f7b1b95f11
Add RTCRemoteOutboundRtpStreamStats for audio streams
...
Changes:
- adding the `RTCRemoteOutboundRtpStreamStats` dictionary (see [1])
- collection of remote outbound stats (only for audio streams)
- adding `remote_id` to the inbound stats and set with the ID of the
corresponding remote outbound stats only if the latter are available
- unit tests
[1] https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats
Tested: verified from chrome://webrtc-internals during an appr.tc call
Bug: webrtc:12529
Change-Id: Ide91dc04a3c387ba439618a9c6b64a95994a1940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211042
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33545}
2021-03-23 18:44:12 +00:00
a9ba450339
stats: add address as alias for ip
...
this was renamed in https://github.com/w3c/webrtc-pc/issues/1913 and https://github.com/w3c/webrtc-stats/pull/381
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatestats-address
BUG=chromium:968203
Change-Id: If75849fe1dc87ada6850e7b64aa8569e13baf0d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212681
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com >
Cr-Commit-Position: refs/heads/master@{#33534}
2021-03-23 06:29:10 +00:00
c366d51836
Fix unit for inbound RTP stat lastPacketReceivedTimestamp (s -> ms)
...
Both inbound RTP stats `estimatedPlayoutTimestamp` and
`lastPacketReceivedTimestamp` are surfaced to JS land as
`DOMHighResTimeStamp` - i.e., time values in milliseconds.
This CL fixes `lastPacketReceivedTimestamp` which is incorrectly
surfaced as time value in seconds.
Bug: webrtc:12605
Change-Id: I290103071cca3331d2a3066b6b6b9fcb4f4fd0af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212742
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33530}
2021-03-22 18:47:33 +00:00
049e6113a8
Add missing EXPECT_CALL for RTCStatsCollectorTest tests
...
Bug: webrtc:12529
Change-Id: I89e90f48ebcf5e4085c6a8403d733f416f27e6a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212441
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33502}
2021-03-18 12:52:42 +00:00
668dbf66ce
[Stats] Populate "frames" stats for video source.
...
Spec: https://www.w3.org/TR/webrtc-stats/#dom-rtcvideosourcestats-frames
Wiring up the "frames" stats with the cumulative fps counter on the video source.
Tests:
./out/Default/peerconnection_unittests
./out/Default/video_engine_tests
Bug: webrtc:12499
Change-Id: I4103f56ed04cb464f5f7e70fbf2d77c25a867a68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208782
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33404}
2021-03-09 08:54:38 +00:00
88a51b2902
Populate "total_round_trip_time" and "round_trip_time_measurements" for remote inbound RTP streams
...
Spec: https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict *
Adding them into the stats definition as well.
Bug: webrtc:12507
Change-Id: Id467a33fe7bb256655b68819e3ce87ca9af5b25f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209000
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33363}
2021-03-01 20:49:22 +00:00
86f04ad135
Populate “fractionLost” stats for remote inbound rtp streams
...
Tests:
./out/Default/peerconnection_unittests
Manually tested with Chromium to see the data populated
Spec: https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict *
Bug: webrtc:12506
Change-Id: I60ef8061fb31deab06ca5f115246ceb5a8cdc5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208960
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@{#33361}
2021-03-01 16:48:37 +00:00
8af6b4928a
Populate jitter stats for video RTP streams
...
Trying to take my first stab at contributing to WebRTC and I chose to populate jitter stats for video RTP streams. Please yell at me if this isn't something I'm not supposed to pick up. Appreciate a review, thanks!
Bug: webrtc:12487
Change-Id: Ifda985e9e20b1d87e4a7268f34ef2e45b1cbefa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208360
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33325}
2021-02-23 15:10:02 +00:00
95157a054b
stats: add transportId to codec stats
...
BUG=webrtc:12181
Change-Id: Ib8e38f19ef2ddcb98455356087781f146af8c6b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193280
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@{#32618}
2020-11-17 12:34:39 +00:00
aa83cc7fda
getstats: fix inbound-rtp audio level range
...
converting to the [0..1] range as done in other
places.
BUG=chromium:1142626
Change-Id: I190b23f54a29505b526a4fdfb733b841b823ff29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190441
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#32502}
2020-10-27 10:10:33 +00:00
4e8c115960
Reland "introduce an unsupported content description type"
...
This is a reland of 239f92ecf7fc8ca27e0376dd192b33ce33377b3c
Original change's description:
> introduce an unsupported content description type
>
> This carries around unsupported content descriptions
> (i.e. things where webrtc does not understand the media type
> or protocol) in a special data type so that a rejected content or
> mediasection is added to the answer SDP.
>
> BUG=webrtc:3513
>
> Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
> Reviewed-by: Harald Alvestrand <hta@webrtc.org >
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com >
> Cr-Commit-Position: refs/heads/master@{#32410}
Bug: webrtc:3513
Change-Id: I48e338100f829f1df5b8165217c89b5ef860fe79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188820
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#32457}
2020-10-21 08:20:05 +00:00
ad2ec76387
Revert "introduce an unsupported content description type"
...
This reverts commit 239f92ecf7fc8ca27e0376dd192b33ce33377b3c.
Reason for revert: Breaks downstream projects.
Original change's description:
> introduce an unsupported content description type
>
> This carries around unsupported content descriptions
> (i.e. things where webrtc does not understand the media type
> or protocol) in a special data type so that a rejected content or
> mediasection is added to the answer SDP.
>
> BUG=webrtc:3513
>
> Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
> Reviewed-by: Harald Alvestrand <hta@webrtc.org >
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com >
> Cr-Commit-Position: refs/heads/master@{#32410}
TBR=kthelgason@webrtc.org ,hta@webrtc.org ,philipp.hancke@googlemail.com
Change-Id: I055fe001fe2757d79be7c304eccc43a8e3104f69
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:3513
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188581
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#32411}
2020-10-15 10:03:13 +00:00
239f92ecf7
introduce an unsupported content description type
...
This carries around unsupported content descriptions
(i.e. things where webrtc does not understand the media type
or protocol) in a special data type so that a rejected content or
mediasection is added to the answer SDP.
BUG=webrtc:3513
Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
Reviewed-by: Kári Helgason <kthelgason@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com >
Cr-Commit-Position: refs/heads/master@{#32410}
2020-10-15 09:28:28 +00:00
9cb42c8690
Move pc/media_stream_track.h to the api/ directory
...
This file is being accessed from Chrome. Moving it lessens the
dependency of Chrome on files in the pc/ directory, and allows
easier refactoring of pc/.
Bug: webrtc:11967
Change-Id: Iccd568f84e9cf4086e37c58db1b4cba6c376f413
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187489
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#32378}
2020-10-12 07:28:01 +00:00
edacbd53de
Reland "Implement packets_(sent | received) for RTCTransportStats"
...
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}
2020-07-10 11:50:59 +00:00
4e5bc9f081
Reland "Complete migration from "track" to "inbound-rtp" stats"
...
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}
2020-07-10 10:17:50 +00:00
e6f3897945
Revert "Complete migration from "track" to "inbound-rtp" stats"
...
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}
2020-07-10 00:06:53 +00:00
3a034e15b4
Split DataChannel into two separate classes for RTP and SCTP.
...
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}
2020-07-10 00:03:21 +00:00
94fe0d3de5
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}
2020-07-09 10:02:26 +00:00
9b35da880b
Revert "Implement packets_(sent | received) for RTCTransportStats"
...
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}
2020-07-08 09:42:41 +00:00
6fcd0f8031
Migrate pc/ to webrtc::Mutex.
...
Bug: webrtc:11567
Change-Id: I1adc22d2998966958750138e66108cf39a8c3d57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178840
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Markus Handell <handellm@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31654}
2020-07-07 18:25:09 +00:00
fb6f975401
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}
2020-07-07 10:45:05 +00:00
c07e904a25
Fix missing local and remote ids in RtpStreamStats
...
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}
2020-07-03 09:27:52 +00:00
7d3cfbf90d
Inject signaling and network threads to DataChannel.
...
Add a few DCHECKs and comments about upcoming work.
Bug: webrtc:11547
Change-Id: I2d42f48cb93f31e70cf9fe4b3b62241c38bc9d8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177106
Reviewed-by: Taylor <deadbeef@webrtc.org >
Commit-Queue: Tommi <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31530}
2020-06-16 10:22:19 +00:00
9276e2c39b
Remove enable_simulcast_stats config flag as not needed anymore
...
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}
2020-06-10 15:59:32 +00:00
10ef847289
Correct name of DC.dataChannelIdentifier stats member
...
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}
2020-06-07 21:57:50 +00:00
6efc14b33d
VideoTrackSourceInterface: make some newly introduced methods pure virtual.
...
Bug: webrtc:11114
Change-Id: Ic4d3835ae84b6a652c49f30a9c275870bbf3dacf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174440
Commit-Queue: Markus Handell <handellm@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31211}
2020-05-11 12:28:32 +00:00
a0ff50c031
Reland "Improve outbound-rtp statistics for simulcast"
...
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}
2020-05-05 20:22:19 +00:00
9a925c9ce3
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}
2020-05-05 13:38:51 +00:00
da6cda839d
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}
2020-04-17 11:28:00 +00:00
e618cc9c1e
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
...
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}
2020-03-11 12:08:32 +00:00
72d6915d5f
Populate sdp_fmtp_line and channels of RTCCodecStats
...
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}
2020-02-13 10:10:37 +00:00
0c626afcf3
Use newer version of TimeDelta and TimeStamp factories in webrtc
...
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format
Bug: None
Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 12:21:17 +00:00
910cdc2a08
Add a round-trip test that stats.toJson output is parseable
...
Bug: webrtc:10173
Change-Id: Icf22901824fc85cc390e9450c480d3b7a728dc34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165385
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30194}
2020-01-09 13:05:04 +00:00
4f40fa5cef
Implement RTCOutboundRtpStreamStats::remoteId.
...
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}
2020-01-07 17:26:01 +00:00
00376e190a
Add totalInterFrameDelay to RTCInboundRTPStreamStats
...
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}
2019-11-25 10:50:37 +00:00
5cb7807a36
Implement crypto stats on DTLS transport
...
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}
2019-10-28 11:30:23 +00:00
fcf79cca7b
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
...
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}
2019-10-23 07:46:39 +00:00
ac0a4cbbd8
Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
...
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}
2019-10-15 10:43:59 +00:00
ef0627fb50
Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
...
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}
2019-10-15 08:55:06 +00:00
fbde32e596
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}
2019-10-14 13:07:13 +00:00
1b575417b3
Always pass arguments to INSTANTIATE_TEST_SUITE_P.
...
Passing an empty arg is working at the moment but it is not
guaranteed to continue to work in the future.
This CL has been generated with:
git grep -l "INSTANTIATE_TEST_SUITE_P(," | xargs sed -i \
"s/INSTANTIATE_TEST_SUITE_P(,/INSTANTIATE_TEST_SUITE_P(All,/g"
Bug: None
Change-Id: Icd2fb9d9d29aed5d692a234124bd990d0f097db4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153890
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29282}
2019-09-24 08:56:24 +00:00
cc62b16658
Add qualityLimitationResolutionChanges stat
...
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}
2019-09-09 15:22:57 +00:00
149dc72dfa
Add support for RTCTransportStats.selectedCandidatePairChanges
...
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}
2019-08-28 13:22:08 +00:00
0c141c591a
Fix frames dropped statistics
...
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}
2019-08-27 07:43:01 +00:00
6b430867b8
Reland "[GetStats] Expose video codec implementation in standardized metrics."
...
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}
2019-08-19 09:09:18 +00:00
df625f46c0
Revert "[GetStats] Expose video codec implementation in standardized metrics."
...
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 15:29:28 +00:00
2b9fa09fa3
[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}
2019-08-16 14:10:46 +00:00
928e7a3e79
Make ID of datachannel stats not depend on dc.id
...
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}
2019-07-31 13:19:08 +00:00
a4d873786f
Format almost everything.
...
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}
2019-07-08 13:45:15 +00:00
