41124 Commits

Author SHA1 Message Date
Evan Shrubsole
b8abf5199a Remove TRACE_ASYNC without matching TRACE_BEGIN in rtc_sender_audio
This seems to confuse perfetto, and the data ends up on its own track
and the end event is just ignored. As it was invalid, I am assuming it
is not used, and can be simply removed.

#rtc_fixit


Bug: webrtc:15867
Change-Id: I77e59adcd35c51911474446a5f92505bf6b860f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342780
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41892}
2024-03-13 09:45:57 +00:00
Tommi
6417bbfd80 Change Port::Type() to IceCandidateType
Bug: webrtc:15846
Change-Id: Ibda55129f13d22ac84a730ba54d915c81a90cde9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340041
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41891}
2024-03-13 09:07:40 +00:00
Evan Shrubsole
9849bfdb10 Remove unused TRACE_*COPY* macros
#rtc_fixit

Bug: webrtc:15867
Change-Id: Id9198a5df4c4e5a4dace69cc8487b6ded40137ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342721
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41890}
2024-03-13 08:08:27 +00:00
webrtc-version-updater
c6e502e362 Update WebRTC code version (2024-03-13T04:03:28).
Bug: None
Change-Id: Ic4f600b3b1d2427bd56567718a20d791856c2323
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342840
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@{#41889}
2024-03-13 05:57:54 +00:00
Tim Na
4473d75651 Add TCP keep-alive options to rtc::Socket
Enabling Socket options on keep-alive related function that may enable clients to detect any stale connection early on.

Bug: webrtc:15866
Change-Id: Ib4f15e0c933aeb6cf4fd18ff8cc708d118ea8645
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342223
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41888}
2024-03-13 04:36:58 +00:00
Danil Chapovalov
f3096afd48 Propagate Environment to create VideoEncoder through java wrappers
Bug: webrtc:15860
Change-Id: If1a2873a899e1b839822a4b56aa87d4bae70c581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342740
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41887}
2024-03-12 15:34:12 +00:00
Harald Alvestrand
afaae4e38a Remove remaining .cc files from rtc_media_base
Also remove all dependencies on rtc_media_base except for a few
that are suspected of being linker directives.

Bug: webrtc:14775
Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 14:09:38 +00:00
Per K
8df31c915a Propagate ECN information on posix sockets to rtc::ReceivedPacket
Two new socket options are introduced OPT_SEND_ECN used for setting ECN bits. OPT_RECV_ECN used for reading the ECN bits.

If ECN bits are set on received IP packets,  ECT(1) and CE is propagated via rtc::ReceivedPacket.

Bug: webrtc:15368
Change-Id: I3ac335007e2f7d30564569bbc80ce47fa541bef1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332380
Reviewed-by: Jonas Oreland <jonaso@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41885}
2024-03-12 11:12:56 +00:00
Danil Chapovalov
329f0ead43 Provide Environment when creating VideoEncoder in test code
Bug: webrtc:15860
Change-Id: I8c79ff58619716842e02f33e78a0529c631494e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342280
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41884}
2024-03-12 11:09:31 +00:00
Oleh Prypin
a1d8665c31 Allow including internal-only tryjobs via a footer
Bug: None
Change-Id: I60728f0e07aca188dd2de9984795cc8cd2c7d5d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342700
Auto-Submit: Oleh Prypin <oprypin@google.com>
Commit-Queue: Oleh Prypin <oprypin@google.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41883}
2024-03-12 11:08:28 +00:00
Björn Terelius
1fc79ce4c4 Temporarily remove Linux MSan from LKGR
Bug: b/329130536
Change-Id: Iaa236db97ece69aa182b0f61a9c2966e241a0083
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342680
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41882}
2024-03-12 11:01:15 +00:00
Keiichi Enomoto
a70274a82f Remove duplicated parentheses from deprecated attribute
These lines cause an error when building a project with libwebrtc as a dependency in Microsoft Visual Studio.

Bug: webrtc:15864
Change-Id: I1abfe257d0ea1c16c4c5b718594e8085036f7763
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342320
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41881}
2024-03-12 10:58:59 +00:00
Victor Boivie
cd3d29b6fb pc: Simplify StreamId class
Before this CL, the StreamId class represented either a valid SCTP
stream ID, or "nothing", which means that it was a wrapped
absl::optional. Since created data channels don't have a SCTP stream ID
until it's known whether this peer will use odd or even numbers, the
"nothing" value was used for that state.

This unfortunately made it a bit hard to work with objects of this type,
as one always had to check if it contained a value. And even if a caller
would check this, and then pass the StreamId to a different function,
that function would have to do the check itself (often as a RTC_DCHECK)
since the passed StreamId always could have that state.

This CL simply extracts the "absl::optional" part of it, forcing holders
to wrap it in an optional type - when it can be "nothing". But allowing
the other code to just pass StreamId that can't be "nothing". That
simplifies the code a bit, potentially removing some bugs.

Bug: chromium:41221056
Change-Id: I93104cdd5d2f5fc1dbeb9d9dfc4cf361f11a9d68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342440
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41880}
2024-03-12 10:57:56 +00:00
Danil Chapovalov
b4913a549f Add factory functions to pass Environment to VideoEncoders
Bug: webrtc:15860
Change-Id: I4a9d2678dcfe5b0f178863242e27600fcc95325d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342480
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41879}
2024-03-12 09:43:14 +00:00
Jeremy Leconte
83d29d5988 Remove GetScalabilityMode2.
Change-Id: Ibe3162dbcaca31c3c22df0fdc8fe55b78ad7990b
Bug: b/327381318
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342400
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41878}
2024-03-12 09:20:48 +00:00
Björn Terelius
793add9dfb Temporarily remove linux_msan from cq
Bug: b/329130536
Change-Id: Id4933de9bbe98abf8e19e8418ce67cfe0a48eb49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342600
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41877}
2024-03-11 22:30:15 +00:00
webrtc-version-updater
0268a05fd0 Update WebRTC code version (2024-03-09T04:12:29).
Bug: None
Change-Id: Id1db760e67dbe31bc0aa8ee9c906151ca059c72e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342189
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@{#41876}
2024-03-09 06:06:08 +00:00
Tomas Gunnarsson
0242939296 Reland "Deprecate old constructors and set_type() in Candidate and Port"
This reverts commit ed8390d21a7b15091d01bc8e843193d0a6efd23a.

Reason for revert: Fix has landed in chrome, ready to reland.

Original change's description:
> Revert "Deprecate old constructors and set_type() in Candidate and Port"
>
> This reverts commit aaa6851d53741179a591d79fc82c4dd6651a7ba5.
>
> Reason for revert: breaks chromium webrtc import
>
> Original change's description:
> > Deprecate old constructors and set_type() in Candidate and Port
> >
> > * Deprecates constructors that use string based `type`
> > * Deprecates string based type functions in favor of enum based.
> > * Restrict possible values of Candidate::type. Ensure a valid value
> >   is assigned at construction.
> > * Make Port constructors protected to limit their use to subclasses.
> >   - The reason for this is to make sure that use of SharedSocket()
> >     is controlled (it adds a bit of complexity).
> > * Simplify construction of Port (remove Construct() etc)
> >
> > Bug: webrtc:15846
> > Change-Id: If24ed674e175642efa49da37fd2bc847dd14f613
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339860
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#41865}
>
> Bug: webrtc:15846
> Change-Id: Ic8b7cba97f8fb207ef51a88900e704658ade28b7
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342140
> Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#41867}

Bug: webrtc:15846
Change-Id: I3d52643bbb537d1c072643528828d26eb18fea94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342200
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@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@{#41875}
2024-03-08 20:39:59 +00:00
Johannes Kron
17e358096e Add AV1 encoder speed setting for screen share
There's an AV1 encoder speed setting 11 that is supposed to be used
for screen sharing content.

Bug: chromium:328598314
Change-Id: Id97898554a740eb1684d03c782c718c19f4c95e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342201
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41874}
2024-03-08 14:53:54 +00:00
Danil Chapovalov
9a9f6a8441 Add VideoEncoderFactory::Create to pass Environment for VideoEncoder construction
Bug: webrtc:15860
Change-Id: I6197780aaaa9c29717cb94df5790645b674c3bc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341900
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41873}
2024-03-08 11:46:39 +00:00
Victor Boivie
cd54fd8606 sctp: Pass webrtc::Environment to DcSctpTransport
The DcSctpTransport will soon use field trials to conditionally enable
some options.

And overall, there is a migration project to start using the Environment
and this CL is in that direction, also setting the boundary; The dcSCTP
library should not depend on it. But the transport is allowed to.

Bug: webrtc:14997
Change-Id: I1f3c2c0d8dd7bdc698dd1d58bde7651b682bcba4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341480
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41872}
2024-03-08 09:45:12 +00:00
webrtc-version-updater
4c1c9157d6 Update WebRTC code version (2024-03-08T04:01:32).
Bug: None
Change-Id: I50fb78e58bfe03670bef74d7fa5adff6664a447e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342184
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@{#41871}
2024-03-08 05:33:16 +00:00
Jeremy Leconte
51f98ccb5d Prepare the removal of GetScalabilityMode2.
Change-Id: I4b41fd1faee0e27b2b05842d7825b6b0785735ec
Bug: b/327381318
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341600
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#41870}
2024-03-07 17:57:16 +00:00
Bjorn Terelius
b41f07bc51 Explicitly initialize the SctpTransportState to kNew
Bug: webrtc:15814
Change-Id: I94325979777741a2798e1bfac3474bcc364592bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341020
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41869}
2024-03-07 14:27:35 +00:00
Danil Chapovalov
d055f77276 Delete legacy name AudioLevel in favor of the AudioLevelExtension
AudioLevel name was deprecated two weeks ago.

Bug: webrtc:15788
Change-Id: Idb26ab6ea701bcbceeda51640d521b78fa0d8162
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341264
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41868}
2024-03-07 12:49:27 +00:00
Ilya Nikolaevskiy
ed8390d21a Revert "Deprecate old constructors and set_type() in Candidate and Port"
This reverts commit aaa6851d53741179a591d79fc82c4dd6651a7ba5.

Reason for revert: breaks chromium webrtc import

Original change's description:
> Deprecate old constructors and set_type() in Candidate and Port
>
> * Deprecates constructors that use string based `type`
> * Deprecates string based type functions in favor of enum based.
> * Restrict possible values of Candidate::type. Ensure a valid value
>   is assigned at construction.
> * Make Port constructors protected to limit their use to subclasses.
>   - The reason for this is to make sure that use of SharedSocket()
>     is controlled (it adds a bit of complexity).
> * Simplify construction of Port (remove Construct() etc)
>
> Bug: webrtc:15846
> Change-Id: If24ed674e175642efa49da37fd2bc847dd14f613
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339860
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41865}

Bug: webrtc:15846
Change-Id: Ic8b7cba97f8fb207ef51a88900e704658ade28b7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342140
Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41867}
2024-03-07 09:43:38 +00:00
webrtc-version-updater
dd39c03feb Update WebRTC code version (2024-03-07T04:13:24).
Bug: None
Change-Id: I45ef8f031bccbd77fcf3325640844522a794ebc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341992
Commit-Queue: 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@{#41866}
2024-03-07 06:02:40 +00:00
Tommi
aaa6851d53 Deprecate old constructors and set_type() in Candidate and Port
* Deprecates constructors that use string based `type`
* Deprecates string based type functions in favor of enum based.
* Restrict possible values of Candidate::type. Ensure a valid value
  is assigned at construction.
* Make Port constructors protected to limit their use to subclasses.
  - The reason for this is to make sure that use of SharedSocket()
    is controlled (it adds a bit of complexity).
* Simplify construction of Port (remove Construct() etc)

Bug: webrtc:15846
Change-Id: If24ed674e175642efa49da37fd2bc847dd14f613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41865}
2024-03-06 18:36:14 +00:00
Danil Chapovalov
ac2720e967 Remove unnecessary RtcEventLog parameter in RtpTransportControllerSend::CreateRtpVideoSender
RtpTransportControllerSend has access to the same Environment as the caller, and thus can take RtcEventLog directly from it.

Bug: None
Change-Id: I4b20811d3f6de8193c63d6c58d0fe1204b3ec7b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342040
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41864}
2024-03-06 16:24:06 +00:00
philipel
5ace0710bf Remove unused PacketOptions::additional_data.
Bug: none
Change-Id: I642ad5fde070d7c9c708d99ec9a91b28e294d11e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341960
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41863}
2024-03-06 11:17:52 +00:00
webrtc-version-updater
36e38757d7 Update WebRTC code version (2024-03-06T04:06:44).
Bug: None
Change-Id: I078afc8ce2c168f484ecec58e1b578b637c73870
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341985
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2024-03-06 05:28:42 +00:00
Danil Chapovalov
c9bb2c6c4e Propagate Environment into VideoStreamEncoder
VideoStreamEncoder creates VideoEncoders. To pass an Environment to VideoEncoder, it should be available in the VideoStreamEncoder.

Bug: webrtc:15860
Change-Id: Id89ac024ce61fdd9673bb66f03f94f243fc0c7f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341840
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41861}
2024-03-05 09:33:02 +00:00
Christoffer Dewerin
9f11b96e6b add xctest to gn args for ios sim
Bug: webrtc:14786
Change-Id: I293835eb33ee0304930985ba44442bb0c60ce74e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341841
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Christoffer Dewerin <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41860}
2024-03-05 08:37:00 +00:00
webrtc-version-updater
cebded9b54 Update WebRTC code version (2024-03-05T04:11:56).
Bug: None
Change-Id: I94df4ac41dfc0d1f8b0bd44ca69db536fbbb33c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341881
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@{#41859}
2024-03-05 05:55:17 +00:00
chromium-webrtc-autoroll
e166214cba Roll chromium_revision d6b7dad43f..b9338390df (1267774:1267934)
Change log: d6b7dad43f..b9338390df
Full diff: d6b7dad43f..b9338390df

Changed dependencies
* src/base: a648098fc5..57610ea6da
* src/build: a20111f3fd..d48ea92a42
* src/buildtools: 1db15eb420..9491ff1efc
* src/ios: 44a1b90ebb..e1f09315ee
* src/testing: 0040b2b278..1ada31861f
* src/third_party: 2ed07aa758..3db9b0ba6d
* src/third_party/androidx: X795kcd7b3VobEty5e4NWY4grh5PlCvRCPnyt-cXV3AC..GWbo7p3_LfXNsOnuuQIP6VWA9aJ8YP6czcHvgqhAfxAC
* src/third_party/depot_tools: fbb0301f1f..875647ed03
* src/third_party/googletest/src: dda72ef321..e4fdb87e76
* src/third_party/libc++/src: b5fe27de93..80307e66e7
* src/third_party/perfetto: 22d2e541be..3fe34e7c3e
* src/third_party/re2/src: 2d866a3d07..45c9985092
* src/third_party/turbine: ZsrSMKpQt5d43K50XC1both1bFWzoloH6xOKYKZK_64C..RmqZxX5J0fjQAxIVGLBnWAsmcU_2_bfgH85YgcNv6lAC
* src/tools: a47f932da8..fd6f55bb24
DEPS diff: d6b7dad43f..b9338390df/DEPS

No update to Clang.

BUG=None

Change-Id: Ie3492fae4116878b1a8c208d5e8087cc8e7ee533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341821
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@{#41858}
2024-03-04 19:27:48 +00:00
Danil Chapovalov
38c1ab1e6c Delete CreateVideoDecoder from VideoDecoderFactory interface
Instead require Create to be implemented

Bug: webrtc:15791
Change-Id: I17477b5f047d86b6a05bda594c66d20f8f43a2c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340480
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41857}
2024-03-04 16:05:51 +00:00
chromium-webrtc-autoroll
80d07289fd Roll chromium_revision 67f77562a2..d6b7dad43f (1267659:1267774)
Change log: 67f77562a2..d6b7dad43f
Full diff: 67f77562a2..d6b7dad43f

Changed dependencies
* src/base: 81977015e5..a648098fc5
* src/build: c06d7b5cb4..a20111f3fd
* src/testing: ab0ead57af..0040b2b278
* src/third_party: 174a3b4a8b..2ed07aa758
* src/third_party/androidx: -hKL4aNs2f-WxaYX42KZQqg7ytafBADY8TVVzhUQtVAC..X795kcd7b3VobEty5e4NWY4grh5PlCvRCPnyt-cXV3AC
* src/tools: 9cc615980b..a47f932da8
DEPS diff: 67f77562a2..d6b7dad43f/DEPS

No update to Clang.

BUG=None

Change-Id: Ibade1c8c97a73618976a3282ef56543e35ca119f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341769
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@{#41856}
2024-03-04 12:55:07 +00:00
webrtc-version-updater
206bdaf26c Update WebRTC code version (2024-03-04T04:13:18).
Bug: None
Change-Id: Ie6d93e49c7ea04ab5f80ea6c17168919a2ab753f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341767
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@{#41855}
2024-03-04 05:20:38 +00:00
chromium-webrtc-autoroll
04d22681e3 Roll chromium_revision 0bfdc8c539..67f77562a2 (1267549:1267659)
Change log: 0bfdc8c539..67f77562a2
Full diff: 0bfdc8c539..67f77562a2

Changed dependencies
* src/build: 3915ccffa2..c06d7b5cb4
* src/testing: e2900fac8e..ab0ead57af
* src/third_party: e61acf937c..174a3b4a8b
* src/tools: 93a213c07f..9cc615980b
DEPS diff: 0bfdc8c539..67f77562a2/DEPS

No update to Clang.

BUG=None

Change-Id: I299e1333be123ba8182140135ed4c52dcdb347b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341785
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@{#41854}
2024-03-04 02:45:22 +00:00
webrtc-version-updater
89e62f305a Update WebRTC code version (2024-03-03T04:12:48).
Bug: None
Change-Id: I2fd7942657d24718c1baf8bac89ce4211a56cf55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341760
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@{#41853}
2024-03-03 05:46:17 +00:00
chromium-webrtc-autoroll
c8068f68f2 Roll chromium_revision 16ca06c8c8..0bfdc8c539 (1267445:1267549)
Change log: 16ca06c8c8..0bfdc8c539
Full diff: 16ca06c8c8..0bfdc8c539

Changed dependencies
* src/base: fef2b5e6b7..81977015e5
* src/testing: 410689e90a..e2900fac8e
* src/third_party: 336e6a4e68..e61acf937c
* src/third_party/perfetto: 98921c2a0c..22d2e541be
* src/tools: 546c584d90..93a213c07f
DEPS diff: 16ca06c8c8..0bfdc8c539/DEPS

No update to Clang.

BUG=None

Change-Id: I277fe1cdf9fb8f5398a04689662725ed65496869
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341697
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@{#41852}
2024-03-02 19:38:37 +00:00
webrtc-version-updater
77590862d5 Update WebRTC code version (2024-03-02T04:12:36).
Bug: None
Change-Id: I3cb435804e4510ccc7f15b45853faf212a911299
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341690
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@{#41851}
2024-03-02 05:58:58 +00:00
chromium-webrtc-autoroll
572ce2719c Roll chromium_revision 6312fa2472..16ca06c8c8 (1267340:1267445)
Change log: 6312fa2472..16ca06c8c8
Full diff: 6312fa2472..16ca06c8c8

Changed dependencies
* src/build: 0f6697fc2b..3915ccffa2
* src/testing: 54f2661b52..410689e90a
* src/third_party: 7563c75d12..336e6a4e68
* src/third_party/android_build_tools/manifest_merger: ebz_Y3LqXzAa7YSsVInCAghbwoZuC4tySvJ1XPJLCzIC..bmxKmBbioYv3d9nmRIo_xYGXwobb91K5RM7xU0RUQu4C
* src/third_party/androidx: iX0cDzVg1LYwl-VFNJPfNgZUPK5RCN7PUW7VBxtqx_8C..-hKL4aNs2f-WxaYX42KZQqg7ytafBADY8TVVzhUQtVAC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2ee42b4fee..f226e76aa5
* src/third_party/perfetto: 77ac4b7528..98921c2a0c
* src/third_party/r8: XyJJ5GEKJUXldBnoKKraiUIjSbnXGqjNBcLoNuJvKccC..dlcpQz73JQc8czs_ASn1itNoISc9wNEMBb5YTvTyQtEC
* src/tools: 158705d708..546c584d90
DEPS diff: 6312fa2472..16ca06c8c8/DEPS

No update to Clang.

BUG=None

Change-Id: Ie53ae124eebae9dad716ebc8c448c484a7015873
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341702
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@{#41850}
2024-03-01 19:36:23 +00:00
chromium-webrtc-autoroll
59ed9e8ebe Roll chromium_revision 38dcc53cf5..6312fa2472 (1267235:1267340)
Change log: 38dcc53cf5..6312fa2472
Full diff: 38dcc53cf5..6312fa2472

Changed dependencies
* src/base: fa26aeb00d..fef2b5e6b7
* src/build: b484740dba..0f6697fc2b
* src/ios: 9ec2be606c..44a1b90ebb
* src/testing: 5d3c6792d9..54f2661b52
* src/third_party: bf93900a20..7563c75d12
* src/third_party/depot_tools: 1ac3eb7b98..fbb0301f1f
* src/third_party/googletest/src: 76bb2afb8b..dda72ef321
* src/tools: 00e519d947..158705d708
DEPS diff: 38dcc53cf5..6312fa2472/DEPS

No update to Clang.

BUG=None

Change-Id: Ied69ca463f61f945e14e55ef2987dd94574a2940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341623
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@{#41849}
2024-03-01 02:50:18 +00:00
chromium-webrtc-autoroll
ede75295d4 Roll chromium_revision 1e40594b88..38dcc53cf5 (1267092:1267235)
Change log: 1e40594b88..38dcc53cf5
Full diff: 1e40594b88..38dcc53cf5

Changed dependencies
* src/base: 3d0b3c7162..fa26aeb00d
* src/buildtools/linux64: git_revision:e05c0aa00938adc0797bda1e8f2c15675aa13c30..git_revision:88e8054aff7bd0cb2295c7d9361d2be0b7355f27
* src/buildtools/mac: git_revision:e05c0aa00938adc0797bda1e8f2c15675aa13c30..git_revision:88e8054aff7bd0cb2295c7d9361d2be0b7355f27
* src/buildtools/win: git_revision:e05c0aa00938adc0797bda1e8f2c15675aa13c30..git_revision:88e8054aff7bd0cb2295c7d9361d2be0b7355f27
* src/ios: d49d4c013b..9ec2be606c
* src/testing: 9a0f787478..5d3c6792d9
* src/third_party: 35ff337157..bf93900a20
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/a2d599c975..14010c6f0f
* src/third_party/perfetto: 3fa1408bbc..77ac4b7528
* src/tools: d7f2f98a48..00e519d947
DEPS diff: 1e40594b88..38dcc53cf5/DEPS

No update to Clang.

BUG=None

Change-Id: Id10a86b8cb0e3259ffe17c54ad390477bcafe168
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341660
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@{#41848}
2024-02-29 22:52:39 +00:00
chromium-webrtc-autoroll
e5ac106a35 Roll chromium_revision f770766245..1e40594b88 (1266950:1267092)
Change log: f770766245..1e40594b88
Full diff: f770766245..1e40594b88

Changed dependencies
* src/base: 9f13d878d5..3d0b3c7162
* src/ios: f81fcc51c8..d49d4c013b
* src/testing: 8d2ca7caa0..9a0f787478
* src/third_party: afe1d14b38..35ff337157
* src/third_party/androidx: rTiFKohCdnT81G3SjzFlb536YE6DnBkp_3Ig-Pt7gCUC..iX0cDzVg1LYwl-VFNJPfNgZUPK5RCN7PUW7VBxtqx_8C
* src/third_party/freetype/src: 546237e1bb..2a790a9f49
* src/third_party/perfetto: 1553701a9f..3fa1408bbc
DEPS diff: f770766245..1e40594b88/DEPS

No update to Clang.

BUG=None

Change-Id: I3e5349ea1552435c33f588e32484956690a40114
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341622
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@{#41847}
2024-02-29 18:35:05 +00:00
chromium-webrtc-autoroll
015de612e9 Roll chromium_revision 248b5659e1..f770766245 (1266836:1266950)
Change log: 248b5659e1..f770766245
Full diff: 248b5659e1..f770766245

Changed dependencies
* src/base: 7dfbdde7b6..9f13d878d5
* src/build: 100a65f1dd..b484740dba
* src/ios: 53ae48db44..f81fcc51c8
* src/testing: 6ac6be6e29..8d2ca7caa0
* src/third_party: dd5b48d517..afe1d14b38
* src/third_party/androidx: Qdbpp4CESrciZ3ZF1ZZmOg-NQSUdK-DkNAddEJeZbbgC..rTiFKohCdnT81G3SjzFlb536YE6DnBkp_3Ig-Pt7gCUC
* src/third_party/perfetto: 609cb8ef02..1553701a9f
* src/third_party/re2/src: f9550c3f72..2d866a3d07
* src/tools: 76e998060b..d7f2f98a48
DEPS diff: 248b5659e1..f770766245/DEPS

No update to Clang.

BUG=None

Change-Id: I67c28310e8d3c2319bc0c991bd5af769c3189c9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341549
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@{#41846}
2024-02-29 14:31:56 +00:00
Philipp Hancke
a5cd6643f6 Add killswitch for receive-only setCodecPreferences change
Adds a killswitch
  WebRTC-SetCodecPreferences-ReceiveOnlyFilterInsteadOfThrow
to accompany the spec-change to throw when codec capabilities
are taken from the RtpSender instead of the RtpReceiver.
With the killswitch triggered, such codecs will be filtered.

BUG=webrtc:15396

Change-Id: I7d27111c72085eb7a7b2a1e66d0a08d12883ce17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341460
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41845}
2024-02-29 12:43:05 +00:00
Jan Grulich
16ac10d9f7 PipeWire camera: use length of device id instead display name
We want to copy device id to _lastUsedDeviceName variable, but we use
length of display name instead of length of device id, which might be
longer than expected and we end up reading beyond the source string.

Bug: webrtc:15853
Change-Id: Id278ed7e361ead85475910adec18b9db51e6890b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341521
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41844}
2024-02-29 10:20:09 +00:00
Philipp Hancke
20a90295fc sdp: set content to rejected if the list of common codecs is empty
which avoids throwing an error when using setCodecPreferences
to set a recvonly codec on a sendonly transceiver. See
  https://github.com/w3c/webrtc-pc/issues/2936

BUG=webrtc:15396

Change-Id: I435a98c944ed2eeef87d9b8a7f791d095ec25502
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338642
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@{#41843}
2024-02-29 08:02:26 +00:00