79326eaca7
Implement missing candidate pair packets/bytes sent/received stats.
...
Specifically:
* packetsSent
* packetsReceived
* packetsDiscardedOnSend
* bytesDiscardedOnSend
Bug: webrtc:10569
Change-Id: Id92c20b93dea57637239a6321bd8aa644867f272
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232961
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org >
Reviewed-by: Jonas Oreland <jonaso@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35113}
2021-09-28 23:27:05 +00:00
2562cf0105
Reland "Wire up non-sender RTT for audio, and implement related standardized stats."
...
This reverts commit 2c41cbae37cac548a1133589b9d2c2e8614fa6cb.
Reason for revert: The breaking test in Chromium has been temporarily disabled in https://chromium-review.googlesource.com/c/chromium/src/+/3139794/2 .
Original change's description:
> Revert "Wire up non-sender RTT for audio, and implement related standardized stats."
>
> This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e.
>
> Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.
>
> Original change's description:
> > Wire up non-sender RTT for audio, and implement related standardized stats.
> >
> > The implemented stats are:
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
> >
> > Bug: webrtc:12951, webrtc:12714
> > Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> > Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
> > Reviewed-by: Henrik Boström <hbos@webrtc.org >
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org >
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
> > Cr-Commit-Position: refs/heads/main@{#34861}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=hta,hbos,minyue
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
> Reviewed-by: Olga Sharonova <olka@webrtc.org >
> Commit-Queue: Björn Terelius <terelius@webrtc.org >
> Cr-Commit-Position: refs/heads/main@{#34897}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12951, webrtc:12714
Change-Id: I786b06933d85bdffc5e879bf52436bb3469b7f3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231181
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#34930}
2021-09-06 14:26:55 +00:00
2c41cbae37
Revert "Wire up non-sender RTT for audio, and implement related standardized stats."
...
This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e.
Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.
Original change's description:
> Wire up non-sender RTT for audio, and implement related standardized stats.
>
> The implemented stats are:
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
> Reviewed-by: Henrik Boström <hbos@webrtc.org >
> Reviewed-by: Harald Alvestrand <hta@webrtc.org >
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
> Cr-Commit-Position: refs/heads/main@{#34861}
# Not skipping CQ checks because original CL landed > 1 day ago.
TBR=hta,hbos,minyue
Bug: webrtc:12951, webrtc:12714
Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Reviewed-by: Olga Sharonova <olka@webrtc.org >
Commit-Queue: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#34897}
2021-09-01 17:32:00 +00:00
fb0dca6c05
Wire up non-sender RTT for audio, and implement related standardized stats.
...
The implemented stats are:
- https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
- https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
- https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
Bug: webrtc:12951, webrtc:12714
Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#34861}
2021-08-30 09:03:50 +00:00
cfea2182f8
Use backticks not vertical bars to denote variables in comments
...
Bug: webrtc:12338
Change-Id: I89c8b3a328d04203177522cbdfd9e606fd4bce4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228246
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34696}
2021-08-10 10:40:03 +00:00
0e61fdd27c
Use backticks not vertical bars to denote variables in comments for /api
...
Bug: webrtc:12338
Change-Id: Ib97b2c3d64dbd895f261ffa76a2e885bd934a87f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226940
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34554}
2021-07-26 18:27:34 +00:00
28a2c63526
Adding packetsDiscarded to RTCReceivedRtpStreamStats.
...
Bug: webrtc:12532, webrtc:7065, webrtc:8199
Change-Id: I3ba62ec65e5660e98787f629aec3ee7a0889207a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225261
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Commit-Queue: Minyue Li <minyue@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34468}
2021-07-13 20:34:45 +00:00
e91c992fa1
Implement nack_count metric for outbound audio rtp streams.
...
Bug: webrtc:12510
Change-Id: Ia035885bced3c3d202bb9ffeb88c2556d4830e92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225021
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34444}
2021-07-09 13:29:10 +00:00
e54914a79e
Implement nack_count metric for inbound audio rtp streams.
...
Bug: webrtc:12925
Change-Id: I4542ca0f14a7dd7485ad5a2b6f2bd7051076f71f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224085
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34401}
2021-07-01 10:38:44 +00:00
64851c0bfb
Reland: Fix echo return loss stats and add to RTCAudioSourceStats.
...
Relanding after adding to chromium stats whitelist:
https://chromium-review.googlesource.com/c/chromium/src/+/2983329
This solves two problems:
* Echo return loss stats weren't being gathered in Chrome, because they
need to be taken from the audio processor attached to the track
rather than the audio send stream.
* The standardized location is in RTCAudioSourceStats, not
RTCMediaStreamTrackStats. For now, will populate the stats in both
locations.
Bug: webrtc:12770
Change-Id: I3633ee428d07b283b0cc503a84d8fa2e79415dfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223761
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34367}
2021-06-25 21:08:20 +00:00
fe6580fb87
Revert "Fix echo return loss stats and add to RTCAudioSourceStats."
...
This reverts commit a27cfbffdfa0bf359628d2164db5b9d6321f9c9c.
Reason for revert: WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsGetStatsPromise failing.
Original change's description:
> Fix echo return loss stats and add to RTCAudioSourceStats.
>
> This solves two problems:
> * Echo return loss stats weren't being gathered in Chrome, because they
> need to be taken from the audio processor attached to the track
> rather than the audio send stream.
> * The standardized location is in RTCAudioSourceStats, not
> RTCMediaStreamTrackStats. For now, will populate the stats in both
> locations.
>
> Bug: webrtc:12770
> Change-Id: I47eaf7f2b50b914a1be84156aa831e27497d07e3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223182
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org >
> Reviewed-by: Henrik Boström <hbos@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#34344}
TBR=deadbeef@webrtc.org ,hbos@webrtc.org ,hbos@chromium.org ,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I6b2587d762f005adef67c0d5121f1b58c3b76688
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12770
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223068
Reviewed-by: Evan Shrubsole <eshr@google.com >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Evan Shrubsole <eshr@google.com >
Cr-Commit-Position: refs/heads/master@{#34352}
2021-06-22 08:10:50 +00:00
a27cfbffdf
Fix echo return loss stats and add to RTCAudioSourceStats.
...
This solves two problems:
* Echo return loss stats weren't being gathered in Chrome, because they
need to be taken from the audio processor attached to the track
rather than the audio send stream.
* The standardized location is in RTCAudioSourceStats, not
RTCMediaStreamTrackStats. For now, will populate the stats in both
locations.
Bug: webrtc:12770
Change-Id: I47eaf7f2b50b914a1be84156aa831e27497d07e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223182
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34344}
2021-06-21 21:18:02 +00:00
7d23535108
Populate qualityLimitationDurations stats for outbound RTP streams
...
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationdurations
Tested in chromium using wpt/webrtc-stats.
Bug: webrtc:10686
Change-Id: I05ac344e6caa7a663675de4c06ccfd17e1efb6ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219300
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34179}
2021-05-31 21:39:37 +00:00
ef036cdff2
[Stats] Cleanup obsolete stats - isRemote & deleted
...
Deleting obsolete stats. Spec: https://www.w3.org/TR/webrtc-stats/
1. RTCInbound/OutboundRtpStats.isRemote: No longer useful with remote stream stats
2. RTCIceCandidateStats.deleted: This field was obsoleted because if the ICE candidate is deleted it no longer appears in getStats()
I also marked as many other obsoleted stats possible according to spec. I am not as confident to delete them but feel free to comment to let me know if anything is off / can be deleted.
Bug: webrtc:12583
Change-Id: I688d0076270f85caa86256349753e5f0e0a44931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211781
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33549}
2021-03-24 10:49:34 +00:00
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
fd1e9d1af4
[Stats] Add minimum RTCReceivedRtpStreamStats with jitter and packetsLost
...
Spec: https://www.w3.org/TR/webrtc-stats/#receivedrtpstats-dict *
According to the spec, |RTCReceivedRtpStreamStats| is the base class for |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats|. This structure isn't visible in JavaScript but it's important to bring it up to spec for the C++ part. This CL adds the barebone |RTCReceivedRtpStreamStats| with a bunch of TODOs for later migrations.
This commit makes the minimum |RTCReceivedRtpStreamStats| and rebase |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats| to use the new class as the parent class.
This commit also moves |jitter| and |packets_lost| to |RTCReceivedRtpStreamStats|, from |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats|. Moving these two first because they are the two that exist in both subclasses for now.
Bug: webrtc:12532
Change-Id: I0ec74fd241f16c1e1a6498b6baa621ca0489f279
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210340
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@{#33435}
2021-03-11 11:58:58 +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
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
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
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
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
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
189849fa0f
[Stats] Remove jitterBufferDelay TODO; it's already implemented.
...
This TODO says this metric is only available for audio and should also
be implemented for video, but ever since M76 this has been implemented
for both audio and video (https://crbug.com/webrtc/10450 ).
TBR=guido@webrtc.org , hta@webrtc.org
NOTRY=True
Bug: webrtc:10450
Change-Id: Icf2b60fdacae606c66f9d03492f107df9e32ba33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168343
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30485}
2020-02-07 15:14:38 +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
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
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
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
d2c336f892
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
...
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}
2019-07-04 08:13:45 +00:00
bfd343b9be
Add totalDecodeTime to RTCInboundRTPStreamStats
...
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}
2019-07-02 08:28:06 +00:00
