Commit graph

126 commits

Author SHA1 Message Date
Johannes Kron
bff2e27076 Deflake RTCStatsIntegrationTest::GetStatsFromCallee
The test may sometimes fail because the round trip time has not been
estimated. Wait until the report contains the round trip time before
proceeding, or fail after 10 s.

Fixed: webrtc:14952
Change-Id: I9127b8ee6afa7454d061de96f002422d7d4af428
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295724
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39438}
2023-03-01 14:01:10 +00:00
Fredrik Hernqvist
5adc2b6969 Correct RTCAudioPlayoutStats type and add kind field.
Bug: webrtc:14653
Change-Id: Idb85ce440620fc5b818a3b23a63ac062a443cc81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291330
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39196}
2023-01-25 14:30:41 +00:00
Fredrik Hernqvist
828de8036d Populate RTCInboundRtpStreamStats::playoutId when appropriate
Bug: webrtc:14653
Change-Id: I0c59604b218d0839a126c02914626b8ed2bee76c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291040
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39149}
2023-01-19 15:44:36 +00:00
Evan Shrubsole
9b235cd93b Add scalability mode to RTCOutboundRtpStreamStats stats
This is in the webrtc-stats spec at
https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats-scalabilitymode.

This adds the scalability mode to CodecSpecificInfo which is used to
plumb the modes for each simulcast layer.

TBR=orphis@webrtc.org

Tested: Compiled into Chrome and confirmed the scalability mode set for AV1, VP9, VP8 and H264 software encoders in chrome://webrtc-internals.
Bug: webrtc:14730
Change-Id: I71ceba8f6485a4f4a73e0856031b8d5f16f913f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285085
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38847}
2022-12-08 11:46:06 +00:00
Evan Shrubsole
13c0be44b3 Add power efficient stats to RTC stats
As the exposure of power efficient stats to JavaScript are limited as
to reduce the fingerprinting surface to getStats, a new RTCStatsMember
derivation, RTCLimitedStatsMember, was added in this change. This sets
the exposure criteria of the stat on the type, which keeps the size of
the RTCStatsMember class the same and allows for extension in the future
for new types of stat restrictions.

Bug: webrtc:14483
Change-Id: Ib0303050a112441ba2416fd5f004dd8be26b47ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279021
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38576}
2022-11-08 08:35:47 +00:00
Philipp Hancke
0487c5797a stats: implement candidate-pair lastPacket(Sent|Received)Timestamp
https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-lastpacketsenttimestamp
https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-lastpacketreceivedtimestamp

which are useful together with the ice-restart-necessary logic mentioned
in
  https://w3c.github.io/webrtc-pc/#dictionary-rtcofferoptions-members

BUG=webrtc:14619

Change-Id: I4a8ab00a37fbd4af8b948720c83787cbdfc6b9a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281281
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38534}
2022-11-02 12:16:21 +00:00
Henrik Boström
adbcbf73fa [Stats] Delete 'track' metrics that have previously been moved.
These have all been moved to "inbound-rtp" and now that upstream
projects have migrated we can delete the old location.

Unblocks https://crbug.com/webrtc/14175

Bug: webrtc:14521, webrtc:14524
Change-Id: Ia2bfa399d62304cc0ead0e65c340dfad20acc530
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281183
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38532}
2022-11-02 09:21:04 +00:00
Henrik Boström
45b35d442d Unship track.totalFramesDuration/sumSquaredFrameDurations.
These metrics were not only non-standard, but residing in the
non-standard "track" stats object that we want to delete. As per
https://github.com/w3c/webrtc-stats/issues/695#issuecomment-1259611462
these metrics are no longer needed because we already have
inbound-rtp.totalInterFrameDelay/totalSquaredInterFrameDelay which is
basically the same thing.

// mac_rel infra failures are unrelated
NOTRY=True

Bug: webrtc:14522
Change-Id: I565da42514a93f15532ba8357dd006547a5296ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278100
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38509}
2022-10-31 15:09:10 +00:00
Henrik Boström
aebba7b468 [Stats] Expose totalPacketSendDelay for audio as well.
This information is now readily available. Let's expose it.

In practise we don't pace audio by default and the delay is ~0, however
we can tell that this metric is working as intended by setting
PacingController's pace_audio_ to true via the "WebRTC-Pacer-BlockAudio"
field trial. In this case chrome://webrtc-internals/ plots neats graphs
for audio send delay.

Bug: webrtc:10635
Change-Id: Iecfd93bb84ec61e5d54232769a9e7a500601b199
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280523
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38483}
2022-10-27 10:33:16 +00:00
Henrik Boström
c5f8f800a2 [Stats] Add googTimingFrameInfo to the modern API.
This is exposing something that is already exposed in the legacy
getStats() API and is only available if the "video-timing" header
extension is used. Adding this metric here should unblock legacy
getStats() API deprecation. The follow-up to unship or standardize this
metric is tracked by https://crbug.com/webrtc/14586.

Bug: webrtc:14587
Change-Id: Ic3d45b0558d7caf4be2856a4cd95b88db312f85e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38444}
2022-10-19 17:02:18 +00:00
Henrik Boström
15166b2fa4 [ModernStats] Mark obsolete stats as [[deprecated]].
This includes the stats dictionaries that have been made obsolete in
the spec and whose IDs are prefixed "DEPRECATED_":
- RTCMediaStreamTrackStats
- RTCMediaStreamStats

There is an ongoing experiment to unship these stats dictionaries in
Chrome (https://crbug.com/1374215). Marking then as [[deprecated]] helps
alert other dependencies that these classes are deprecated.

In the meantime, the "DEPRECATED_RTCMediaStreamTrackStats" prefix makes
it possible to use the deprecated classes.

# Unrelated infra failures
NOTRY=True

Bug: webrtc:14175, webrtc:14419
Change-Id: I498d370294058a628278e1e5b027cd12e24ad31a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38439}
2022-10-19 09:58:37 +00:00
Henrik Boström
2fb83072db Move more non-standard metrics to inbound-rtp.
They may be non-standard, but they shouldn't be on a stats dictionary
that is deprecated (track is going away soon-ish). By moving them to
inbound-rtp they can continue to exist beyond track deprecation and
live in the right place in case we decide to standardize them later.

To help downstream projects transitions, the metrics are temporarily
available in both old and new locations. Delete of old location will
happen in a follow-up CL. TODOs added.

Bug: webrtc:14524
Change-Id: I2008060fa4ba76cde859d9144d2bb9648c7ff9af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38315}
2022-10-07 07:22:04 +00:00
Henrik Boström
c57a28c46b Move pause and freeze metrics to standardized location.
These metrics were recently standardized. Part of the standardization
effort was to move them from obsolete "track" stats (on track for
deprecation and removal: https://crbug.com/webrtc/14175) into the
"inbound-rtp" stats which are not deprecated.

To ease transition for downstream projects, the metrics are temporarily
duplicated in both the old and new locations. In a follow-up CL, they
will be deleted from "track".

Bug: webrtc:14521
Change-Id: I0d9036472607a8c717ec823a458a79a49ddb80c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278080
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38308}
2022-10-06 10:52:22 +00:00
Henrik Boström
a494e4b517 Move packetsDiscarded to inbound-rtp.
packetsDiscarded was previously moved to RTCInboundRtpStreamStats:
https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*

Bug: webrtc:14514
Change-Id: I322b64ede4e64cef1c8234e9626121d96d945355
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277820
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38297}
2022-10-05 09:00:18 +00:00
Philipp Hancke
0e3cd63062 stats: add missing ice candidate stats
added in https://github.com/w3c/webrtc-stats/pull/611
* foundation
* relatedAddress
* relatedPort
* usernameFragment
* tcpType

BUG=webrtc:14480

Change-Id: I5f43373fbbc7c780b8dafb6e2ace2c27f5e22970
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276780
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38292}
2022-10-04 18:02:28 +00:00
Byoungchan Lee
636dc3d208 Implement RTCOutboundRtpStreamStats.targetBitrate for video
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-targetbitrate

Bug: webrtc:13394
Change-Id: I4749b38088a24d1a775137d5fe2c65f96effd185
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276380
Auto-Submit: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38170}
2022-09-22 12:37:30 +00:00
Philipp Hancke
684e241323 stats: implement outbound-rtp.active
implementing
  https://github.com/w3c/webrtc-stats/pull/649

BUG=webrtc:14291

Change-Id: Ib8453d4d7c335834cd8dd2aa29111aef26211dff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269520
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#37639}
2022-07-28 13:35:40 +00:00
Ivo Creusen
1a84b565ac Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay
This metric was recently added to the standard (see https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferminimumdelay). This CL implements it for audio streams.

Bug: webrtc:14141
Change-Id: I79d918639cd12361ebbc28c2be41549e33fa7e2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262770
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37567}
2022-07-20 09:14:03 +00:00
Ivo Creusen
11fdb08282 Implement RTCInboundRTPStreamStats.JitterBufferTargetDelay
This CL also removes the existing non-standard implementation of the metric.

Bug: webrtc:14147, webrtc:11789
Change-Id: I70fd1c451dfd59380fe5ce959086f37b31697c16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265360
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37441}
2022-07-05 11:34:53 +00:00
Byoungchan Lee
d58f526384 Always inject PacketSocketFactory in FakePortAllocator
This CL removes the use of the rtc::Thread::socketserver() method
in one place.

Bug: webrtc:13145
Change-Id: I1a1b2501450788263d5280c43e4328ade46f4146
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263320
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#37340}
2022-06-27 12:45:28 +00:00
Philipp Hancke
a116da5d03 test: fix usage of legacy media_type
since the check is done for the more modern "kind" attribute

BUG=None

Change-Id: I520b62f312348b006d8147ce7b2b483c78657b98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266141
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37258}
2022-06-17 12:54:39 +00:00
Henrik Boström
1ab61886a9 Implement Outbound/InboundRtpStreamStats.mid.
This is what allowed us to remove "transceiver" stats from the spec.

Bug: webrtc:14191
Change-Id: I687a2dd97de016832005cb4271f6e1a0e0560cd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266022
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37247}
2022-06-17 08:44:09 +00:00
Henrik Boström
a6c7d5c8ce Implement RTCInboundRtpStreamStats.trackIdentifier.
This should allow standard stats users not to have to rely on the
obsolete "track" stats.

Bug: webrtc:14174
Change-Id: I24e5e1478ee47c73c12fcdecf7314f41fcc76bc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266020
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37246}
2022-06-17 08:32:09 +00:00
Henrik Boström
5abfc920b5 Reland "[Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs."
This is a reland of commit 626f87d905

Original change's description:
> [Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs.
>
> In preparation for the spec moving closer to PR, let's not have
> placeholder metrics not implemented.
>
> Bug: webrtc:14167
> Change-Id: If4688ef85b57f88154d490186b306b30414874e4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37205}

Bug: webrtc:14167
Change-Id: Ib12488fb8510fb3430e92bcd72d88c7879ecb0ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265861
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37226}
2022-06-15 15:03:18 +00:00
Niels Möller
105711e9ad Move rtc::make_ref_counted to api/
Bug: webrtc:12701
Change-Id: If49095b101c1a1763c2a44a0284c0d670cce953f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265390
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37219}
2022-06-15 09:47:38 +00:00
Henrik Boström
67d2d35443 Revert "Reland "[Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs.""
This reverts commit 2843bbc96d.

Reason for revert: Even more references to unimplemented metrics remaining...

Original change's description:
> Reland "[Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs."
>
> This is a reland of commit 626f87d905
>
> Original change's description:
> > [Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs.
> >
> > In preparation for the spec moving closer to PR, let's not have
> > placeholder metrics not implemented.
> >
> > Bug: webrtc:14167
> > Change-Id: If4688ef85b57f88154d490186b306b30414874e4
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37205}
>
> Bug: webrtc:14167
> Change-Id: Ifdc37e7a48fea516c727c06d2f510780386cb204
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265805
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37215}

Bug: webrtc:14167
Change-Id: I959d61512d5896224302a70aadbac6f75afc819e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265810
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#37217}
2022-06-15 08:11:48 +00:00
Henrik Boström
2843bbc96d Reland "[Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs."
This is a reland of commit 626f87d905

Original change's description:
> [Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs.
>
> In preparation for the spec moving closer to PR, let's not have
> placeholder metrics not implemented.
>
> Bug: webrtc:14167
> Change-Id: If4688ef85b57f88154d490186b306b30414874e4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37205}

Bug: webrtc:14167
Change-Id: Ifdc37e7a48fea516c727c06d2f510780386cb204
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265805
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37215}
2022-06-15 06:29:38 +00:00
Henrik Boström
378b1c6826 Revert "[Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs."
This reverts commit 626f87d905.

Reason for revert: Breaks one downstream project, will re-land after the dependency stops referencing an unimplemented RTT metric

Original change's description:
> [Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs.
>
> In preparation for the spec moving closer to PR, let's not have
> placeholder metrics not implemented.
>
> Bug: webrtc:14167
> Change-Id: If4688ef85b57f88154d490186b306b30414874e4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37205}

Bug: webrtc:14167
Change-Id: I7e9ac60eb474b44fab678d4c08ddcae846ce456c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265800
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37206}
2022-06-14 08:48:37 +00:00
Henrik Boström
626f87d905 [Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs.
In preparation for the spec moving closer to PR, let's not have
placeholder metrics not implemented.

Bug: webrtc:14167
Change-Id: If4688ef85b57f88154d490186b306b30414874e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37205}
2022-06-14 07:46:57 +00:00
Philipp Hancke
6fb8d1a2d7 stats: expose minPlayoutDelay as nonstandard stat
This currently only exists as a goog legacy stat and has no spec
equivalent according to
  https://docs.google.com/document/d/1z-D4SngG36WPiMuRvWeTMN7mWQXrf1XKZwVl3Nf1BIE/edit
Yet it is useful to debug issues sometimes. Exposing it as a
nonstandard stat will make it show up in chrome://webrtc-internals,
removing a need to switch to the legacy stats API there.

BUG=webrtc:14118

Change-Id: I506357ad54ff33df3ba46fb81558aa32187ac8e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264420
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37055}
2022-05-31 10:05:35 +00:00
Jianhui Dai
b1ba85385e Eliminate unnecessary RTC_TRACE_EVENTS_ENABLED
Bug: webrtc:14073
Change-Id: I6365cc17393be52c11312dfa954783a3e135cb8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262263
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36929}
2022-05-19 09:52:47 +00:00
Philipp Hancke
0359ba2225 stats: add frame assembly time stats
implements a total frame assembly time statistic that measures the
cumulative time between the arrival of the first packet of a frame
(the lowest reception time) and the time all packets of the frame have
been received (i.e. the highest reception time)

This is similar to totalProcessingDelay
  https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame.

This statistic is useful for evaluating mechanisms like NACK and FEC
and gives some insight into the behavior of the pacer sending the
packets.
Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added.

Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as

totalAssemblyTime of type double
	Only exists for video. 	The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received.
    Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible.

    This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket.

framesAssembledFromMultiplePacket of type unsigned long
	Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet.
	For such frames the totalAssemblyTime is incremented.

BUG=webrtc:13986

Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36922}
2022-05-18 09:16:10 +00:00
Philipp Hancke
1f49157b41 stats: implement transport iceState
https://w3c.github.io/webrtc-stats/#dom-rtctransportstats-selectedcandidatepairid

BUG=webrtc:14022

Change-Id: I206bff7048d2df3e3aff0af55072097f49d54e8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261720
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36840}
2022-05-10 13:55:21 +00:00
Philipp Hancke
95b1a3497c stats: implement iceLocalUsernameFragment
https://www.w3.org/TR/webrtc-stats/#dom-rtctransportstats-icelocalusernamefragment

BUG=webrtc:14022

Change-Id: If56ebe66d83f4e535c2245f2ca3848469914679f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261243
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36772}
2022-05-05 08:08:48 +00:00
Philipp Hancke
cc1b9b060d stats: implement iceRole
https://www.w3.org/TR/webrtc-stats/#dom-rtctransportstats-icerole

BUG=webrtc:14022

Change-Id: I88de2c843a2042ce99076d55ce41be22589e2d92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261201
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36766}
2022-05-05 05:05:40 +00:00
Philipp Hancke
a16a6a6341 stats: implement inbound-rtp totalProcessingDelay for video
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay

BUG=webrtc:13984

Change-Id: Ifd821bd8553add46218f09a11366096d62f5d09f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259768
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36732}
2022-05-02 10:56:22 +00:00
Philipp Hancke
69c1df2f44 stats: add dtlsRole to transport
https://w3c.github.io/webrtc-stats/#dom-rtctransportstats-dtlsrole

BUG=webrtc:13978

Change-Id: Ib158427d2df0307884381bdd46c411f60f56a371
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259761
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36730}
2022-05-02 10:13:54 +00:00
Philipp Hancke
a3b5c4e027 test: replace media_type with kind
media_kind is the old name (that is kept around since we can't deprecate)

BUG=None

Change-Id: I445441a54bb4ff408502d1aba6834cdac874324b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259766
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36625}
2022-04-22 14:53:08 +00:00
Niels Möller
afb246b5a9 Update pc/ to not use implicit conversion from scoped_refptr<T> to T*.
Bug: webrtc:13464
Change-Id: I768646af8ded6338ef51486b8d69db1ad71e9a2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259500
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/main@{#36588}
2022-04-20 13:18:33 +00:00
Jonas Oreland
0d13bbd4b1 Extend RTCIceCandidateStats with non-standard network_adapter_type
This cl/ extends the RTCIceCandidateStats object with
network_adapter_type and vpn, so that it maps the underlying
WebRTC objects completly.

Bug: webrtc:13773
Change-Id: I5cf79972c60ca6bf2a127dc96fa90811263ba6fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253241
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36110}
2022-03-02 11:13:18 +00:00
Harald Alvestrand
c24a2189d7 Update IWYU tool with a mapping file
Also apply IWYU to all .cc files in pc/, and correct BUILD file to match.
Note: Some files came out wrong when iwyu was applied. These are not included.

Bug: none
Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-24 11:05:06 +00:00
Henrik Boström
62995db2fc Change default sdp_semantics to kNotSpecified.
In preparation for switching the default from kPlanB to kUnifiedPlan,
which could cause subtle bugs for those not prepared for it, we change
the default to kNotSpecified. The only purpose of kNotSpecified is to
crash, forcing any dependencies to explicitly set their sdp_semantics
value.

Tests are updated to explicitly set sdp_semantics when necessary, and
where the test does not care we update to kUnifiedPlan.

If this change lands without getting reverted we can let it sit for a
few weeks, after which we should change the default to kUnifiedPlan and
delete kNotSpecified.

Bug: webrtc:11121
Change-Id: I19b669b0735d78e269e19eaae86c2d7d95a91141
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242968
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35651}
2022-01-10 14:56:03 +00:00
Byoungchan Lee
efe46b6bee Change the type of RTCVideoSourceStats.framesPerSecond
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcvideosourcestats-framespersecond

Bug: webrtc:12905
Change-Id: If53e2e480e2d6f687c3f8bb95a9e1d1e386fe9c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237420
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35352}
2021-11-16 11:21:41 +00:00
Jakob Ivarsson
bf0874568c Implement RTCOutboundRtpStreamStats.targetBitrate for audio.
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-targetbitrate

Bug: webrtc:13377
Change-Id: I98dd263e0b9d6e2ca94969d2a91857b14cd65f70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237402
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35337}
2021-11-12 09:24:34 +00:00
Taylor Brandstetter
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
Artem Titov
880fa8169b Reland "Use backticks not vertical bars to denote variables in comments for /pc"
Original change's description:
> Revert "Use backticks not vertical bars to denote variables in comments for /pc"
>
> This reverts commit 37ee0f5e59.
>
> Reason for revert: Revert in order to be able to revert https://webrtc-review.googlesource.com/c/src/+/225642
>
> Original change's description:
> > Use backticks not vertical bars to denote variables in comments for /pc
> >
> > Bug: webrtc:12338
> > Change-Id: I88cf10afa5fc810b95d2a585ab2e895dcc163b63
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226953
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Artem Titov <titovartem@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34575}
>
> TBR=hta@webrtc.org,titovartem@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I5eddd3a14e1f664bf831e5c294fbc4de5f6a88af
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12338
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227082
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34577}

Bug: webrtc:12338
Change-Id: I96bd229b73613c162d11d75fa4f5934e1b4295c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227087
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34611}
2021-07-30 22:13:59 +00:00
Björn Terelius
fd05d6f504 Revert "Use backticks not vertical bars to denote variables in comments for /pc"
This reverts commit 37ee0f5e59.

Reason for revert: Revert in order to be able to revert https://webrtc-review.googlesource.com/c/src/+/225642

Original change's description:
> Use backticks not vertical bars to denote variables in comments for /pc
>
> Bug: webrtc:12338
> Change-Id: I88cf10afa5fc810b95d2a585ab2e895dcc163b63
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226953
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34575}

TBR=hta@webrtc.org,titovartem@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: I5eddd3a14e1f664bf831e5c294fbc4de5f6a88af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227082
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34577}
2021-07-27 22:10:24 +00:00
Artem Titov
37ee0f5e59 Use backticks not vertical bars to denote variables in comments for /pc
Bug: webrtc:12338
Change-Id: I88cf10afa5fc810b95d2a585ab2e895dcc163b63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226953
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34575}
2021-07-27 20:52:02 +00:00
Minyue Li
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
Jakob Ivarsson
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