Also change the generation of compile_commands.json so that it
does not include reclient arguments.
Bug: webrtc:15874
Change-Id: I1bc2e7a8768a3467d31528d584f7e70d946a3301
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347900
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42110}
Also initial implementation wrapping the libaom AV1 encoder.
Note that for now this is intended for prototype purposes.
Bug: none
Change-Id: Iac42ca4aecb6a204601c9f00bfb300e3eda3c4f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306181
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42108}
fuchsia.ui.display.singleton
We previously used fuchsia.ui.scenic.Scenic/GetDisplayInfo to get
fuchsia.ui.gfx.DisplayInfo. This has been migrated to
fuchsia.ui.display.singleton.Info/GetMetrics and
fuchsia.ui.display.singleton.Metrics.
Bug: fuchsia:64206
Test: applied changes manually to local chromium repo's third_party/webrtc directory and compiled
Change-Id: If3c7fbd641ebd3b3333e7e5f1126f8f3ae3b97e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322780
Commit-Queue: Caroline Liu <carolineliu@google.com>
Reviewed-by: Emircan Uysaler <emircan@google.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#42104}
Link c++ tips of the week as best practices for better discoverability.
Extend deprecated section with recommendation to use recently published ABSL_DEPRECATE_AND_INLINE macro.
No-Try: True
Bug: None
Change-Id: I3336e5929ae31542ff12ef6b1dfd9c272b9c480a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347743
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42100}
Instead of passing it as optional parameter during construction, pass field trials as required parameters on use.
Test that create the EncoderStreamFactory might not have an easy access to the actual field trials, but prod code has appropriate field trials when uses the factory.
This way EncoderStreamFactory doesn't need to depend on global field trial string through FieldTrialBaseConfig class.
Bug: webrtc:10335
Change-Id: I8f7030e41579ff2c5dd362c491a4e1624b23e690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347700
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42098}
This removes many references to the unsupported ProxyInfo struct
but leaves temporary implementations for methods while downstream
code gets updated.
Bug: none
Change-Id: Iab4410b362a8296b2e00cf71080010e515f9f4ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344660
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42096}
That forces external field trials, thus VideoCoding will be able to remove dependency on the global field trials string through FieldTrialBasedConfig class.
Bug: webrtc:10335
Change-Id: I6d22a7d20a4433801a0086b0863cda78e91f4f60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347646
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42092}
This allows for a clangd version to be downloaded that matches
the clang release used by the build system.
Bug: webrtc:15874
Change-Id: Iced12b2e6b4a1a91bb1a97e0a6a2bb9d35cc51c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347741
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42089}
This reverts commit 501c4f37bfee47b26999ee291c5355ad64554df7.
Patch set 1 contains pure reland.
The reason why we want to do this is because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate.
That is the, initial probe will try to probe up to the max configured bitrate.
Bug: webrtc:14928
Change-Id: I6a8660da20ac54237f04a29461e03b31bd988bb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347643
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@google.com>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42086}
Since iwyu is now deprecated, we need to enable use of include-cleaner.
This approach gives some error messages when running, but does the job.
Bug: webrtc:15874
Change-Id: I431deef0f2e5ce99eb256a4d82aa32769ae58b41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347642
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42085}
Setting the test duration parameter is a bug fix.
The addition of test clip parameter is a new feature for screenshare_loopback, although this already exists for video_loopback.
Bug: None
Change-Id: I7c307b07df9896456a1372fcd22cb13a25e2ca34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347640
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42084}
This reverts commit 33cc83595ae7dd144c57c614fb62d286d9d7bf68.
Reason for revert: Perf bots showed that this cl cause a change in metrics. It looks like it is for the better, but we want this to be behind a field trial.
Original change's description:
> Ignore allocated bitrate during initial exponential BWE.
>
> The reason why we want to do this is because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate.
> That is the, initial probe will try to probe up to the max configured bitrate.
>
> ProbeController::SetFirstProbeToMaxBitrate will allow the first probe to
> continue up to the max configured bitrate, regardless of of the max
> allocated bitrate.
>
> Bug: webrtc:14928
> Change-Id: I6e0ae90e21a78466527f3464951e6033dc846470
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346760
> Reviewed-by: Diep Bui <diepbp@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42049}
Bug: webrtc:14928
Change-Id: I56ba58560b6857b6069552c02df822691f7af64d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347622
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42081}
The field trial has been default on for ages. This CL removes it.
Bug: b/40200151
Change-Id: I171f663a3e725b856238b14b26d083f6684586e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347621
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42080}
This expose a new GetSupportedH265Level API for WebRTC external
factories to calculate H.265 levels to be use for SDP negotation.
Bug: webrtc:13485
Change-Id: Ib420da2b9b1b7af00129294be5b3efec172e8faf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345544
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42079}
To address the limitations of select(), poll() is now used for Apple
(MacOS and iOS) systems in the PhysicalSocketServer.
Bug: webrtc:15421, webrtc:15908
Change-Id: Ic6703a08653ca608a714ea37ecbbfeaf29743c1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#42075}
This makes it easier to remove use of sigslot for SignalEvent
since the tests were written in a way that could set more than one
event handlers to the same callback method, which places unnecessary
requirements on the definition of the callback object. I.e. the
sigslot can't be replaced with a simple (single) std::function - which
would be consistent with how the event callback is used elsewhere
in the code.
Bug: webrtc:11943
Change-Id: I7e596295b1b534d4d49334449b1e01535eedf06d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344723
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42072}
following the audio changes. Note that RTT-related fields require
DLRR and are not implemented yet.
BUG=webrtc:12529
Change-Id: I3f9449fbe876a1b282a32f2bcebe1cf3e10989bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346580
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42069}
This CL updates RtpVideoStreamReceiver2 to use H26xPacketBuffer for
H.264 and H.265 packets. H.264 specific fixes are moved to
H26xPacketBuffer as well.
H26xPacketBuffer is behind field trial WebRTC-Video-H26xPacketBuffer.
Bug: webrtc:13485
Change-Id: I1874c5a624b94c2d75ce607cf10c939619d7b5b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346280
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42062}
