Because rtc::Thread inherits from TaskQueueBase, it already implements
a pair of PostTask/PostDelayedTask methods that we want to keep. But in
addition to those, rtc::Thread defines its own PostTask/PostDelayedTask
using templates. These are the versions that we want to deprecate.
They were originally implemented prior to rtc::Thread inheriting from
TaskQueueBase. We want to deprecate them because...
- We don't want to have multiple code paths that do the same thing.
- We want to move away from rtc::Thread to TaskQueueBase long-term.
- These versions are not overridable in Chromium.
- These versions don't have high/low precision versions of PDT.
Helper methods are added to rtc::Thread so that callers don't have to
wrap every lambda in webrtc::ToQueuedTask() and update dependencies.
Bug: webrtc:13582
Change-Id: I58702c53f4cb3705681bd9f1ea16b7aaa5052c18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247660
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35750}
As per go/postdelayedtask-precision-in-webrtc we want to reduce the
precision of PostDelayedTask() in order to schedule work on the CPU
more efficiently. In order not to break "high precision" use cases, a
new API is added to allow opting in to high precision.
PostDelayedHighPrecisionTask() has the same precision that
PostDelayedTask() has today, but by changing the interface's
requirements on PostDelayedTask(), adding the high precision version
of it will unblock making the old PostDelayedTask() API lower
precision.
This CL does not update implementations to support low precision so
until those are updated, both PostDelayedTask() and
PostDelayedHighPrecisionTask() have the same precision (=high).
This CL also adds TODOs to make some rtc::Thread-specific versions
of PostTask/PostDelayedTask obsolete, see
https://crbug.com/webrtc/13582 for more info.
Bug: webrtc:13583, webrtc:13582
Change-Id: I4c6d53d22bb299c49893ce9f3ef73a40d8c75de1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247367
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35748}
A number of utility functions to be shared between frame buffer 2
and the new frame scheduling implementation based on frame buffer 3.
Change-Id: Icc932c6c76fddeeedc8aa64ec27c7e0c955abfd0
Bug: webrtc:13343
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241604
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35743}
This CL updates both the static GOF pattern with the correct flags for
temporal_up_switch, as well the flexible mode logic to base the flag
on dependency descriptors instead use reference buffers.
Bug: webrtc:13576
Change-Id: I578f744bec51d1f3531da5f4a89d12f05a16a6c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247187
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35741}
Feedback can include lost packets. Ensure FeedbackGenerator support lost packets to be able to use this class properly when testing with loss.
Bug: none
Change-Id: Ibd740dfae358c0543fbee62cd40ef13a7ac1123f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247372
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35740}
...and one when destroying a channel object.
This CL removes Init_n() and Deinit_n() from the BaseChannel class.
Channel classes now use SetRtpTransport to do initialization and
uninitialization on the network thread.
Notably if an implementation has called SetRtpTransport() with a valid
transport pointer, it is required that SetRtpTransport be called again
with a nullptr before the channel object can be deleted.
In situations where multiple channels are created, this can mean
a substantial reduction in thread hops. We still hop to the worker
in order to construct the objects - this can probably be avoided
and SetChannel() is still a synchronous operation for the transceivers.
Furthermore, teardown of channel objects also still happens
synchronously and across network/worker/signaling threads.
Bug: webrtc:11992
Change-Id: I68ca7596e181fc82996e3e290733d97381aa5e78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246740
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35738}
Changing to an index for-loop (instead of using std::max_element & std::distance) tracking even & odd elements separately allows the compiler to produce code with less pipeline stall.
Bug: None
Change-Id: Iaa3e820a3a3b61e2eb276f0dac9106c848db1891
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240061
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Christian Schuldt <cschuldt@google.com>
Cr-Commit-Position: refs/heads/main@{#35729}
When screencast session is closed, there won't be any other stream we
can reconnect to and in that case we are supposed to disconnect our
stream to prevent accidentally connecting to any other stream in case it
gets assigned same node ID from PipeWire
Bug: webrtc:13429
Change-Id: Iec8e93a108c789c32cb93e1460e693fabc247491
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241086
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35728}
Minor clean up of BUILD file.
Add explicit events for begin and end of log.
Add a helper function to populate timestamps.
Add a GroupKey method that will be used for grouping events by for example SSRC in additon to event type.
Bug: webrtc:11933
Change-Id: Ie3c5f5a5582c89805a0273f4b27978f47ed0fb4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234260
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35725}
ChromeOS uses int64_t for its IDs (see display::Display::id()) so there is a potential for errors if casting (or attempting to hash and translate the larger ID to a smaller ID and vice versa).
Instead we should update the desktop_capture component to use int64_t natively.
Bug: webrtc:13571
Change-Id: I78b3456ce11b75755b90863a02f8c6455c63acf9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246240
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Jeroen Dhollander <jeroendh@google.com>
Commit-Queue: Joe Downing <joedow@google.com>
Cr-Commit-Position: refs/heads/main@{#35724}
WebRTC can switch encoder on-fly when encoder fails or by request from
encoder selector. Putting the current send codec to the front of the
codecs list provides a simple way for apps to know what is actually
used without retrieving stats.
Bug: webrtc:13572
Change-Id: Iaaa5f7ad8667f59016dc92bff9e9a57a7425ef44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246500
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35723}
The logged condition occurs very frequently (essentially log spam)
and has been ignored for a couple of years. So, logging it as a warning
seems unnecessary.
Bug: none
Change-Id: Ia7163c7a8e9af618685fa5655e90ee85f3cff87b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247366
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35721}
Remove advice to keep legacy style for consistency because it condridicts a recent addition to style guide goals
Remove debatable reason for banning std::shared_ptr, keep main reason
to ban it as the only one.
Bug: None
Change-Id: I4c31f9247a33430df0b286a15f1a35ab23060d1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247183
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35720}
Updates all webrtc code, to have a small followup cl to just add the
"explicit" keyword. Patchset #24 passed all webrtc tests, with explicit.
Bug: webrtc:13464
Change-Id: I39863d3752f73209b531120f66916dc9177bf63a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242363
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35718}
the new spelling is more standard and more compact, in particular doesn't need extra include and thus dependency
Bug: None
Change-Id: Iaea69d2154e4d9eff2468514f5734cb3fe016ff8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245080
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35709}
Remove Init_w and Deinit(), both of which were wrappers around Invoke()
calls from the worker thread to the network thread.
Instead, replace them with Init_n() and Deinit_n() that are currently*
required to be called by external code in order to associate/disassociate
the channels with the transport.
This CL mostly moves things around in order to prepare for upcoming
changes, but it does change channel destruction in the following way:
- When destroying channels, we don't block the worker thread anymore
while uninitialization happens on the network thread. Previously
both signal and worker threads were blocked during the
uninitialization in the ChannelManager.
* In an upcoming CL, Init_n() and Deinit_n() will be called internally
from a different method that's always called on the network thread
when a channel is associated/disassociated with a transceiver. When
we're there, we will have removed several invokes that currently are
a part of constructing/destructing channel objects.
Bug: webrtc:11992
Change-Id: Ibc30447a40749ceb36d37834b0cfc5c5ea60e895
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246502
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35707}
This is a simple check for upcoming changes for media channels to
be able to check if the state on the network thread is consistent.
Bug: webrtc:11992
Change-Id: I8ed2d091ecf3869a66970fc4733aebf209c4ef82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246681
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35706}
This is a necessary precondition for upcoming changes that will remove
calls to Invoke() that currently occur during construction/destruction
of media channel objects.
Subsequently fix RtpSenderReceiverTest to destroy channels that were
created in the constructor, in the destructor.
Bug: webrtc:11992
Change-Id: I526908d997d27495659805d84113c84c48568712
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246680
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35702}
