2226 Commits

Author SHA1 Message Date
Ilya Nikolaevskiy
68a7c415c5 Revert "Enforce stream id uniqueness in RtpSender::set_stream_ids"
This reverts commit 315b95ca11161bdea715d5316f92828edd41f0d5.

Reason for revert: Breaks internal bots.

Original change's description:
> Enforce stream id uniqueness in RtpSender::set_stream_ids
>
> https://w3c.github.io/webrtc-pc/#dfn-create-an-rtcrtpsender
> has a step saying
>   For each stream in streams, add stream.id to
>   [[AssociatedMediaStreamIds]] if it's not already there
>
> This applies to addTrack and setStreams and the set of streams in
> addTransceiver.
>
> BUG=webrtc:14769
>
> Change-Id: If6be813396a1987dfe49fd73f976f96c71459eaf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287864
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38937}

Bug: webrtc:14769
Change-Id: I6fd22ff0550c0894057fb1dc15f1b95819fa6df2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288744
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38940}
2022-12-21 13:56:05 +00:00
Philipp Hancke
315b95ca11 Enforce stream id uniqueness in RtpSender::set_stream_ids
https://w3c.github.io/webrtc-pc/#dfn-create-an-rtcrtpsender
has a step saying
  For each stream in streams, add stream.id to
  [[AssociatedMediaStreamIds]] if it's not already there

This applies to addTrack and setStreams and the set of streams in
addTransceiver.

BUG=webrtc:14769

Change-Id: If6be813396a1987dfe49fd73f976f96c71459eaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287864
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38937}
2022-12-21 11:28:49 +00:00
Markus Handell
2ba941e6bc ConnectionContext: remove media engine without blocking.
Bug: webrtc:14449
Change-Id: I445114c14f4d440a5a8cac003266047fe4588dab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288580
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38928}
2022-12-20 14:05:30 +00:00
Philipp Hancke
a2f5d45b81 Reland "sdp: measure rtp header extension collisions"
This is a reland of commit 6c27d56a2aeb2cff10a216d714552f4970d99d32
avoiding the Fuchsia/C++17 compilation issues.

Original change's description:
> sdp: measure rtp header extension collisions
>
> since extension ids are required to be unique in a BUNDLE group:
>   https://www.rfc-editor.org/rfc/rfc8843#name-rtp-header-extensions-consi
>
> Measure how much enforcing this would break in UMA first.
>
> BUG=webrtc:14782
>
> Change-Id: Ieaf7a436feea677032499e11ca14973eebda322e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288362
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38914}

Bug: webrtc:14782
Change-Id: If9449b0381ebe33ba30eff0d733a1f3c94470d5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288383
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38919}
2022-12-19 15:19:35 +00:00
Andrey Logvin
de57c57e1e Revert "sdp: measure rtp header extension collisions"
This reverts commit 6c27d56a2aeb2cff10a216d714552f4970d99d32.

Reason for revert: Breaks C++ 17 compilation (https://ci.chromium.org/ui/p/webrtc/builders/perf/Fuchsia%20Builder/157/overview). While the proposed fix doesn't seem to be trivial and causes some disagreements: https://webrtc-review.googlesource.com/c/src/+/288460

The bot will be added to CQ.

Original change's description:
> sdp: measure rtp header extension collisions
>
> since extension ids are required to be unique in a BUNDLE group:
>   https://www.rfc-editor.org/rfc/rfc8843#name-rtp-header-extensions-consi
>
> Measure how much enforcing this would break in UMA first.
>
> BUG=webrtc:14782
>
> Change-Id: Ieaf7a436feea677032499e11ca14973eebda322e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288362
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38914}

No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I454acc80ac222395acd640dc9f8bcea941855861
Bug: webrtc:14782
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288382
Commit-Queue: Andrey Logvin <landrey@google.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Owners-Override: Andrey Logvin <landrey@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#38915}
2022-12-19 09:36:05 +00:00
Philipp Hancke
6c27d56a2a sdp: measure rtp header extension collisions
since extension ids are required to be unique in a BUNDLE group:
  https://www.rfc-editor.org/rfc/rfc8843#name-rtp-header-extensions-consi

Measure how much enforcing this would break in UMA first.

BUG=webrtc:14782

Change-Id: Ieaf7a436feea677032499e11ca14973eebda322e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288362
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38914}
2022-12-17 09:35:53 +00:00
Philipp Hancke
fa67bcc27a generateKeyFrame: use GetParametersInternal
since the externally visible GetParameters() method modifies
the last transaction id which is neither required nor desirable
for generateKeyFrame.

BUG=chromium:1354101

Change-Id: Iae9ea36f78258d3be6d02eb2858c8bf9c4c672c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286740
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38911}
2022-12-16 12:24:12 +00:00
Harald Alvestrand
794d599741 Split media_channel and its dependencies from the rtc_media_base target
This helps in figuring out which dependencies exist, and gets closer
to obeying the "one target per .cc file" rule.

Test failures seem unrelated, so using No-Try.

No-Try: true
Bug: webrtc:14775
Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 12:15:22 +00:00
Philipp Hancke
a6c79d4bcf sdp: remove duplicate and shadowing variable
BUG=webrtc:14420

Change-Id: I814e5d7ee13767d0c677ebfddfb9b3bcf808972a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288360
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38908}
2022-12-16 11:02:57 +00:00
Harald Alvestrand
50454ef84a Apply PIMPL pattern to MediaSender and Receiver objects
This detaches the implementation (which is still merged)
from the objects used to interface to it.

Bug: webrtc:13931
Change-Id: I872ee10e4ed9fa432bfa231f723af1d3989d79d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288080
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38906}
2022-12-16 09:23:14 +00:00
Harald Alvestrand
c0d44d9d63 Split audio and video channels into Send and Receive APIs.
The implementation here has a number of changes that force the callers
that called the "channel" functions into specific interfaces rather than
just letting C++ take care of it; this should go away once there stops
being a common implementation class for those interfaces.

Bug: webrtc:13931
Change-Id: Ic4e279528a341bc0a0e88d2e1e76c90bc43a1035
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287640
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38888}
2022-12-14 11:00:17 +00:00
Philipp Hancke
b493db9b4d sdp: add tests for ParseMsidAttribute
increasing the line coverage to 100%.
Note that code coverage is still a very basic metric and
the tests still do not enforce the length of the attributes
nor their token-char-ness defined in
  https://www.rfc-editor.org/rfc/rfc8830.html#section-2

BUG=webrtc:14745

Change-Id: If5de5e1f8c32f7dae029916c6cd0a4f2b094f672
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286900
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38864}
2022-12-09 16:05:22 +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
Harald Alvestrand
36fafc8827 Split MediaChannel class to sender and receiver
This allows callers to differentiate on whether they need the
channel for sending or receiving purposes.

Note: This CL is incomplete, in that many places cast the pointers
to the concrete subclasses "VideoMediaChannel" and "AudioMediaChannel", which are not split into sending and receiving APIs.

The long term goal is to make two MediaChannel-like class APIs, with distinct implementations, and let the RtpSender and RtpReceiver manage those objects, rather than keeping them in the RtpTransceiver.

Bug: webrtc:13931
Change-Id: I8d56defe2287bd6552b71571cc6a5ec842927fa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287040
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38844}
2022-12-08 10:51:52 +00:00
Philipp Hancke
3c529893e0 sdp: ignore duplicate stream ids in msid parsing
https://www.rfc-editor.org/rfc/rfc8830.html#section-3.2.2
says
  Check if a MediaStream with the same WebIDL "id"
  attribute already exists. If not, create it.

Ignoring duplicates here satisfies this and brings the behavior
closer to Firefox:
  https://github.com/w3c/webrtc-pc/issues/2803

Also make tests use a std::string for the sdp input string.

BUG=webrtc:14745

Change-Id: Iccaabc08d865b779416f6ba4d2dfd5cff04133f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286422
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38840}
2022-12-07 18:03:59 +00:00
Philipp Hancke
e2652e168a sdp: limit mid attribute to 16 characters
removing the temporary limitation to 32 characters
since metrics suggests this is now fixed.

Metrics removal:
  https://chromium-review.googlesource.com/c/chromium/src/+/4079261

BUG=webrtc:12517,chromium:1375724

Change-Id: I11bec89463044afa99eeef2b3ecbe108eaa5c954
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286620
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38836}
2022-12-07 12:19:33 +00:00
Philipp Hancke
e093c481bf sdp: parse a=msid:<stream_id> w/o msid-appdata
parse
  a=msid:<stream_id>
since JSEP stipulates sending this syntax as track identifers
have become meaningless. The track id will be set to a random string.

  a=msid:<stream_id> <track_id>
remains supported for backward compability.

BUG=webrtc:14729

Change-Id: I86c073eb97cd613324271125de18a773235fc79d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285783
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38814}
2022-12-05 16:44:54 +00:00
Henrik Boström
a445e6a489 Delete deprecated disable_ipv6 flag.
M108 Stable has been released, which does not contain googIPv6 anymore,
and today the last downstream dependency on this flag was removed.

Let's delete!

Bug: webrtc:14608
Change-Id: Ia2d201f0da04b14961f891687b6135fc69b7767e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285720
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38786}
2022-12-01 11:01:02 +00:00
Philipp Hancke
41a8357170 Limit number of TURN servers to 32
Limit the number of TURN servers to 32 in order to allow the
prioritization to assume a fixed offset for (de)prioritizing
candidates. See
  https://github.com/w3c/webrtc-pc/pull/2679
for discussion including some data on current usage.

Guarded by WebRTC-LimitTurnServers which is used as a killswitch.

BUG=webrtc:13195

Change-Id: Ib12726af426ae4238aa7eb6aa062c71af52d495f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285340
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38767}
2022-11-29 17:04:11 +00:00
Philipp Hancke
f0ea56a0a2 sdp: measure codec collisions in bundle
as described in
  https://www.rfc-editor.org/rfc/rfc8843#name-payload-type-pt-value-reuse

... all codecs associated with the payload type number MUST share an
identical codec configuration

See also https://github.com/w3c/webrtc-stats/issues/664

Measure how much this would break in UMA first

BUG=webrtc:14420,webrtc:12716

Change-Id: Iafdc70248aa22bc37c15cc88a0c244398cb58176
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273881
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38759}
2022-11-29 14:09:30 +00:00
Ivan Rosales
447b9f3fde Unvirtualize more wasteful functions
Cleaning up instances of virtual functions with no overrides.
Savings of 1.4kb for chrome.dll file.

Note: These are the savings for Windows, relocation savings are
probably larger on other platforms.

GN args for builds:
use_goma=true
is_debug=false
target_cpu="x64"
use_lld=false
fatal_linker_warnings=false
symbol_level=2
dcheck_always_on = false

pe_summarize analysis pre-change -> change:
Size of out\Default\chrome.dll is 188.844544 MB
Size of out\SessionDescription\chrome.dll is 188.843520 MB
Memory size change from out\Default\chrome.dll
to out\SessionDescription\chrome.dll
       .text:    -704 bytes change
      .rdata:    -512 bytes change
      .pdata:     -48 bytes change
      .reloc:    -168 bytes change
Total change:   -1432 bytes

Bug: chromium:1371503
Change-Id: I51ad0a8acf3595fc499dbbcde2fab2d1bdf90fb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283940
Commit-Queue: Ivan Rosales <rosalesi@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38713}
2022-11-22 22:44:21 +00:00
Henrik Boström
a3a3b6d798 [Stats] If remote-inbound-rtp has no RTT, leave it undefined.
Bug: webrtc:14692
Change-Id: I49878449cd91b590f1aedef7676c3715d563ac61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284660
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38706}
2022-11-22 10:52:49 +00:00
Ilya Nikolaevskiy
6eb1e709da Reland "[DVQA] Create separate BUILD.gn file for video analyzer"
This reverts commit 76793c300fdd87fa8fd8be3dd2e5faf8c1916e96.

Reason for revert: Can't cleanly revert the old one. A forward fix will be provided.

Original change's description:
> Revert "[DVQA] Create separate BUILD.gn file for video analyzer"
>
> This reverts commit 116c0a53d4a35c6dee857eb4cc2b6ae233a0427c.
>
> Reason for revert: Breaks bot: https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_compile_dbg_ng/1415352/overview
>
>
> Original change's description:
> > [DVQA] Create separate BUILD.gn file for video analyzer
> >
> > Bug: None
> > Change-Id: I37dd2262bf3f52b2f5abe7934b9c41eaa27ffd17
> > No-try: True
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283141
> > Commit-Queue: Artem Titov <titovartem@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#38662}
>
> Bug: None
> Change-Id: Ieeb8c569560cb9d60d0c4d3c1268fa57f56b8157
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284000
> Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38672}

Bug: None
Change-Id: I74506eaa6a1060bf87e651881c86b4f576f447ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284020
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38676}
2022-11-18 11:43:45 +00:00
Ilya Nikolaevskiy
76793c300f Revert "[DVQA] Create separate BUILD.gn file for video analyzer"
This reverts commit 116c0a53d4a35c6dee857eb4cc2b6ae233a0427c.

Reason for revert: Breaks bot: https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_compile_dbg_ng/1415352/overview


Original change's description:
> [DVQA] Create separate BUILD.gn file for video analyzer
>
> Bug: None
> Change-Id: I37dd2262bf3f52b2f5abe7934b9c41eaa27ffd17
> No-try: True
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283141
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38662}

Bug: None
Change-Id: Ieeb8c569560cb9d60d0c4d3c1268fa57f56b8157
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284000
Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38672}
2022-11-18 09:18:32 +00:00
Alessio Bazzica
17887eb04a Reland "[ACM] iSAC audio codec removed"
This is a reland of commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1

Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}

Bug: webrtc:14450
Change-Id: Ia22c4d7724b6022238235fede93e36e570a49376
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283843
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38665}
2022-11-17 12:52:35 +00:00
Artem Titov
116c0a53d4 [DVQA] Create separate BUILD.gn file for video analyzer
Bug: None
Change-Id: I37dd2262bf3f52b2f5abe7934b9c41eaa27ffd17
No-try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283141
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38662}
2022-11-17 11:53:44 +00:00
Alessio Bazzica
fbeb76ab51 Revert "[ACM] iSAC audio codec removed"
This reverts commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1.

Reason for revert: breaks a downstream project

Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}

Bug: webrtc:14450
Change-Id: Ice138004e84e8c5f896684e8d01133d4b2a77bb7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283800
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38655}
2022-11-16 20:40:52 +00:00
Alessio Bazzica
b46c4bf27b [ACM] iSAC audio codec removed
Note: this CL has to leave behind one part of iSAC, which is its VAD
currently used by AGC1 in APM. The target visibility has been
restricted and the VAD will be removed together with AGC1 when the
time comes.

Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319

Bug: webrtc:14450
Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38652}
2022-11-16 16:42:55 +00:00
Shaofan Qi
45c882e4be Allow maxptime equal to minptime in the SDP.
The DCHECK crashes debug builds running some applications such as Webex.

Bug: None
Change-Id: I0061286c4c1d04964678a00014896f1fccd4685d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276460
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38644}
2022-11-16 10:54:03 +00:00
Henrik Boström
da4c102cbd Refactor some config plumbing in call/.
Address perkj's comments left in
https://webrtc-review.googlesource.com/c/src/+/283420. I was a bit
trigger-happy with the submit button.

Bug: chromium:1354491
Change-Id: Ifd052f75af3763b0b52807c31ea790e3efee921d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283521
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38638}
2022-11-16 09:18:40 +00:00
Ivan Rosales
2803ca27fb Remove virtual from wasteful functions
I've been working with SizeBench (binary size analysis tool) and it
reported that 39 virtual functions were not overridden. Removed
virtual keyword from each. SizeBench estimated waste 2.1kb. Change
made chrome.dll 5.3kb smaller. Since these 39 virtual functions
are never overridden, they are wasteful.

Note: These are the savings for Windows, relocation savings are probably larger on other platforms.

GN args for builds:
use_goma=true
is_debug=false
target_cpu="x64"
use_lld=false
fatal_linker_warnings=false
symbol_level=2
dcheck_always_on = false

pe_summarize analysis pre-change -> change:
Size of out\Default\chrome.dll is 187.205120 MB

Size of out\MediaContentDescription\chrome.dll is 187.199488 MB

Memory size change from out\Default\chrome.dll to
out\MediaContentDescription\chrome.dll
       .text:   -2624 bytes change
      .rdata:   -1984 bytes change
      .pdata:     -48 bytes change
      .reloc:    -644 bytes change
Total change:   -5300 bytes

Bug: chromium:1371503
Change-Id: Ib33829fada54abdf8fed33ec96f11a03ce6fcb68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281442
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ivan Rosales <rosalesi@google.com>
Cr-Commit-Position: refs/heads/main@{#38630}
2022-11-15 19:46:52 +00:00
Florent Castelli
acabb3641b pc: Add asynchronous RtpSender::SetParameters() call
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.

Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
2022-11-15 15:31:40 +00:00
Henrik Boström
cf2856b01c Add parameter to control the pacer's burst outside of field trials.
BurstyPacer is currently controlled via field trials. In order for
Chrome to be able to have burst without relying on a field trial, this
parameter is added.

When all burst experiments have concluded we may be able to have a
hardcoded constant instead, but for now the parameter is added to
RTCConfiguration.

NOTRY=True

Bug: chromium:1354491
Change-Id: I386c1651dbbcbf309c15ea3d3380cf8f632b5429
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283420
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38621}
2022-11-15 08:46:30 +00:00
Henrik Boström
1bef09708a Delete api/stats_types.h in favor of api/legacy_stats_types.h
The file was renamed, see
https://groups.google.com/u/1/g/discuss-webrtc/c/ZQiP4f_bpw4

Bug: webrtc:14180
Change-Id: Ia76c85ba7d9da6b3a93d0a67a4b6a5187e07e230
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283084
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38616}
2022-11-14 12:10:06 +00:00
Henrik Boström
3e6931b183 Rename api/stats_types.h to api/legacy_stats_types.h.
As to not break downstream projects, the old name api/stats_types.h is
kept around to help include api/legacy_stats_types.h. We can delete this
in a follow-up.

NOTRY=True

Bug: webrtc:14180
Change-Id: I270ca5e366ae36e324cbc9f982bbb066ab92d203
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283081
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38610}
2022-11-11 10:29:25 +00:00
Jeremy Leconte
a3e51df5f3 Add a new PeerConnectionE2EQualityTestFixture::AddPeer method.
Change-Id: Ic5879613db51a00e3e958931f5eda19fda1ae94a
Bug: webrtc:14627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282640
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38608}
2022-11-10 16:54:19 +00:00
Markus Handell
15a82c93d0 Metronome: complete API migration.
This CL finalizes the Metronome refactor undertaken in
crbug.com/1381982 and enables it again in call.cc.

Fixed: chromium:1381982
Change-Id: I1642103e9c8a3f2a1f12d7635a1b27310802c1c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282920
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38605}
2022-11-10 13:42:30 +00:00
Philipp Hancke
b83cd92a1a generateKeyFrame: validate rids argument
BUG=chromium:1354101

Change-Id: Ie850d807e47c72470a50daffec5679c7a23111dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282380
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38591}
2022-11-09 13:09:41 +00:00
Markus Handell
be400e465b Metronome: disable & refactor for single-threaded operation.
The Chromium implementation unfortunately has a rare deadlock.
Rather than patching that up, we're changing the metronome
implementation to be able to use a single-threaded environment
instead.

The metronome functionality is disabled in VideoReceiveStream2
construction inside call.cc.

The new design does not have listener registration or
deresigstration and instead accepts and invokes callbacks, on
the same sequence that requested the callback. This allows
the clients to use features such as WeakPtrFactories or
ScopedThreadSafety for cancellation.

The CL will be followed up with cleanup CLs that removes
registration APIs once downstream consumers have adapted.

Bug: chromium:1381982
Change-Id: I43732d1971e2276c39b431a04365cd2fc3c55c25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282280
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38582}
2022-11-08 12:23:40 +00:00
Yaowen Guo
999fff45ed Delete ChannelManager class declaration.
ChannelManager has been deleted, these declaration should also be deleted.

Bug: webrtc:13931
Change-Id: I2739a0424f61d6e659cb694a3f51bb6b90911cf9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282520
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38579}
2022-11-08 10:32:47 +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
a1b4eb2196 generateKeyFrame: add rids argument
and do the resolution of rids to layers. This has no effect yet
since the simulcast encoder adapter (SimulcastEncoderAdapter::Encode), the VP8 encoder (LibvpxVp8Encoder::Encode) and the OpenH264 encoder (H264EncoderImpl::Encode) all generate a key frame for all layers whenever a key frame is requested on one layer.

BUG=chromium:1354101

Change-Id: I13f5f1bf136839a68942b0f6bf4f2d5890415250
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280945
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38565}
2022-11-07 15:47:51 +00:00
Philipp Hancke
42e5ed38a7 stats: more consistent use of has_value() for optionals
replacing
  if (optional) { ...}
with the more explicit
  if (optional.has_value()) { ... }

No functional changes.

BUG=None

Change-Id: I005fd3df307880b07cfda0cbe435efb0e0717a88
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281362
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@{#38544}
2022-11-03 14:09:26 +00:00
Jonas Oreland
3e7e15d240 Reland "move relay server priority assignment to port_allocator"
This is a reland of commit b395f5bd5c5be7a3b54fc85ca482759a56a0cd6e
Downstream project has been fixed.

Original change's description:
> move relay server priority assignment to port_allocator
>
> which knows more about the internals of ICE.
> Remove the relay server config priority field which was used to
> specify the relative priority of TURN servers. This is now handled
> internally by CreateRelayPortArgs without being exposed.
>
> Also rename BasicPortAllocator::AddTurnServer to
> BasicPortAllocator::AddTurnServerForTesting since it is a test-only
> method.
>
> BUG=webrtc:13195,webrtc:14539
>
> Change-Id: Id36cbf0187b7a84d1a9b53860f31994f3c7589f0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280224
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38520}

Bug: webrtc:13195,webrtc:14539
Change-Id: I617b611de97e4013c7286e3345073ca7589065c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281662
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38543}
2022-11-03 12:16:08 +00:00
Philipp Hancke
df0a6228e5 transport controller: move ice transport initialization
out of the DTLS transport creation and into the ICE transport
creation.

BUG=None

Change-Id: I9f1a74bbabde842234e5ae7cc64aa9632ab2ba78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38541}
2022-11-03 08:18:53 +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
Artem Titov
936c1af16d Revert "move relay server priority assignment to port_allocator"
This reverts commit b395f5bd5c5be7a3b54fc85ca482759a56a0cd6e.

Reason for revert: Breaks downstream project. Jonas will help to reland this CL.

Original change's description:
> move relay server priority assignment to port_allocator
>
> which knows more about the internals of ICE.
> Remove the relay server config priority field which was used to
> specify the relative priority of TURN servers. This is now handled
> internally by CreateRelayPortArgs without being exposed.
>
> Also rename BasicPortAllocator::AddTurnServer to
> BasicPortAllocator::AddTurnServerForTesting since it is a test-only
> method.
>
> BUG=webrtc:13195,webrtc:14539
>
> Change-Id: Id36cbf0187b7a84d1a9b53860f31994f3c7589f0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280224
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38520}

Bug: webrtc:13195,webrtc:14539
Change-Id: I7ca087a272793908f003cea6c32efe6214e54028
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281340
Owners-Override: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38524}
2022-11-01 13:29:28 +00:00
Philipp Hancke
b395f5bd5c move relay server priority assignment to port_allocator
which knows more about the internals of ICE.
Remove the relay server config priority field which was used to
specify the relative priority of TURN servers. This is now handled
internally by CreateRelayPortArgs without being exposed.

Also rename BasicPortAllocator::AddTurnServer to
BasicPortAllocator::AddTurnServerForTesting since it is a test-only
method.

BUG=webrtc:13195,webrtc:14539

Change-Id: Id36cbf0187b7a84d1a9b53860f31994f3c7589f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280224
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38520}
2022-11-01 09:59:37 +00:00
Philipp Hancke
fd91d02210 ice: include tiebreaker in computation of foundation attribute (tests)
split from the actual change for easier review and cherry-picking.

BUG=webrtc:14605

Change-Id: I7dbaf8e1f4a03f35a5d8c4da1a2784b00589bfc3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280680
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38519}
2022-11-01 07:47:23 +00:00