Ensure initial second probe can be disabled.
Can configure separate probe duration if the network state estimate is known.
Can probe immediately if network state estimate increase more than a factor
Bug: webrtc:14392
Change-Id: Iefb980f0b10c7c51db62793c3bd3f187fc67593d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273349
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37966}
This reverts commit fbea8c519684577a38cb35b9287ba4645a905094.
Reason for revert: Breaks WebRTC import into Chromium, e.g:
https://chromium-review.googlesource.com/c/chromium/src/+/3863998/
Original change's description:
> Add plumbing to control PipeWire picker visibility
>
> Introduces the notion of a "delegated source list" and corresponding
> controller. This is used by desktop capturers (currently just the
> PipeWire capturer), who control selecting the source through their own
> (often system-level) UI, rather than returning a source list with all
> available options that can then be selected by the embedder.
>
> Adds a method to get the controller which serves to also tell embedders
> if the capturer makes use of a delegated source list. The controller
> currently allows the embedder to request that the delegated source list
> be shown or hidden, and will in the future be used to expose events
> from the source list (e.g. selection, dismissal, error).
>
> Bug: chromium:1351572
> Change-Id: Ie1d36ed654013f59b8d9095deef01a4705fd5bde
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272621
> Reviewed-by: Mark Foltz <mfoltz@chromium.org>
> Commit-Queue: Alexander Cooper <alcooper@chromium.org>
> Cr-Commit-Position: refs/heads/main@{#37956}
Bug: chromium:1351572
Change-Id: I06f76ab9c8bc1aa303dae177d48698951fdc5ecd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273703
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37964}
We might be potentially blocking PipeWire initialization with call to
pw_thread_loop_wait() and waiting undefinitely for response in case
there is a fatal error.
Bug: webrtc:13429
Change-Id: If169e04f75a7d24a03a0fcd0da9ffaba8c0e2ef7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273481
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37959}
Introduces the notion of a "delegated source list" and corresponding
controller. This is used by desktop capturers (currently just the
PipeWire capturer), who control selecting the source through their own
(often system-level) UI, rather than returning a source list with all
available options that can then be selected by the embedder.
Adds a method to get the controller which serves to also tell embedders
if the capturer makes use of a delegated source list. The controller
currently allows the embedder to request that the delegated source list
be shown or hidden, and will in the future be used to expose events
from the source list (e.g. selection, dismissal, error).
Bug: chromium:1351572
Change-Id: Ie1d36ed654013f59b8d9095deef01a4705fd5bde
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272621
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37956}
Those never existed, were likely a copy-paste error in the spec
that we somehow inherited.
Bug: webrtc:11607
Change-Id: Ib4a038f061123e879f1099656273f6392f092213
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273485
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37953}
Replace most instances. SetAlrStartTime is set as is should be cleaned up together with the callsite.
Bug: webrtc:14404
Change-Id: I8ec532828ef665afbf08f0943465a429ab40baa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273300
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37932}
This adds a (default off) flag which makes retransmissions be processed
immediately, just like audio packets normally are.
This might increase send rates and thus losses in some cases, but will
also reduce retranmission delays especially when timer slack or bursting
is used. Usefuleness TBD via experiment.
Bug: chromium:1354491
Change-Id: Icaa83125bfb30826ce72e6e786963d411e05ea57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272483
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37926}
Add field trial to not probe if loss based limited
If both Alr probing and periodic probing of networkstate estimate is enabled, probes are limited by the network state estimate * factor controlled by field trial.
Bug: webrtc:14392
Change-Id: I46e1dbdd8b14f63a7c223b4c03c114717b802023
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272805
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37915}
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}
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}
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}
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}
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}
This experiment will tell if we still see the performance gains that we
saw with the "bursty slacked pacer" even if we don't apply slack (since
the "slack without burst" showed little impact at Stable).
The hope is that without slack all quality regressions will go away but
that bursting will still provide the desired performance benefits.
Bug: chromium:1354491
Change-Id: I95f05d040713addaaa1856c8e374a01c27311612
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272366
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37845}
For now use doubles as units in api/units have insufficient precision for jitter estimation.
Bug: webrtc:14381
Change-Id: I5a691b6a404b734a5bef11d677b72040bc02ff0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272367
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37841}
Now Configure(), Decode() and Release() calls to the decoders should
all happen on the decoder thread. Added thread checkers to verify.
Bug: None
Change-Id: I2a1cf2cf7f3c3c7c50e382d82a3638e916ed9c34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272368
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37840}
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}
This was only set to nullptr in non-test environments and was thusly
unused. With this change, the stats callbacks are gaurenteed to only
come from the VideoStreamBufferController and so the thread checks can
be removed.
Bug: webrtc:14003
Change-Id: Iaf0e77aa7c45a317e38ae27739edcefd3123d832
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272021
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37816}
I've added the proper headers to the only file in Chromium which includes screen_capture_frame_queue.h (see https://chromium-review.googlesource.com/c/chromium/src/+/3836317).
I've also built the remoting host and Chrome on Windows and Linux with this change and did not see any build errors.
The only build error I encountered was in shared_screencast_stream when building webrtc so I added the required header there.
Bug: webrtc:14378
Change-Id: Ie88e606dfa52f18514a87b87e5904424543d7df3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271922
Commit-Queue: Joe Downing <joedow@google.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37811}
Moves FrameBuffer2 to its own GN target to reduce the binary size of the
video target.
Bug: None
Change-Id: I40e86a1eabc0c9e8e6fada3dcdb4e3a043c61c6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271286
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37803}
- Move existing check on `max_frame_size` to the top.
By doing this early, the filter will not end up in an
inconsistent state (predicted but not updated) when
called with a tiny `max_frame_size`.
- Add sanity check on noise variance.
This will avoid sqrt of a negative number.
Bug: webrtc:14151
Change-Id: I2507a9114655d3c3ae35bf5ed83f3f1154c42ad3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271281
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37798}
In migrating rtc::Event to use TimeDelta instead of int,
rtc::Event::kForever will have to become something else.
This change removes dependencies on that kForever is int.
Bug: webrtc:14366
Change-Id: Ic36057dda95513349e7ae60204e7271ff1f58825
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271288
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37795}
This CL reorders the update steps physically, to make them
easier to comprehend. It renames variables to be more verbose,
but also adds succinct mathematical descriptions (using Wikipedia
notation) to all steps.
No functional changes are intended with this change.
Bug: webrtc:14151
Change-Id: I6a4642e89e2b73639f0b4c928e07b317c14d5884
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271546
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37784}
* Update naming of data members.
* Start reordering code blocks in `PredictAndUpdate`.
(The "predict" step is done in this C:. The "update"
step will be improved in another cl.)
Bug: webrtc:14151
Change-Id: Idea1e8e786bd672dadedbcb3cd5598f4a033e81e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271023
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37767}
This CL simplifies and documents the interface of the Kalman filter better. A simple unit test verifying the filter's convergence is
added. No functional changes to the filter are intended.
Future CLs will improve the data member naming and code organization.
Bug: webrtc:14151
Change-Id: I01e60d4b783cabad3ccbdc711c5d746666dd16ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265877
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37766}
Instead of showing individual byte differences, this CL detects
differences in the expected and actual byte streams of the evaluated
AEC dump and, if detected, parses the `audioproc::Event` proto lite
messages and calls EXPECT_EQ() for a subset of individual (sub-)fields.
Note that messages are parsed only if the byte streams of each message
pair do not match, so with no failures the test runs at no extra cost.
Plus, the the added funcionality can only be enabled for local
debugging by flipping the `kDumpWhenExpectMessageEqFails` flag - a
code change cannot land if the flag is set to true.
Note that `MessageDifferencer` (see [1]) could not be used because
it is not implemented for `MessageLite` protos.
[1] https://developers.google.com/protocol-buffers/docs/reference/cpp/google.protobuf.util.message_differencer
Bug: b/241923537
Change-Id: I8e0eda3b1ecfe06945b6dad5ee8871f8200d76d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270922
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37765}
When ScreencastPortal::OnStartRequestResponseSignal receives either a
non-zero response code or is missing the response data, it would
directly cast this to a RequestResponse. However, this direct cast is an
error. Per the documentation, the response signal returns the following
values with their corresponding meanings:
0 - Success
1 - User Cancelled
2 - Error
The RequestResponse enum however, has "kUnknown" as 0, and thus
"kSuccess" as 1 (with all other values also shifted up by 1 value). This
means that when the portal was cancelled, we were still receiving
RequestResponse::kSuccess. This fixes the issue by removing the improper
cast and adding a translation function. This function is local for now
since no where else attempted to cast values to a RequestResponse; but
can be moved if the need arises.
Fixed: chromium:1351824
Change-Id: I4cd44d90055147c9592d590c7969dcfc3297a3d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271240
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37755}
