This tests all existing codecs (VP8, VP9) with the depdendency
descriptor and adds the AV1 tests that requires it as well.
Placeholders for missing modes have been added for both VP9 and AV1.
Bug: webrtc:11607
Change-Id: Ie900bddc54ccbf4dcc466f3a7a6c8241906a243a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272807
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37906}
This simplifies the logic of what simulcast layers to signal, and avoids
situations where the upper layers get confused about which layers exist.
Bug: chromium:1350245
Change-Id: I9edeb93cbb30e872c4d3f3429a85a1fccf17996a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272902
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37905}
And add a unit test that verifies that next probe time is set at is
the expected if the recommended probe size is used.
Bug: webrtc:14392
Change-Id: I239bb3a1c8eefc85509aacc82037c64e3ce49ed7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272648
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37901}
The libyuv fix for this issue was submitted > 2 months ago so it
should be safe to remove the workaround from webrtc.
Bug: chromium:1330019
Change-Id: Ibcf3818739673005e40d7ef9917c5f5692c50df4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272861
Commit-Queue: Joe Downing <joedow@google.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37895}
rtc::Thread::Dispose is only used in test code,
but complicates the main thread loop.
Bug: webrtc:8324
Change-Id: I2dccdadcdc932b9992958d1e70fb93d1879b7618
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272821
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37894}
I would like to run a separate capturer for each desktop on Linux and
I ran into the DCHECK in XErrorTrap when I was prototyping that
solution. I addressed it by using a Mutex and then experienced and
occasional hang when capturing which I traced down to
SharedXDisplay::ProcessPendingXEvents(), this is a shared display
instance used by each unique capturer instance so I added a mutex
there as well.
I ran 2 capturer instances concurrently for well over an hour and did
not experience any hangs or capture artifacts.
Bug: webrtc:2022
Change-Id: Ia6778cae4bbae48886fe45f2991f02e0ea08fef6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271920
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Joe Downing <joedow@google.com>
Cr-Commit-Position: refs/heads/main@{#37892}
Remote Peek function as unused
Move Get and Dispatch into private section to ensure they are not used
from outside.
Bug: webrtc:9702
Change-Id: Ibd0b236fe43543d60f97f988524526493bbeaaa7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272804
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37889}
Fix reporting of time_between_freezes_ms when all peers were
unregistered before end of the call.
Bug: b/243501613
Change-Id: Ie2b77d8399e4f4bf672e80f62ed27609336af171
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272650
Reviewed-by: Andrey Logvin <landrey@google.com>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37885}
In rare cases, it is possible to queue a call to SendData from the signaling
thread on a channel being closed or already closed in the network thread.
By keeping track of currently open streams, we avoid sending messages
with a stream id of channels that the other side already considers closed
and has already reused for a new channel.
This caused rare messages to be delivered on the wrong data channel if
a message was quickly sent, channel closed and a new one reopened.
Bug: webrtc:14277
Change-Id: If35fed8d12d5d2c18cdc6601085d8b632c37a0ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272624
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37880}
This reverts commit bc8a62b244c4038d2a005bffcbe7a82bd933d7d1.
Reason for revert: reverting per
https://bugs.chromium.org/p/webrtc/issues/detail?id=14368#c5
This needs more careful consideration and should be put behind a finch flag or origin trial
Original change's description:
> Enable Multithreaded H264 Encoding For OpenH264
>
> Re-enabled multithreaded encoding using OpenH264, as the issue described in crbug.com/583348 no longer applies.
>
> Bug: webrtc:14368
> Change-Id: I5ae768a6edf3b40d99c13fb4ee4662626c993a66
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271820
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37837}
Bug: webrtc:14368
Change-Id: Icebedfe4eb8e3901670b9f90e229379fca95206b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272600
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@{#37878}
The default implementation of CropAndScale uses ToI420() and then Scale,
and this implementation behaves inefficiently with RTCCVPixelBuffer.
Bug: None
Change-Id: I422ef80d124db0354a2e696892e882a78db445bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271140
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Auto-Submit: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37877}
To allow the transport to be able to know which ranges of
stream identifiers it can use, the negotiated incoming/inbound
and outgoing/outbound stream counts will be exposed. They are
added to Metrics, and guaranteed to be available from within
the OnConnected callback.
In this CL, dcSCTP will not validate that the client is sending
on a stream that is within the negotiated bounds. That will be
done as a follow-up CL.
Bug: webrtc:14277
Change-Id: Ic764e5f93f53d55633ee547df86246022f4097cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272321
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37876}
In case ScreenCast portal fails right at the beginning, we need to check
the response before trying to get session handle to avoid accessing
non-existing portal data.
Also on early failure do not continue making source request if we failed
before and don't have session handle.
Bug: webrtc:13429
Change-Id: I2bfbd2c6e96e3cda1e62aa9dc07f66d4c7496b53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272400
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37872}
callback are know at construction time and only need some synchronization at destruction time. In this case such synchronization can be done with cheaper/simpler WeakPtr concept.
Asynchronous call to SetCertificate is no longer needed thanks to
previous removal of sigslot in
https://webrtc-review.googlesource.com/c/src/+/192362
Bug: webrtc:11943
Change-Id: Icadbcb4f83be9ed4b8f53a72beaef8573f2c9356
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272402
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37868}
This moves the ownership away from VideoReceiveStream2 and closer to
VCMDecoderDataBase. That facilitates unregistration (upcoming change)
without recreating receive streams.
Bug: none
Change-Id: I812175134730a0ffbf7077fd149c8489481c73d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272481
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37866}
To allow the transport to be able to know which ranges of
stream identifiers it can be use, the negotiated incoming/inbound
and outgoing/outbound stream counts will be exposed.
This is first added to handover state, with the actual implementation
to follow.
Bug: webrtc:14277
Change-Id: Idd821ecbd8fcb588c88d69f617889318b4b03d43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272320
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37863}
