The Chromium RTCVideoEncoder unfortunately doesn't set if the
result is at target quality, and the definition of the threshold
is buried in libvpx_vp8_encoder.h.
This change
* Updates VideoStreamEncoder to postprocess an incoming EncodedImage
by interpreting the incoming QP information instead.
* Updates the related VideoStreamEncoder test to simulate an encoder
producing images around the QP threshold.
* Updates the steady state VP8 screencast QP threshold to a central
include file.
* Moves this and previously existing EncodedImage post-processing to a
new method AugmentEncodedImage.
Bug: b/245029833
Change-Id: I69ae29ffe501e84f28908f7d9a8cfd066ba82b43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38091}
This functionality should have been added as part of the original CLs,
but was missed. The purpose of the validation is avoid catastrophic
failures due to misconfiguration (such as RTC_CHECK crashes).
The purpose is not to always provide practically reasonable values.
Bug: webrtc:14151
Change-Id: Icbddade865bd6a868f467a1df7055026935f36f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275560
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38090}
The value is today set to 200 which is too low for an audio packet to trigger sending probes.
For the initial probing, it would be good if audio packets, that may arrive before the first video frame can trigger sending a probe.
Also fix field trial parsing of required number of probes.
Bug: webrc:14392
Change-Id: I1f3cebcda38b71446e3602eef9cfa76de61a1ccf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275620
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@{#38089}
https://crrev.com/c/3885576 removed the last downstream consumer of the
constructor which took a ScreenCastPortal::CaptureSourceType. Now that
it has been removed, that constructor definition can also be removed and
the CaptureSourceType enum can be made private. There's still benefit in
storing and using this internally as the enum, since it's values match
that of the underlying system API.
The previously anonymous-namespaced function |ToCaptureSourceType| had
to be converted to a private static method as part of this change, since
it would be unable to access the type otherwise.
Fixed: chromium:1359411
Change-Id: I81ff24fbdddf9db02c9c5152d007dd82c194865a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274680
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38084}
* Make `percentile` configurable and rename class.
* Introduce convenience type `MovingMedianFilter` that
maintains the behaviour of the old class with that name.
* Move home grown moving 95th percentile filter in
`JitterEstimator` to this new utility class.
Bug: webrtc:14151
Change-Id: I17d525b6e0bc98c28568c7dfe94b72eeab4a1ca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275311
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38082}
This change adds a median filter that can replace the
IIR filter that is currently used to estimate the
avg frame size (in bytes). It is enabled through a boolean,
and reuses the window length from the max percentile filter.
The median filter is only used by the delay calculation in
`CalculateEstimate()`. It does not replaced the use of the
IIR estimate in the size outlier rejection heuristic.
Bug: webrtc:14151
Change-Id: I519b6b57a8bee3c41a300ed2e92a1981c61cca15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275121
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38077}
Loss based BWE v2 rate is updated immediately when transport feedback is received.
This ensure that when GoogCcNetworkController::MaybeTriggerOnNetworkChanged is invoked, the loss based estimate is updated.
Bug: webrtc:14392
Change-Id: If404576c5793a29096cea52884862807cde8b615
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275306
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38070}
Detected by the new UBSan configurations (crbug.com/1352721):
[ RUN ] TestDecodingState.FrameContinuity
../../modules/video_coding/session_info.cc:250:10: runtime error: null pointer passed as argument 1, which is declared to never be null
../../build/linux/debian_bullseye_amd64-sysroot/usr/include/string.h:44:28: note: nonnull attribute specified here
No-Try: True
Bug: None
Change-Id: Ib4fb20d948b41da1a35dacb8abe944eb24f806f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275200
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38063}
Main changes:
- `AgcManagerDirectTestHelper::FirstProcess()` replaced by
`CallAgcSequence()`, which is API contract compliant
- `ExpectCheckVolumeAndReset()`, `SetVolumeAndProcess()` and
`ExpectInitialize() `removed
- TODOs added for the next batch of improvements
- `AgcManagerDirectTestHelper::mock_agc` now using `NiceMock`
- `AgcManagerDirect::(AnalyzePre)Process()` now receives a
const ref
- `AnalyzePreProcess(const float* const*,size_t )` removed
Bug: webrtc:7494
Change-Id: Ie5bbaa590586dd806b30494fb00ca9c742c241e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273490
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38056}
The current design of the modified getter is error-prone since the
returned value changes meaning based on when (which point in the code)
the getter is called - namely, before `ProcessStream()` is called the
getter returns the stream analog level, after it returns the
recommended level.
Plus, the new implementation, which essentially returns a local
member, removes the risks that the non-trivial implementation
is computationally expensive.
Bug: webrtc:7494, b/241923537
Change-Id: I6714444df27bcc055ae693974ecd1f77f79e6ec0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271580
Reviewed-by: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38055}
The `recommended_stream_analog_level()` getter is used to retrieve
both the applied and the recommended input volume. This behavior is
error-prone since the caller must know what is returned based on
the point in the code (namely, before/after the AGC has changed
the last applied input volume into a recommended level).
This CL is a first step to make clarity on which input volume is
handled in different parts of APM. Next in the pipeline: make
`recommended_stream_analog_level()` a trivial getter that always
returns the recommended level.
Main changes:
- When `recommended_stream_analog_level()` is called but
`set_stream_analog_level()` is not called, APM logs an error
and returns a fall-back volume (which should not be applied
since, when `set_stream_analog_level()` is not called, no
external input volume is expected to be present
- When APM is used without calling the `*_stream_analog_level()`
methods (e.g., when the caller does not provide any input volume),
the recorded AEC dumps won't store `Stream::applied_input_level`
Other changes:
- Removed `AudioProcessingImpl::capture_::prev_analog_mic_level`
- Removed redundant code in `GainController2` around detecting
input volume changes (already done by APM)
- Adapted the `audioproc_f` and `unpack_aecdump` tools
- Data dumps clean-up: the applied and the recommended input
volumes are now recorded in an AGC implementation agnostic way
Bug: webrtc:7494, b/241923537
Change-Id: I3cb4a731fd9f3dc19bf6ac679b7ed8c969ea283b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271544
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38054}
Switching to an AGC implementation agnostic solution for the input
volume emulation functionality offered by the
`capture_levels_adjuster` sub-module.
This CL also fixes a (silent) bug due to which, when the input
volume is emulated via the capture adjuster sub-module, AGC2
reads an incorrect value for the applied input volume.
Tested: audioproc_f with `--analog_mic_gain_emulation 1` used
to verify bit-exactness for one Wav file and one AEC dump for
which the input volume varies.
Bug: webrtc:7494, b/241923537
Change-Id: Ide3085f9a5dfd85888ad812ebd56faa175fb2ba7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273902
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38053}
`EchoControllerObservesAnalogAgc1EchoPathGainChange` is incorrect
since it does not call `set_stream_analog_level()`,
`ProcessCapture()` and `recommended_stream_analog_level()`
according to the contract.
`EchoControllerObservesNoDigitalAgc2EchoPathGainChange` is
useless since AGC2 doesn't have any analog controller at the
moment and the test is not written to explictly trigger digital
gain adaptations.
Bug: webrtc:7494, b/241923537
Change-Id: I56203c736448ec060077b00b57e98cd4c29fa737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271541
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38052}
Making it clear that the field is used to store the applied input
volume and not the recommended input volume.
Bug: webrtc:7494, b/241923537
Change-Id: Ib91bc1a12348f63e3a4ba6e068ed02e40786a87b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271342
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38051}
This CL adds the ability to forward aborted retransmission notifications
to specified RTP modules, as well as a way to find the RTX ssrc
associated with a media SSRC.
These will both be used by upcoming logic that can selectively flush
given streams from the pacer queue.
Bug: webrtc:11340
Change-Id: Ief3be47e4fd7dc5a1499bc21890e8979400ecb44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274706
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38050}
This change adds a percentile filter that can replace the
"non-linear IIR" filter that is currently used to estimate the
max frame size (in bytes). The percentile filter is enabled through
the field trial, and it has two tuning parameters: the percentile
that is deemed the "max" frame, and the window length over which
the filter is applied.
Bug: webrtc:14151
Change-Id: I002609edb0a74161aaa6f0934892a1bec2ad8230
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274167
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38047}
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed
Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
Fixing a bug due to which the analog controller could not be disabled.
AudioProcessing::Config::GainController1::AnalogGainController::enabled
was ignored and therefore `recommended_stream_analog_level_locked()` in
APM was returning the level recommended by `AgcManagerDirect`.
When the analog controller is disabled, `stream_analog_level()` now
returns the last value set via `set_stream_analog_level()`.
However, the analog controller code is still running and, in particular,
the existing metrics are reported as if the controller were enabled.
This choice was made to reduce the risks of adding bugs in the digital
compression gain selection part, which is tied to the analog
controller. The metric drawback will be solved in a follow-up CL.
Additional changes:
- log `WebRTC.Audio.GainController.Analog.Enabled` when
AGC1 is created or when its config changes
- first step to replace "analog level" with "input volume"
Bug: webrtc:7909, b/180019868
Change-Id: I28ce9556dd98f3dd9ad546799406c55478730435
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270663
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38044}
For the linked test case that speeds up chromium fuzzer by ~13%
Run time decrease from ~50 seconds to ~44 seconds
Bug: chromium:1357929
Change-Id: I702edf4fda7afd31a5288621220dac063f764ced
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274601
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38038}
In some upcoming use cases we might wish to flush pending
retransmissions from the pacer queue. In order to not make those packets
forever inaccessible this CL adds a way to clear their "pending" status
from the packet history.
Bug: webrtc:11340
Change-Id: I9aac44125899a7f1e5a5e5be3390ac07b1e661ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274600
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38037}
The Chrome Remote Desktop team is looking to support AV1 profile-1
w/ I444 for screen sharing however only I420 is currently supported.
This CL adds I444 support for the Dav1dDecoder, which appears to be
the preferred decoder and adds profile-1 to the
InternalDecoderFactory when the Dav1dDecoder is being used.
I've tested this CL using a CRD host w/ I444 enabled and it seems to
work as expected, though I've only tested on a debug build so I plan
to do some perf testing once this is available in a release build.
Bug: chromium:1329660
Change-Id: I2b8b7b7fd530727456ac5c46e694e7dbad6deff2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273986
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Joe Downing <joedow@google.com>
Cr-Commit-Position: refs/heads/main@{#38022}
The constants are reordered to match the order they are used
when a sample is inserted into the filter. Some of the constants
are renamed to better describe their usage. No functional changes
are intended. Future CLs will add configurability to some of these
constants.
Some basic unit tests are also added.
Bug: webrtc:14151
Change-Id: I731a9cad3d8aeab06ccfa7d212cd160a2d2da27b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274122
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38019}
This makes it easier to read the code, and to visually verify
that the computations make sense from a dimensional perspective.
No functional changes are intended.
Bug: webrtc:14151
Change-Id: Ic059f3c53618903d63a270b901ac5cec6139d2c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274120
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38006}
Change ProbeController field trial to also probe when loss limited but probe at the current estimate.
Bug: webrtc:14392
Change-Id: I8b30e316b935a0f2c375e2204a8e33e6671eb956
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273901
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38004}
Now that we've added the ability to open and close the PipeWire picker
to the DesktopCapture interface, we can split the picker back into a
Window and a Screen picker rather than just having the one combined
picker. This will allow for a better user experience, as we can create
a picker targeted to what the users actually want to share.
Bug: chromium:1351570
Change-Id: I5bec22912ae01c1b0b0709a4979b4698226a2a66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273541
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38000}
Adds an Observer class to the DelegatedSourceListController so that we
can expose user-driven events from the delegated source list. This will
allow embedders to update their UI in response to changes to the state
from the delegated source list. Currently these events are: selection,
cancelled, and error.
Bug: chromium:1351576, chromium:1351577
Change-Id: I248bdb1c4410147ca1a0663eafda40b6b9345801
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272622
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37986}
This reverts commit 0098a441e3990f80bcbe05dfce8c6e2359f7ef25.
Reason for revert: Fix chromium build break
Original change's description:
> Revert "Add plumbing to control PipeWire picker visibility"
>
> 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}
Bug: chromium:1351572
Change-Id: I9e5e691746b81517bf0e211d0ad5a23371ab29ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273685
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37972}
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}
