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}
Since multiple fixes are bundled behind the WebRTC-IPv6NetworkResolutionFixes
field trial, this allows more flexibility if the launch is found to cause any
issues.
Bug: webrtc:14334, webrtc:14131
Change-Id: I2e73b8984db1f3d292cecab98e29c53cfac1c070
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273903
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Auto-Submit: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#37995}
When a RECONFIG has been received with a last assigned TSN that is
not yet seen by the receiver, it will enter deferred reset mode
(https://www.rfc-editor.org/rfc/rfc6525#section-5.2.2, E2).
When more DATA is received, moving the cumulative acknowledgment point,
the request will finally be processed. But the last chunk that has the
same TSN as the last assigned TSN was before this CL not applied before
doing the reset - it was applied after.
This would result of a message getting lost or possibly getting
truncated or incorrectly merged with another.
Handling the message before resetting the stream is the simple
solution here.
Bug: webrtc:14277
Change-Id: Iea9fa227778077a9ff2f78bc77b5d93cc32b702b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273323
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37993}
Ultimately, IDs should be random according to spec[1], so we shouldn't
rely on the ID to convey easily readable information. By making the IDs
shorter we reduce the overhead of string copies and make report dumps a
little bit smaller.
Drive-by: Add "DEPRECATED_" prefic to the RTCMediaStreamStats ID.
[1] https://w3c.github.io/webrtc-pc/#dom-rtcstats-id
# Examples of IDs before and after this CL #
RTCDataChannel_3
-> D3
RTCPeerConnection
-> P
RTCTransport_0_1
-> T01
RTCCodec_RTCTransport_0_1_100_minptime=10;useinbandfec=1
-> CIT01_100_minptime=10;useinbandfec=1
RTCInboundRTPAudioStream_6666
-> IA6666
RTCAudioSource_1
-> SA1
RTCOutboundRTPAudioStream_2943129392
-> OA2943129392
RTCRemoteInboundRtpAudioStream_3541280085
-> RIA3541280085
RTCIceCandidate_6cWRqicY
-> I6cWRqicY
RTCIceCandidatePair_6cWRqicY_haEcM2xD
-> CP6cWRqicY_haEcM2xD
RTCCertificate_FD1:BC:58:90:DF:E8:40:58:8D:04:91:44:93:4E:6C:52:9E:F0:14:98:AA:67:7B:8B:C8:30:C8:31:D0:84:1B:BF
-> CFD1:BC:58:90:DF:E8:40:58:8D:04:91:44:93:4E:6C:52:9E:F0:14:98:AA:67:7B:8B:C8:30:C8:31:D0:84:1B:BF
DEPRECATED_RTCMediaStreamTrack_receiver_3
-> DEPRECATED_TI3
RTCMediaStream_45a6e766-5d1a-40f9-a55c-ea8fdefcde49
-> DEPRECATED_S45a6e766-5d1a-40f9-a55c-ea8fdefcde49
Bug: webrtc:14416, webrtc:14419
Change-Id: I11f0a8b8354203fea1df1093d8864a6d47ee71e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273709
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37992}
The code incorrectly assumed that codecs exist on a per-mid/transceiver
basis, but codec payload types are unique on a per-transport basis and
in practise most applications use BUNDLE (single transport for the
entire PC).
This CL makes the codecs per-transport instead of per-transceiver. We
still need to iterate transceivers because codecs are exposed on a
per-transceiver basis and as shown in
https://jsfiddle.net/henbos/7kqxgnr8/ it is possible for FMTP lines to
be different on different m= sections despite BUNDLE.
Manual testing shows that this CL brings down the number of "codec"
stats in Google Meet 50p from 872 objects to 43 objects.
Bug: webrtc:14414
Change-Id: Ic854b31bd595799554b99fff22cbd48264ebd141
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273707
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37989}
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}
which may allow caching some relatively persistent statistics
such as codec statistics that only change during renegotiation.
BUG=webrtc:8693
Change-Id: Ifd68c9d666d9f328d0efecb64e4201d003788ca8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273324
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37981}
The frame cadence adapter would previously crash on encountering
unconfigured layer updates. Fix this to silently ignore such updates.
Fixed: webrtc:14417
Change-Id: I524aa196f6aed10ffb8cd4ddcb4b053c71dfabba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273325
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37980}
This is a reland of commit 9a0a6a198e8e247884fe01d7e0aa6bd425721c14
Original change's description:
> Reland "ObjC ADM: record/play implementation via RTCAudioDevice [3/3]"
>
> This is a reland of commit 2b9aaad58f56744f5c573c3b918fe072566598a5
>
> Original change's description:
> > ObjC ADM: record/play implementation via RTCAudioDevice [3/3]
> >
> > # Overview
> > This CL chain exposes new API from ObjC WebRTC SDK to inject custom
> > means to play and record audio. The goal of CLs is achieved by having
> > additional implementation of `webrtc::AudioDeviceModule`
> > called `ObjCAudioDeviceModule`. The feature
> > of `ObjCAudioDeviceModule` is that it does not directly use any
> > of OS-provided audio APIs like AudioUnit, AVAudioEngine, AudioQueue,
> > AVCaptureSession etc. Instead it delegates communication with specific
> > system audio API to user-injectable audio device instance which
> > implements `RTCAudioDevice` protocol.
> > `RTCAudioDevice` is new API added to ObC WebRTC SDK in the CL chain.
> >
> > # AudioDeviceBuffer
> > `ObjCAudioDeviceModule` does conform to heavy `AudioDeviceModule`
> > interface providing stubs for unrelated methods. It also implements
> > common low-level management of audio device buffer, which glues audio
> > PCM flow to/from WebRTC.
> > `ObjCAudioDeviceModule` owns single `webrtc::AudioDeviceBuffer` which
> > with the help of two `FineAudioBuffer` (one for recording and one for
> > playout) is exchanged audio PCMs with user-provided `RTCAudioDevice`
> > instance.
> > `webrtc::AudioDeviceBuffer` is configured to work with specific audio:
> > it has to know sample rate and channels count of audio being played and
> > recorded. These formats could be different between playout and
> > recording. `ObjCAudioDeviceModule` stores current audio parameters
> > applied to `webrtc::AudioDeviceBuffer` as fields of
> > type `webrtc::AudioParameters`. `RTCAudioDevice` has it's own variable
> > audio parameters like sample rate, channels count and IO buffer
> > duration. The audio parameters of `RTCAudioDevice` must be kept in sync
> > with audio parameters applied to `webrtc::AudioDeviceBuffer`, otherwise
> > audio playout and recording will be corrupted: audio is sent only
> > partially over the wire and/or audio is played with artifacts.
> > `ObjCAudioDeviceModule` reads current `RTCAudioDevice` audio parameters
> > when playout or recording is initialized. Whenever `RTCAudioDevice`
> > audio parameters parameters are changed, there must be a notification to
> > `ObjCAudioDeviceModule` to allow it to reconfigure
> > it's `webrtc::AudioDeviceBuffer`. The notification is performed
> > via `RTCAudioDeviceDelegate` object, which is provided
> > by `ObjCAudioDeviceModule` during initialization of `RTCAudioDevice`.
> >
> > # Threading
> > `ObjCAudioDeviceModule` is stick to same thread between initialization
> > and termination. The only exception is two IO functions invoked by SDK
> > user code presumably from real-time audio IO thread.
> > Implementation of `RTCAudioDevice` may rely on the fact that all the
> > methods of `RTCAudioDevice` are called on the same thread between
> > initialization and termination. `ObjCAudioDeviceModule` is also expect
> > that the implementation of `RTCAudioDevice` will call methods related
> > to notification of audio parameters changes and audio interruption are
> > invoked on `ObjCAudioDeviceModule` thread. To facilitate this
> > requirement `RTCAudioDeviceDelegate` provides two functions to execute
> > sync and async block on `ObjCAudioDeviceModule` thread.
> > Async block could be useful when handling audio session notifications to
> > dispatch whole block re-configuring audio objects used
> > by `RTCAudioDevice` implementation.
> > Sync block could be used to make sure changes to audio parameters
> > of ADB owned by `ObjCAudioDeviceModule` are notified, before interrupted
> > playout/recording restarted.
> >
> > Bug: webrtc:14193
> > Change-Id: I5587ec6bbee3cf02bad70dd59b822feb0ada7f86
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269006
> > Reviewed-by: Henrik Andreasson <henrika@google.com>
> > Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com>
> > Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37928}
>
> Bug: webrtc:14193
> Change-Id: Iaf950d24bb2394a20e50421d5122f72ce46ae840
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273380
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37946}
Bug: webrtc:14193
Change-Id: I84a6462c233daae7f662224513809b13e7218029
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273662
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37977}
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}
When working within chromium, we need to move up two more levels before
we can import the "find_depot_tools" module. Previously this was
managed by a loop that kept moving up until it found a directory named
'src', but this didn't work if the base-level webrtc code was not in
a directory named 'src'.
Bug: webrtc:14411
Change-Id: I2729603d6d2fd8242a1294ae5d95d946db619d0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273561
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37971}
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}
AV1 tests seem to be running fine now that we have the dependency
descriptor enabled, so remove the need for the RTP header extension
as it doesn't allow discarding frames.
Bug: webrtc:11607
Change-Id: Ifd0670ab61a5b69d0570f65ba30c352a31376992
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273488
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37952}
