Danil Chapovalov
02113a2169
Pass Environment into RtcpReceiver
...
to avoid relying on the global field trials.
Bug: webrtc:362762208
Change-Id: I94e96f0a3f16cfd64f7deb4deb4aaa924ac1bba8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361865
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42982}
2024-09-09 11:36:29 +00:00
Artem Titov
3652dd30db
Review documentation and update review date
...
Bug: b/362492031, b/362492070, b/362492356, b/364207037, b/364831690
Change-Id: Ic889c731b98f8876c0ee31c0bda91a5a18b3add1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362100
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42981}
2024-09-09 11:21:55 +00:00
Dor Hen
65b59a9c2d
Prepend webrtc ns to StrJoin calls in dcsctp ns
...
Bug: webrtc:365299886
Change-Id: I4eb87a2b116c3e18b1a84865fab0a22a6084912c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42980}
2024-09-09 11:16:56 +00:00
Sergey Silkin
26146bbce0
Add support for screencast with temporal layering to SvcRateAllocator
...
SvcRateAllocator assumed no temporal layering for screencast content and allocated all bitrate to base temporal layer. Now it distributes bitrate to spatial and temporal layers (if configured) no matter of content type.
Bug: webrtc:351644568, b/364190191
Change-Id: I445f0157d2c14cad033648693dc0564ae97023e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362080
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42979}
2024-09-09 11:12:33 +00:00
webrtc-version-updater
405f343099
Update WebRTC code version (2024-09-07T04:08:12).
...
Bug: None
Change-Id: I4ee7465d2037c07e2f1b0654a30cd13d0d2e5176
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361709
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42978}
2024-09-07 06:11:23 +00:00
Fanny Linderborg
6f64ae1ff5
Extract corruption detection message to its own target
...
Bug: webrtc:358039777
Change-Id: I6bc064aaba4c5b7f9b55215414e70e55eb0e0f64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361864
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42977}
2024-09-06 13:32:35 +00:00
Victor Boivie
65b46da1f8
dcsctp: Don't send FORWARD-TSN in its own chunk
...
This is a rollback of a change list [1] that was an attempt to avoid a
bug in usrsctp[2], but it wasn't very successful. But with the usrsctp
bug fixed, we can revert to the desired state; that we bundle
FORWARD-TSN with other control and data chunks in a SCTP packet.
[1] https://webrtc-review.googlesource.com/c/src/+/229101
[2] https://github.com/sctplab/usrsctp/issues/597
Bug: webrtc:42223134
Change-Id: I2f3b511c91639e6b9516160190600beb0c04b5fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361862
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42976}
2024-09-06 13:29:52 +00:00
Victor Boivie
7929ef578a
dcsctp: Add test for stream reset pending
...
Bug: None
Change-Id: Ida040461dda53b438d88df4a38518f334cadc379
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361861
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Auto-Submit: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42975}
2024-09-06 13:20:58 +00:00
Henrik Lundin
c9aaf11985
Remove use of AcmReceiver in ChannelReceive
...
ChannelReceive is now owning and interfacing with NetEq directly.
A new ResamplerHelper is added to acm_resampler.cc/.h, to do the
audio resampling that was previously done inside AcmReceiver.
AcmReceiver still remains, since it is used in other places for now.
Bug: webrtc:14867
Change-Id: If3eb6415e06b9b5e729d393713f3fccb31b0570f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361820
Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42974}
2024-09-06 12:47:36 +00:00
Jonas Oreland
3ad2c8d717
Make getNumObservers @VisibleForTesting so that it can be tested outside of package org.webrtc
...
Bug: webrtc:0
Change-Id: I66259bba8c724889df3ee22a8d6b9710a04ba671
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361863
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42973}
2024-09-06 12:36:59 +00:00
Ilya Nikolaevskiy
9f096a8707
Allow VideoEncoderSoftwareFallbackWrapper to return SIMULCAST_PARAMS_NOT_SUPPORTED
...
Now some HW encoders support simulcast. If parameters are not suitable for
single encoder simulcast, the error code should be forwarded back to
SimulcastEncoderAdapter instead of trying software fallback.
Bug: webrtc:347737882
Change-Id: Id02ff1afc012cd46761d9530b1ce368d5dc480bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361744
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42972}
2024-09-06 12:26:43 +00:00
Björn Terelius
c7da857813
Fix lint issues in pacing/
...
(Mostly include-what-you-use.)
Bug: webrtc:42226242
Change-Id: I3717cccb24ac4f0a5443995d1d355561a5a54c8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361601
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42971}
2024-09-06 11:28:43 +00:00
karllen.zheng@ringcentral.com
f92f39e618
Increase the default maximum jitter buffer size to 200 packets for Android.
...
Bug: webrtc:42220461
Change-Id: I6dadd8357173c79af0290199d9ed12a0e0417f4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361673
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42970}
2024-09-06 11:21:34 +00:00
Jakob Ivarsson
4334cdfc5c
Reland "Return audio stats regarless if we have a codec."
...
This is a reland of commit 7fff587a096c6ef40f5601f47ef50b221b3a4abf
Original change's description:
> Return audio stats regarless if we have a codec.
>
> Bug: b/331602608
> Change-Id: I2d12a3ed83645fe1e7cbd8950fd86d5ba2d7c94d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361743
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42964}
Bug: b/331602608
Change-Id: I95c89e7059005bc8dd8569ef41bfe9e863b4082f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361762
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42969}
2024-09-06 08:25:36 +00:00
webrtc-version-updater
5913803350
Update WebRTC code version (2024-09-06T04:04:53).
...
Bug: None
Change-Id: Ic1eb02693eb74155f383ad0b5385acbdc2261555
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361671
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42968}
2024-09-06 05:54:44 +00:00
chromium-webrtc-autoroll
f5c5fb9390
Roll chromium_revision 040c638bdb..c03ff62a28 (1351313:1351560)
...
Change log: 040c638bdb..c03ff62a28040c638bdb..c03ff62a28df5af89cb0..e9e5cfb550259b72d9ae..136f79a9af9c4d44a89a..012c060b769f9e2bf58d..5afb02692a225240b0c1..3dde5eea0ea1aafecb38..aa75790deahttps://boringssl.googlesource.com/boringssl.git/+log/9224e6d138..97322b29e8
* src/third_party/depot_tools: 62fc3a1d24..9bbf9b0541b6405588e4..a4812f9f7276963b6cb4..03d8b3e17ccd9d18ea6d..5a4e9143e8040c638bdb..c03ff62a28https://webrtc-review.googlesource.com/c/src/+/361800
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42967}
2024-09-05 18:42:25 +00:00
Danil Chapovalov
e922cd1262
Use Environment instead of Clock in ModuleRtpRtcp and its RTP subcomponents
...
Bug: webrtc:362762208
Change-Id: I35af5cf3ed48e2c738c12df2ed9117a640ed0ff7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361720
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42966}
2024-09-05 15:58:43 +00:00
Jakob Ivarsson
e94c7da1df
Revert "Return audio stats regarless if we have a codec."
...
This reverts commit 7fff587a096c6ef40f5601f47ef50b221b3a4abf.
Reason for revert: breaks downstream test
Original change's description:
> Return audio stats regarless if we have a codec.
>
> Bug: b/331602608
> Change-Id: I2d12a3ed83645fe1e7cbd8950fd86d5ba2d7c94d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361743
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42964}
Bug: b/331602608
Change-Id: Ia87ef3b3066e1373654e1f0d96726217e7ed4117
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361761
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42965}
2024-09-05 15:16:06 +00:00
Jakob Ivarsson
7fff587a09
Return audio stats regarless if we have a codec.
...
Bug: b/331602608
Change-Id: I2d12a3ed83645fe1e7cbd8950fd86d5ba2d7c94d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361743
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42964}
2024-09-05 14:44:43 +00:00
Markus Handell
5162dc3370
Reland "TaskQueueStdlib: Stop spamming on idle."
...
This reverts commit 9463095df81b737a4be6bf10d14dab679457dd7b.
Reason for revert: Problem fixed.
Original change's description:
> Revert "TaskQueueStdlib: Stop spamming on idle."
>
> This reverts commit 2fbaa8e3a3ef7d7f26e82bc3325fdca33f60c1d3.
>
> Reason for revert: Breaks tests
>
> Original change's description:
> > TaskQueueStdlib: Stop spamming on idle.
> >
> > When the stdlib task queue has no work for >= 3s on
> > Android, it will emit bogus deadlock warnings. Fix
> > this by ensuring we're not triggering this behavior.
> >
> > Fixed: b/364436627
> > Change-Id: I3fe02e13993cbb4a59d5669df06c4c293d237bc4
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361721
> > Commit-Queue: Markus Handell <handellm@webrtc.org>
> > Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#42955}
>
> Change-Id: Ie80d86f6aa0cdff3bc5ee1d79fa4c8144009e33f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361741
> Auto-Submit: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#42960}
Fixed: b/364436627
Change-Id: I7baa8c5c169e5c63b28e1a5363c06e3d8d3f4c7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361742
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Markus Handell <handellm@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42963}
2024-09-05 13:18:08 +00:00
Ilya Nikolaevskiy
5ac7495701
Prepare to use SimulcastToSvcConverter in chromium
...
Allow moving the class, add required RTC_EXPORTs
Bug: webrtc:347737882
Change-Id: Iac14e6f62adfa13ff1e757918a2f92009f5be36f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361760
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42962}
2024-09-05 12:20:57 +00:00
Henrik Boström
d4c5843bae
Undo recent changes to initial frame dropper, fixing a regression.
...
A recent bugfix[1] introduced having to reconfigure the encoder in
response to restrictions updating for the sake of not getting stuck at
the wrong resolution in some cases.
But the newly added ReconfigureEncoder() calls happened too early/often,
and the initial frame dropper got confused thinking resolution changes
were not in response to adaptation (restrictions) when they in fact were.
The CL wrongly disabled the "reset initial frame dropper" logic, when
the correct solution to this problem should have been to delay the
ReconfigurationEncoder() call to the next frame (using
`pending_encoder_reconfiguration_ = true`).
With this delay the initial frame dropper is not confused and we can
restore the old "reset" logic, fixing the regression.
[1] https://webrtc-review.googlesource.com/c/src/+/360200
Bug: b/364252657
Change-Id: I6b93f4cc44eb12b1bbbda0d8d1e9906c29b615a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361740
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42961}
2024-09-05 12:11:11 +00:00
Markus Handell
9463095df8
Revert "TaskQueueStdlib: Stop spamming on idle."
...
This reverts commit 2fbaa8e3a3ef7d7f26e82bc3325fdca33f60c1d3.
Reason for revert: Breaks tests
Original change's description:
> TaskQueueStdlib: Stop spamming on idle.
>
> When the stdlib task queue has no work for >= 3s on
> Android, it will emit bogus deadlock warnings. Fix
> this by ensuring we're not triggering this behavior.
>
> Fixed: b/364436627
> Change-Id: I3fe02e13993cbb4a59d5669df06c4c293d237bc4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361721
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42955}
Change-Id: Ie80d86f6aa0cdff3bc5ee1d79fa4c8144009e33f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361741
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42960}
2024-09-05 11:31:52 +00:00
Jakob Ivarsson
6255a7f3a0
Avoid negative timestamp in SourceTracker.
...
Bug: b/364184684
Change-Id: If03cd697fed05c24549b9ef80bbaf9f11b47d8bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361640
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42959}
2024-09-05 10:43:37 +00:00
chromium-webrtc-autoroll
3047e6449a
Roll chromium_revision 7b9940f0b2..040c638bdb (1351213:1351313)
...
Change log: 7b9940f0b2..040c638bdb7b9940f0b2..040c638bdbc554716049..df5af89cb0bbde36b281..9f9e2bf58dd6ae301ab6..a1aafecb383a7d070966..62fc3a1d240dea786993..cd9d18ea6d7b9940f0b2..040c638bdbhttps://webrtc-review.googlesource.com/c/src/+/361702
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42958}
2024-09-05 10:41:18 +00:00
Björn Terelius
05e1a1f570
Explicitly set encoder_context to nullptr in SimulcastEncoderAdapter after move.
...
Bug: None
Change-Id: Idac2efa73835ff5e1c37e1f19d7d2127c62e5900
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361642
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42957}
2024-09-05 10:26:59 +00:00
Jakob Ivarsson
010c189f76
Move thread handling from source tracker.
...
This makes it simpler to use in more contexts.
Bug: b/364184684
Change-Id: I1b08ebd24e51ba1b3f85261eed503a78cd006fd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361480
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42956}
2024-09-05 08:45:11 +00:00
Markus Handell
2fbaa8e3a3
TaskQueueStdlib: Stop spamming on idle.
...
When the stdlib task queue has no work for >= 3s on
Android, it will emit bogus deadlock warnings. Fix
this by ensuring we're not triggering this behavior.
Fixed: b/364436627
Change-Id: I3fe02e13993cbb4a59d5669df06c4c293d237bc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361721
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42955}
2024-09-05 07:22:36 +00:00
Gavin Mak
3144e204ae
Make build_helpers.py work in the chromium/src superproject
...
The script can't find the `find_depot_tools` module if this repo is
checked out under chromium/src.
Bug: b:364177019
Change-Id: I75997a41e1a9732d0e9f6567b00c86784c7b4ce9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361680
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42954}
2024-09-05 07:11:49 +00:00
webrtc-version-updater
b2a2b1b51e
Update WebRTC code version (2024-09-05T04:05:51).
...
Bug: None
Change-Id: I573f99eb34c5219be17224a05d9b6073d679737b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361701
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42953}
2024-09-05 05:55:00 +00:00
chromium-webrtc-autoroll
e89abdc4e6
Roll chromium_revision 524f525d0b..7b9940f0b2 (1351076:1351213)
...
Change log: 524f525d0b..7b9940f0b2524f525d0b..7b9940f0b23037129f50..259b72d9aec89dfbb0db..bbde36b2817bfc50f825..225240b0c1574401153f..d6ae301ab6https://boringssl.googlesource.com/boringssl.git/+log/2fc7912e26..9224e6d138
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/c282fc3e7b..ee4a947642
* src/third_party/depot_tools: b89b440351..3a7d07096694c1ed8940..76963b6cb4cfb8da1e7a..0dea786993524f525d0b..7b9940f0b2https://webrtc-review.googlesource.com/c/src/+/361664
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42952}
2024-09-05 05:26:35 +00:00
chromium-webrtc-autoroll
55ca3c4b69
Roll chromium_revision 12d29901e2..524f525d0b (1350835:1351076)
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Change log: 12d29901e2..524f525d0b12d29901e2..524f525d0b5eefe82930..c554716049ff49fb0ea5..3037129f502aafafbda0..c89dfbb0db7880569b6d..7bfc50f8252adaba1099..574401153fhttps://boringssl.googlesource.com/boringssl.git/+log/6c3a63970f..2fc7912e26
* src/third_party/jdk/current: BXZwbslDFpYhPRuG8hBh2z7ApP36ZG-ZfkBWrkpnPl4C..sewpT0JmZAgFX_ZzmWhbYPmcPGeDa9os_4Q74ZFbo5sC
* src/third_party/perfetto: 09bfef4252..94c1ed8940138752d8b6..cfb8da1e7a12d29901e2..524f525d0bhttps://webrtc-review.googlesource.com/c/src/+/361661
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#42951}
2024-09-04 23:40:37 +00:00
Philipp Hancke
8480808c2f
ssl: remove SSL_set_read_ahead for DTLS mode
...
since it has no impact for DTLS:
https://docs.openssl.org/1.1.1/man3/SSL_CTX_set_read_ahead/
BUG=webrtc:353750117
Change-Id: I09d135aa69471af9d2feaf82e35625da3ae2e247
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361220
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: David Benjamin <davidben@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42950}
2024-09-04 23:25:15 +00:00
Danil Chapovalov
af8f6264ca
Use Environment instead of Clock in ModuleRtpRtcp2 and its RTP subcomponents
...
Bug: webrtc:362762208
Change-Id: Ie9bbb7f3b505acd8aab1b8552ba64e09a5a1bddf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361481
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42949}
2024-09-04 20:39:39 +00:00
chromium-webrtc-autoroll
d36041e9c0
Roll chromium_revision 621a1d6f77..12d29901e2 (1350725:1350835)
...
Change log: 621a1d6f77..12d29901e2621a1d6f77..12d29901e28685c65788..5eefe8293038e28b8045..ff49fb0ea5c0b937cf55..9c4d44a89acd3fd9a69c..2aafafbda0490a19ba19..7880569b6dc267e78c16..2adaba1099d12ed9caf2..b6405588e4c0a4867c02..09bfef42526e3c183093..138752d8b6621a1d6f77..12d29901e2https://webrtc-review.googlesource.com/c/src/+/361660
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#42948}
2024-09-04 18:32:16 +00:00
Fanny Linderborg
ac505c5b9a
Enable the FrameInstrumentationGenerator if its extension is negotiated
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Bug: webrtc:358039777
Change-Id: I5d1181d174e3e23506baa7f168849f02922311b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361580
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42947}
2024-09-04 15:48:00 +00:00
Björn Terelius
6e5eaea163
Roll chromium_revision 66e155a442..621a1d6f77 (1350363:1350725)
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Migrate turbine dep to cipd.
Change log: 66e155a442..621a1d6f7766e155a442..621a1d6f7777771bf80e..8685c65788896ecbfe9e..38e28b8045b8e3303855..cd3fd9a69c4477ada508..490a19ba19ec6251c1b2..c267e78c16https://chromium.googlesource.com/catapult.git/+log/663fc204a7..c282fc3e7b
* src/third_party/ffmpeg: 3f8a122bfa..468b9c8a52f2aa5c0df5..c0a4867c02059ff0d662..6e3c18309366e155a442..621a1d6f77https://webrtc-review.googlesource.com/c/src/+/361620
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42946}
2024-09-04 14:49:10 +00:00
Jonas Oreland
70a59b6900
Flip default value of AndroidNetworkMonitor field trials
...
getAllNetworksFromCache => true (stop using old Android API)
requestVPN => true (is default in old api)
They have been enabled using field trial
for more than a year.
Bug: webrtc:13741
Change-Id: I288c4067193e95251f79d51e935dce555f6eb198
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361581
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42945}
2024-09-04 11:20:34 +00:00
Björn Terelius
e540648079
Remove trailing semicolons in Java
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Bug: None
Change-Id: I79879691e47255dc2782ddcbfc80db777609a233
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361304
Commit-Queue: Zoé Lepaul <xalep@webrtc.org>
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42944}
2024-09-04 10:34:30 +00:00
Mirko Bonadei
2da07c8ebb
Update docs about supported platforms and compilers.
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Bug: b/364206287
Change-Id: I571bd149154ad017e15426abb9d0c4023f400998
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361600
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42943}
2024-09-04 10:30:16 +00:00
Florent Castelli
64d68c3984
Add WebRTC-MixedCodecSimulcast field trial
...
Disable the checks ensuring we reject mixed-codec simulcast
when the field trial is enabled.
The feature is however not yet implemented.
Bug: webrtc:362277533
Change-Id: Ib1601767c951d61aaa37a3d8767d0a81444d626c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361404
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42942}
2024-09-04 08:45:44 +00:00
Björn Terelius
4a7ea89ac9
Fix lint issues in logging/
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Bug: None
Change-Id: I998b3da0294bbf71f852c5947577622619c832d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361262
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42941}
2024-09-04 07:58:47 +00:00
Fanny Linderborg
dac0805955
Add FrameInstrumentationData to RTPVideoHeader and CodecSpecificInfo
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Bug: webrtc:358039777
Change-Id: If2659240047e1935f7666266bff25ed86a6a234c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361420
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42940}
2024-09-04 07:21:02 +00:00
yazdan0a
55a59337c8
Minor format to extrapolate local time
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- Removing unnecessary else {} blocks for better readability.
- Consistent naming of: timestamp_diff with explicit typecast.
BUG=None
Change-Id: I35161ffed245737c789336316f0cfb6821b12349
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361060
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42939}
2024-09-04 07:19:15 +00:00
Danil Chapovalov
45065a749d
Delete deprecated AudioDecoderFactory::MakeAudioDecoder
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Bug: webrtc:356878416
Change-Id: I672796e5ec749c3ae0141802922951d4fc562d78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359680
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42938}
2024-09-04 07:17:59 +00:00
webrtc-version-updater
40a038efa4
Update WebRTC code version (2024-09-04T04:08:21).
...
Bug: None
Change-Id: I46506e277c2eb489470e2fa24427915db4c0e0e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361560
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#42937}
2024-09-04 06:01:53 +00:00
chromium-webrtc-autoroll
ada1720db4
Roll chromium_revision 01d6daf051..66e155a442 (1350197:1350363)
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Change log: 01d6daf051..66e155a44201d6daf051..66e155a442b0531386e9..77771bf80e469c6d5062..896ecbfe9ed33351b991..c0b937cf55a891634072..b8e3303855c5261753f7..4477ada508https://chromium.googlesource.com/catapult.git/+log/21462dff93..663fc204a7
* src/third_party/perfetto: 4acdcdfaa0..f2aa5c0df5992e77c5a9..059ff0d66201d6daf051..66e155a442https://webrtc-review.googlesource.com/c/src/+/361541
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#42936}
2024-09-03 20:49:21 +00:00
Björn Terelius
0c2cd6264f
Fix lint issues in congestion_controller.
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Bug: None
Change-Id: I178c26044d5d678259181a9290c621bf64d3d7a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361361
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42935}
2024-09-03 17:55:07 +00:00
chromium-webrtc-autoroll
f8cb8b7854
Roll chromium_revision cae6b92cf5..01d6daf051 (1349874:1350197)
...
Change log: cae6b92cf5..01d6daf051cae6b92cf5..01d6daf0513b7d5e5523..b0531386e90811404e1e..d33351b991e5a4dbfef1..a891634072abf58fd834..c5261753f741333efead..ec6251c1b232eb84a959..a29e31cb00e9dbf36b50..d12ed9caf224eef196e1..4acdcdfaa0661b095f1f..992e77c5a9cae6b92cf5..01d6daf051https://webrtc-review.googlesource.com/c/src/+/361500
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#42934}
2024-09-03 17:19:40 +00:00
Harald Alvestrand
c17ca01f54
Move the payload type picker to call/
...
Since media/ and pc/ both have to use this, and both
depend on call/, this seems to be the right place to put it.
Also factor out the interface that media will use in a separate
interface class.
Bug: webrtc:360058654
Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42933}
2024-09-03 12:36:50 +00:00
