > > Spec issue: https://github.com/w3c/webrtc-stats/issues/445 > Spec PR: https://github.com/w3c/webrtc-stats/pull/473 > > Now that the spec's RTCCodecStats.implementation has moved to > RTCOutboundRtpStreamStats.encoderImplementation and > RTCInboundRtpStreamStats.decoderImplementation, this CL implements them > using the same string that the legacy getStats() API used. > > Bug: webrtc:10890 > Change-Id: Ic43ce44735453626791959df3061ee253356015a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149168 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28877} TBR=ilnik@webrtc.org Bug: webrtc:10890 Change-Id: Ib874b608856c2795b1ca08f6af43c61dd859ea21 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149800 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28887}
2019-08-16Revert "[GetStats] Expose video codec implementation in standardized metrics."Henrik Andreassson
This reverts commit 2b9fa09fa3e3379fd8e76490c394f25670352ef2. Reason for revert: speculative revert since it seems to break Chrome FYI bots. See https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/4206 Original change's description: > [GetStats] Expose video codec implementation in standardized metrics. > > Spec issue: https://github.com/w3c/webrtc-stats/issues/445 > Spec PR: https://github.com/w3c/webrtc-stats/pull/473 > > Now that the spec's RTCCodecStats.implementation has moved to > RTCOutboundRtpStreamStats.encoderImplementation and > RTCInboundRtpStreamStats.decoderImplementation, this CL implements them > using the same string that the legacy getStats() API used. > > Bug: webrtc:10890 > Change-Id: Ic43ce44735453626791959df3061ee253356015a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149168 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28877} TBR=ilnik@webrtc.org,hbos@webrtc.org Change-Id: Ia0b7f9806564cf28881c50d6371b8141a22e3431 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10890 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149175 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28879}
2019-08-16[GetStats] Expose video codec implementation in standardized metrics.Henrik Boström
Spec issue: https://github.com/w3c/webrtc-stats/issues/445 Spec PR: https://github.com/w3c/webrtc-stats/pull/473 Now that the spec's RTCCodecStats.implementation has moved to RTCOutboundRtpStreamStats.encoderImplementation and RTCInboundRtpStreamStats.decoderImplementation, this CL implements them using the same string that the legacy getStats() API used. Bug: webrtc:10890 Change-Id: Ic43ce44735453626791959df3061ee253356015a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149168 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28877}
2019-07-31Make ID of datachannel stats not depend on dc.idHarald Alvestrand
The ID of stats was based on the datachannel's "id" attribute, but that could change - it was -1 before ID allocation, and a number afterwards. This CL changes the stats ID to depend on a monotonically increasing counter for allocated datachannels. Bug: webrtc:10842 Change-Id: I3e0c5dc07df8a7a502396de06bbedc9f676994a0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147642 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28720}
2019-07-08Format almost everything.Jonas Olsson
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-04[getStats] Implement "media-source" audio levels, fixing Chrome bug.Henrik Boström
Implements RTCAudioSourceStats members: - audioLevel - totalAudioEnergy - totalSamplesDuration In this CL description these are collectively referred to as the audio levels. The audio levels are removed from sending "track" stats (in Chrome, these are now reported as undefined instead of 0). Background: For sending tracks, audio levels were always reported as 0 in Chrome (https://crbug.com/736403), while audio levels were correctly reported for receiving tracks. This problem affected the standard getStats() but not the legacy getStats(), blocking some people from migrating. This was likely not a problem in native third_party/webrtc code because the delivery of audio frames from device to send-stream uses a different code path outside of chromium. A recent PR (https://github.com/w3c/webrtc-stats/pull/451) moved the send-side audio levels to the RTCAudioSourceStats, while keeping the receive-side audio levels on the "track" stats. This allows an implementation to report the audio levels even if samples are not sent onto the network (such as if an ICE connection has not been established yet), reflecting some of the current implementation. Changes: 1. Audio levels are added to RTCAudioSourceStats. Send-side audio "track" stats are left undefined. Receive-side audio "track" stats are not changed in this CL and continue to work. 2. Audio level computation is moved from the AudioState and AudioTransportImpl to the AudioSendStream. This is because a) the AudioTransportImpl::RecordedDataIsAvailable() code path is not exercised in chromium, and b) audio levels should, per-spec, not be calculated on a per-call basis, for which the AudioState is defined. 3. The audio level computation is now performed in AudioSendStream::SendAudioData(), a code path used by both native and chromium code. 4. Comments are added to document behavior of existing code, such as AudioLevel and AudioSendStream::SendAudioData(). Note: In this CL, just like before this CL, audio level is only calculated after an AudioSendStream has been created. This means that before an O/A negotiation, audio levels are unavailable. According to spec, if we have an audio source, we should have audio levels. An immediate solution to this would have been to calculate the audio level at pc/rtp_sender.cc. The problem is that the LocalAudioSinkAdapter::OnData() code path, while exercised in chromium, is not exercised in native code. The issue of calculating audio levels on a per-source bases rather than on a per-send stream basis is left to https://crbug.com/webrtc/10771, an existing "media-source" bug. This CL can be verified manually in Chrome at: https://codepen.io/anon/pen/vqRGyq Bug: chromium:736403, webrtc:10771 Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28480}
2019-07-02Add totalDecodeTime to RTCInboundRTPStreamStatsJohannes Kron
Pull request to WebRTC stats specification: https://github.com/w3c/webrtc-stats/pull/450 Bug: webrtc:10775 Change-Id: Id032cb324724329fee284ebc84595b9c39208ab8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144035 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28440}
2019-06-27Add RTCStats for keyFramesEncoded, keyFramesDecoded.Rasmus Brandt
This implements the correspondingly named JavaScript fields defined in https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict* and https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*. Bug: webrtc:7066 Change-Id: I431045bca80bf5faf27132c54f59c1f723c92952 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143683 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28404}
2019-06-26Delete RTCInboundRTPStreamStats::fraction_lostNiels Möller
And delete corresponding plumbing via the internal stats attribute MediaReceiverInfo::fraction_lost. The latter attribute is not deleted yet, since downstream projects have to be updated first. Bug: webrtc:10744 Change-Id: Id5401aeee7e5637a406ddf2fa33fbfe336abec9f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143178 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28385}
2019-06-24[getStats] Make remote-inbound-rtp.ssrc match outbound-rtp.ssrc.Henrik Boström
Per discussions at https://crbug.com/webrtc/10753, the remote-outbound-rtp.ssrc is supposed to reflect the SSRC of the RTP media stream (i.e. outbound-rtp.ssrc) and not the sender that the corresponding RTCP report block was transmitted on. Bug: webrtc:10753 Change-Id: Id88f5fdbe6397ba81a46f0ef430bd6f08e66b145 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143484 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28354}
2019-05-29Add jitterBufferDelay and jitterBufferEmittedCount stats for videoGuido Urdaneta
Bug: webrtc:10450 Change-Id: I6f586a3c6781450b9bfdcc31dc3f49f6289d70e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138265 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28096}
2019-05-28Implement QualityLimitationReasonTracker and expose "reason".Henrik Boström
This CL implements the logic behind qualityLimitationReason[1] and qualityLimitationDurations[2] This CL also exposes qualityLimitationReason in the standard getStats() API, but does not expose qualityLimitationDurations because that is blocked on supporting the "record<>" type in RTCStatsMember[3]. [1] https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationreason [2] https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationdurations [3] https://crbug.com/webrtc/10685 TBR=stefan@webrtc.org Bug: webrtc:10451, webrtc:10686 Change-Id: Ifff0be4ddd64eaec23d59c02af99fdbb1feb3841 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138825 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28090}
2019-05-27Implement RTCRemoteInboundRtpStreamStats for both audio and video.Henrik Boström
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}
2019-05-22Implement RTCMediaSourceStats and friends in standard getStats().Henrik Boström
This implements RTCAudioSourceStats and RTCVideoSourceStats, both inheriting from abstract dictionary RTCMediaSourceStats: https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats All members are implemented except for the total "frames" counter: - trackIdentifier - kind - width - height - framesPerSecond This means to make googFrameWidthInput, googFrameHeightInput and googFrameRateInput obsolete. Implemented using the same code path as the goog stats, there are some minor bugs that should be fixed in the future, but not this CL: 1. We create media-source objects on a per-track attachment basis. If the same track is attached multiple times this results in multiple media-source objects, but the spec says it should be on a per-source basis. 2. framesPerSecond is only calculated after connecting (when we have a sender with SSRC), but if collected on a per-source basis the source should be able to tell us the FPS whether or not we are sending it. Bug: webrtc:10453 Change-Id: I23705a79f15075dca2536275934af1904a7f0d39 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137804 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28028}
2019-05-22Implement RTCOutboundRtpStreamStats.totalEncodedBytesTarget.Henrik Boström
This is a standardized metric: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodedbytestarget We estimate the target frame size in bytes from the current encoder target bitrate and encoder framerate. We would expect that the average bytes produced by the encoder would over time match the average target, which is calculated by polling getStats() twice and dividing the delta totalEncodedBytesTarget with the delta framesEncoded. This is meant to make googTargetEncBitrate obsolete. Bug: webrtc:10446 Change-Id: Ib10ce236476a2f965582d5c536f419952926d4e6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137200 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28022}
2019-05-17Implement RTCOutboundRtpStreamStats.totalPacketSendDelay for video.Henrik Boström
This is a standardized metric. Spec: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay It is meant to replace the legacy googBucketDelay. The average packet delay over any interval can be calculated as the delta totalPacketSendDelay divided by the delta packetsSent between two calls to getStats(). Bug: webrtc:10506 Change-Id: I3d6c6d66e5a06937d0ea8d182a82cd255084ad19 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137044 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27979}
2019-05-03Expose new audio stats on the APIIvo Creusen
Several new audio stats were recently standardized and implemented in WebRTC in https://webrtc-review.googlesource.com/c/src/+/133887. This CL adds these to the GetStats API. Bug: webrtc:10442, webrtc:10443, webrtc:10444 Change-Id: I0e898ac14777e82b1a9099b5e0a5584eb9cb5934 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134213 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27839}
2019-04-29Reland "Piping audio interruption metrics to API layer"Henrik Lundin
The metrics are now added as RTCNonStandardStatsMember objects in RTCMediaStreamTrackStats. Unit tests are updated. This is a reland of https://webrtc-review.googlesource.com/c/src/+/134303, with fixes. TBR=kwiberg@webrtc.org Bug: webrtc:10549 Change-Id: I29dcc6fbfc69156715664e71acfa054c1b2d9038 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134500 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27806}
2019-04-17Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent.Henrik Boström
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent These are already existed in StreamDataCounters. This CL takes care of the plumbing of these values to the standard stats collector. TBR=solenberg@webrtc.org Bug: webrtc:10447 Change-Id: I27d6c3ee3ab627d306303e6ee67e586ddf31cc81 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132012 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27663}
2019-04-15Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp.Henrik Boström
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}
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-06 13:36:09 +0000