which is displayed nicely in the git webview.
BUG=webrtc:11375
No-Try: True
Change-Id: I2dbe1ef0c74a0de8c5619b522fab39527e797d9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210681
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39545}
This presumably worked before because some libc++ header included it
transitively, but that's no longer the case.
Bug: chromium:1423839
Change-Id: I6ed1c3474c1bfa02084a665c0b9e249484ac50d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297420
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Hans Wennborg <hans@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39544}
Chromium has done this in 2021 (https://crrev.com/c/2405530).
This will allow the new BoringSSL to roll into WebRTC (since now
BoringSSL requires the NDK level to be >= 18).
NO_TRY=True
Bug: None
Change-Id: I4b722aef56cb1e189f4df6de64a87ab1e5b620a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297362
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39543}
Encoders that do not support simulcast in the first place does not
expect to have to handle simulcast configurations, and as such may not
necessarily return
WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED from
InitEncode().
This CL updates EncoderSimulcastProxy to respect this info to avoid
silent errors when LibvpxVp9Encoder (which does not support simulcast)
is attempted to be used in simulcast.
Alternatively we can try to get rid of EncoderSimulcastProxy altogether
since SimulcastEncoderAdapter already has a passthrough mode. A TODO is
added to get rid of the proxy.
Bug: webrtc:14884
Change-Id: Id3703f1768b0aebf617b7d9b935914cd5f1b0f52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296885
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39541}
Instead there are direct member variables for the various relevant
states, some weren't needed, some can be const but the `id` member
in particular needs special handling and can't be const.
For dealing with the stream id, we now have SctpSid. A class that does range validation, checks thread safety, handles the special `-1` case (for what's essentially an unsigned 16 bit int). Using a special type
for this also has the effect that range checking happens more
consistently (although I'm not modifying the structs in api/).
With upcoming steps of avoiding thread hops, the ID may need to
migrate to the network thread, which the thread checks will help with.
Along the way, update SctpSidAllocator to use flat_set instead of std::set and moving some of the sctp data channel code to the cc file
to help with more accurately tracking code coverage.
Bug: webrtc:11547
Change-Id: Iea6e7647ab8f93052044c5afbcc449115206b4e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296444
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39539}
This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.
No functional changes are intended.
Bug: webrtc:14876
Change-Id: Ieb6effd55f0ecba17cefc2f07f5eda1e85dbd016
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296441
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39535}
The test was assuming that after all thee layers have bytesSent > 0 we
would have fully ramped up to the expected resolutions. But there are
reasons why this may not be true, such as if adaptation kicks in.
This CL attempts to de-flake by using kLongTimeoutForRampUp when
checking the resolutions as well.
// Just increasing a timeout...
NOTRY=True
Bug: webrtc:14884
Change-Id: I5ef57ec3e3cc99552c9ae32a6fdf07889ff06ee1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296883
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39534}
It takes several seconds until we get an RTT measurement because that
requires RTCP packets to be received and those are not sent very often.
This CL makes the test faster on average by unblocking it as soon as
we see an RTT measurement (as opposed to always blocking for 10
seconds), this usually unblocks after around 5 seconds.
But to de-flake those rare instances where the test takes more than 10s
to run, the maximum timeout is extended to 20 seconds.
Patch Set 4: also fix use-of-uninitialized value.
Bug: webrtc:14981
Change-Id: Ieca94c90dfb52c3b17584a06660ff66c6462aa8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296822
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39531}
Instead, just use the posted task to release the reference to a
pending data channel object.
Bug: none
Change-Id: I34f0bfd604cab88587a892eaa218856c890fc907
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296767
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39527}
The width/height need to be updated in the VP9 SVC case since resolution
alignment may be applied inside GetSvcConfig(). This is not needed in
the VP9 simulcast case since we don't support simulcast + SVC combo and
resolution alignment is not needed for non-SVC.
This CL gates the "resolution update" behind
"numberOfSimulcastStreams == 1".
Bug: webrtc:14884
Change-Id: Ic3551721dcf6775fea6ff0c85fba48e88069fa5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296766
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39524}
https://bugs.chromium.org/p/webrtc/adminLabels has the field IDs for WebRTC.
Note that Priority and Issue type is required.
Bug: None
Change-Id: Ic088e6434452916169ed0da4a054c435e684e236
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296880
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39522}
This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.
No functional changes are intended.
Bug: webrtc:14876
Change-Id: Ie6820a820f22635fe7a970db70b9c28d37499848
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296443
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39518}
payload type frequency is not communicated inside an RTP packet and
thus do not belong to the RTPHeader
Bug: None
Change-Id: Ic3e48f1b0507a96ddc697503145f7c8785962926
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296763
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39515}
If a data channel object was closed (via calling Close()) right after
construction and before attaching to a transport, it would never
transition to the `kClosed` state. This addresses that corner case,
which caused a DCHECK to trigger but might also cause a situation
whereby more than one DC instance existed for a given sctp sid.
Bug: chromium:1421534
Change-Id: Id757c0528f929f2e2daa5343236d7f62e309f6cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296341
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39513}
This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.
No functional changes are intended.
Bug: webrtc:14876
Change-Id: Id1c7fbb969a63eee96fd88c376371aa7eafd0919
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296440
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39512}
following the updates from
https://github.com/w3c/webrtc-extensions/pull/142
BUG=chromium:1051821
Change-Id: I2d561bad1ddffb412bdd7e66cf62a3cb5fc73791
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296480
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39500}
to RTCInboundRtpStreamStats and RTCOutboundRtpStreamStats respectively
which follows the camel-casing convention used elsewhere.
The old name is kept around as an alias for a limited amount of time
to allow upgrading dependencies.
BUG=webrtc:14973
Change-Id: Ibf4e65933fd6cc2e7e89955042f6f8fb0f6c7853
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296261
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@{#39497}
