Only has an effect on Windows versions higher than 2104 (10.0.20348.0).
Bug: webrtc:15451
Change-Id: I3ca48c88a6c2b9b87d43805fcb2ade444cd90480
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318060
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40721}
This is a reland of commit a22c2a0c581cbe3f612f7a7d9fb9840186cc1e06
after systems depending on this have been fixed.
Original change's description:
> rtp sender: don't send BYE on deactivating streams
>
> as this breaks RTCP assumptions about SSRCs being no longer
> active as defined in
> https://www.rfc-editor.org/rfc/rfc3550#section-6.6
>
> This should not be sent in reaction to temporarily disabling
> a stream via RTCRtpParameters.active as this does not mean that
> the participant is leaving the session as defined in
> https://www.rfc-editor.org/rfc/rfc3550#section-6.3.7
> and does not indicate end of participation as defined in
> https://www.rfc-editor.org/rfc/rfc3550#section-6.1
> which stipulates BYE should be the last packet sent from this SSRC.
>
> BUG=webrtc:11082
>
> Change-Id: Ia5144857f85303643146b0759184f0f3f50b66e4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273348
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#38059}
Bug: webrtc:11082
Change-Id: Iad8b503b3101d1e684a4da2d1547b879e77b85dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293861
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@{#40716}
Structs make api harder to evolve:
deprecated unused properties,
change how data is represented.
Classes with accessors allow more graduated and safer api evolution.
Bug: None
Change-Id: I8ebd5e072d51cf7f5800666cfdac523d0f9a937f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40714}
These two functions contain complicated logic that will be used as
a fallback in Chromium if the new macOS picker code does not work
as intended.
Bug: chromium:1478172
Change-Id: I5f2878c5a8da38d59aa42ec1358398e3c921b65c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319260
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40711}
The main goal of this change is to disable the quality scaler when multiple spatial layers are used.
Bug: b/295129711
Change-Id: I25e0b7440a8c2adee3e97720a1e0ee5e0a914334
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319181
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40709}
Support the Pipewire videotransform meta via the already existing shared
infrastructure. This is needed for mobile devices which often have a 90
degree rotated camera - which is likely the reason there is already
support in the shared code paths.
Bug: webrtc:15464
Change-Id: I15223055d8675502ae326d270ebd2debbcfbfa50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318641
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40708}
This can be helpful in various situations, such as debugging with an
unrestricted Pipewire socket or for downstream projects like
B2G/Capyloon. Additionally it will help once we move from the camera
portal to the more generic device portal.
Original patch by Fabrice Desré <fabrice@desre.org>
Bug: webrtc:15464
Change-Id: Iae6802f242d68244bca85947cb15ef3eee923ab0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318642
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40706}
This reverts commit 2d71807fe09aad67efcd660fe286044ff10982ba.
Reason for revert: With the new AsyncAudioProcessing API, the issue that was introduced can now be worked around.
Original change's description:
> Revert "ConnectionContext: remove media engine without blocking."
>
> This reverts commit 2ba941e6bc1d20acb9cfda4b87ba53c80640bbcb.
>
> Reason for revert: Temporarily reverting due to b/269628432.
>
> Original change's description:
> > 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}
>
> Bug: webrtc:14449
> Change-Id: If2f23662e486a1c1f85c318fc98c441aab9ace31
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295862
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Markus Handell <handellm@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39454}
Bug: webrtc:14449
Change-Id: I43bb7a3b366eb60b3dc4b88dd9d47d570bb99bc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311941
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40705}
Also removing has_slots depdency from OpenSSLStreamAdapter and moving
it to the OpenSSLStreamAdapter subclass where it's still needed.
Bug: webrtc:11943
Change-Id: Ibcae5ea1efff146d78b32bb0eca63d7f44ed08c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318885
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40702}
move this a bit later in the process since the current handling will consider two ssrc-lines with a cname in the same RTX FID ssrc-group to be part of separate streams due to the different randomly assigned msids. This leads to a misdetection as plan-b SDP.
BUG=None
Change-Id: Ie8acce9c2c7fb9eabda479b90e8cc7406dcb1696
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318820
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@{#40701}
BitrateTracker uses same implementation as RateStatistics, but provides api using Timestamp and DataRate types instead of plain numbers
Bug: webrtc:13756
Change-Id: Ie37fa58ede7590f870ec4376a64e7cf2c94431d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318841
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40697}
Remove internal use of SignalSSLHandshakeError and prepare removal of
sigslot dependency from SSLStreamAdapter.
Bug: webrtc:11943
Change-Id: I9768e2e31529945620bdd8d0d285042bb2388b7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318881
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40695}
disallowing more than one ssrc-group with the same semantic
and primary ssrc.
BUG=chromium:1477075
Change-Id: I4bce0555cd49834725d9b97693d26c971bc5d5c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318822
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40694}
The code currently issues frames for encode before scheduling
a new repeat. Swap this order to account for time taken by for
slow encodes.
Bug: webrtc:15456
Change-Id: I74177069e30c1bf65268231ffba033411a0f7b9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318580
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40690}
Combine all parameters into single struct so that it is easier to add and remove optional parameters
Use Timestamp type instad of plain int to represent capture time
Use rtc::ArrayView instead of pointer+size to represent payload
Merge passing audio level into send function.
Bug: webrtc:13757, webrtc:14870
Change-Id: I0386b710eb99b864334d61235add9abcde9bc69d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317442
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40688}
We get this automatically from the //build checkout now
Bug: chromium:1432399
Change-Id: I223d7c5448244ed62821207068f979555617da57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318560
Auto-Submit: Chong Gu <chonggu@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/main@{#40686}
similar to what is done for FID and FEC-FR but SIM can have more than
one secondary SSRC.
BUG=chromium:1477075
Change-Id: I4c9b4feaa421f53e424fc17bfc9ee2c185c68fb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318520
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40679}
This constructor isn't used in production. Removing it further
made the construction state of the class simpler, allowed for removal
of the separate Init() method and making more members const.
Bug: none
Change-Id: Ibc8516a01ce7e385207251d841d21bb7b72c9d9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318281
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40678}
following the previous change to rename the classes derived from
cricket::RtpParameters
Also rename ChangedRecvParameters to ChangedReceiveParameters.
BUG=webrtc:13931
Change-Id: Ia51dd39905a5cbb98162c3948930e43ccaf3786d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314500
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@{#40677}
The synthetic delay added in ZeroHzAdapterMode::OnFrame does not
account for delay with respect to the initial frame post from
FrameCadenceAdapter::OnFrame. Fix this to account for time spent
in contention on the encode sequence.
Bug: webrtc:15456
Change-Id: I63446e8dfe8f62b09d972434a705e912f8a73d69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318420
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40675}
