Similar to BGRA/RGBA we added recently, formats from PipeWire are in
big-endian, while WebRTC (using libyuv) is little-endian, therefore we
have to map BGR to RGB and not RGB to RGB as colors would be off. Also
add some additional formats supported by libyuv.
Bug: webrtc:42225999
Change-Id: Iee8303f0922fe434069b2b3f88994abecf7d2cc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355860
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#42609}
* Simplified ctor. Get settings (max_qp, content_type, etc) from encoder_config passed to CreateEncoderStreams().
* Some tests assigned VideoEncoderConfig::video_stream_factory to EncoderStreamFactory they created. That's not really needed. VideoStreamEncoder creates the factory if video_stream_factory is not provided [1]. Removed video_stream_factory initialization in tests.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/video_stream_encoder.cc;l=1002;drc=1d7d0e6e2c5002815853be251ce43fe88779ac85
Bug: b/347150850, webrtc:42233936
Change-Id: Ie0322abb6c48e1a9bd10e9ed3879e3ed484fea5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355321
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42608}
For better consistency with the rest codebase (it is min_/max_ for all params in video_encoder.h; only qp is for some reason prefixed with minimum_).
Also fixed constant names in libaom AV1 encoder wrapper (moved min from suffix to prefix, minimum -> min_).
Bug: chromium:328598314
Change-Id: I6d8521a3abff3a0595a5241c02ef4746eb4694df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356600
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42604}
Before this change the AV1 encoder wrapper converted target frame rate from double to integer with rounding to the middle. That approach resulted in a bitrate mismatch caused by rounding error. The mismatch was especially high at low frame rates. For example, at target frame rate 1.4fps the bitrate mismatch reached 40%:
out/debug/video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --framerate_fps=1.4 --width=320 --height=180 --bitrate_kbps=32 --num_frames=600
...
RESULT s0t0_bitrate_mismatch_pct: DISABLED_EncodeDecode= {39.171875,0} n%
After the change the mismatch reduced to ~2% in the same scenario:
RESULT s0t0_bitrate_mismatch_pct: DISABLED_EncodeDecode= {-2.178125,0} n%
Bug: b/337757868
Change-Id: Ia51f92b3dfdce103eed1d04cac0e084b69fa8213
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356500
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42601}
This is cherry-picked from WebKit's patch for fixing a fuzzer failure.
The original patch: https://github.com/WebKit/WebKit/pull/30438
Bug: chromium:41480904
Change-Id: Ic8eddb9de816c4c8d720dac6d4c55d1db3f0596e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356361
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#42598}
If traffic policing is enforced by dropping packets, RTT can still be low.
If a packet is dropped that is needed to contninue decoding, it make sense that a nack request is sent until the packet is received, or a new key frame is requested. A key frame will be requested after 3s.
For now, this cl only increase the number of times a packet can be requested.
Bug: b/317178411
Change-Id: Iea75d36ed06f346af1dd4e55a9961d5eca45f519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356482
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42594}
This helps with making AudioBuffer compatible with current and upcoming
code that uses audio_views.h (a simpler abstraction).
Bug: chromium:335805780
Change-Id: Ib59bba274c7abfb441e3c4d606f804b365df236d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355844
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42590}
Bitrate limits should have been applied to the spatial layers in ApplySpatialLayerBitrateLimits (and usage is restricted to a single active stream/layer).
Bug: b/299588022
Change-Id: Iaae4ece28b8a95eea7d4bacba292847ba5b4000b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355841
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42588}
This switches from accepting a sample rate and convert to channel
size over to accepting the channel size.
Instead of InitializeIfNeeded:
* Offer a way to explicitly initialize PushResampler via the ctor
(needed for VoiceActivityDetectorWrapper)
* Implicitly check for the right configuration from within Resample().
(All calls to Resample() were preceded by a call to Initialize)
As part of this, refactor VoiceActivityDetectorWrapper (VADW):
* VADW is now initialized in the constructor and more const.
* Remove VADW::Initialize() and instead reconstruct VADW if needed.
Add constants for max sample rate and num channels to audio_util.h
In many cases the numbers for these values are embedded in the code
which has led to some inconsistency.
Bug: chromium:335805780
Change-Id: Iead0d52eb1b261a8d64e93f51401147c8fba32f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353360
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42587}
Some dependencies still exist but are a bit more complex to remove.
This CL removes either unused or easily replaced with ToString()
instances of ostream usage. In one case, moving the operator<<
implementation to the one test file that requires it.
Bug: webrtc:8982
Change-Id: Ia5c840b12a42893494af401317a3daf2fe50ba9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356240
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42582}
* FrameCombiner is simpler. No additional channel pointers for buffers.
* Improve consistency in using views in downstream classes.
* Deprecate older methods (some have upstream dependencies).
* Use samples per channel instead of sample rate where the former is
really what's needed.
Bug: chromium:335805780
Change-Id: I0dde8ed7a5a187bbddd18d3b6c649aa0865e6d4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352582
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42575}
This reverts commit 86ff48adaea08fd4e7044595e1c25a22fcceac34.
Reason for revert: Speculative revert due to failing downstream tests
Original change's description:
> Rewrite simulcast config to equivalent SVC for vp9 simulcast
>
> This allows to utilize libvpx optimizations considerably improving performance.
> The change happens inside libvpx_vp9_encoder and is invisible to other parts of webrtc.
>
> This CL includes unit tests, an E2E test already exists: StandardPath/PeerConnectionEncodingsIntegrationParameterizedTest.Simulcast/VP9 in peerconnection_unittests.
>
> Bug: webrtc:347737882
> Change-Id: Ic48316ad597700ed07e594d592413cf84b6b20d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355003
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42554}
Bug: webrtc:347737882
Change-Id: Ib84c9c0e20763348abfae838f2fb1aff31581a55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355943
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42564}
The idea is to reduce the risk of calculating a packet as lost if a
packet is reordered between two feedback reports.
It works as long as the recevied feedback does not complete an
observation.
Bug: webrtc:42222865 b/349765923
Change-Id: Iaf1595e624f546951baf3998d161f4cd1d5d491b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355942
Reviewed-by: Diep Bui <diepbp@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42562}
std::optional<T>::emplace() without an initializer is broken on clang++
with gnu libstdc++. this workarounds the bug by removing the
absl::optional wrapping, which is actually pointless.
https://gcc.gnu.org/bugzilla/show_bug.cgi?id=101227
Bug: chromium:41455655
Change-Id: I05354e57cc4cdda3fa6d3cd23f46462b69cc3bee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355900
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42561}
The minimum QP field will be used to signal what the QP value will be
once the encoder reach its target video quality. This will be used
in the generalized QP convergence detection.
Bug: chromium:328598314
Change-Id: I82299cd921e3c091e651218d1e3f337875176567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355701
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Cr-Commit-Position: refs/heads/main@{#42559}
This allows to utilize libvpx optimizations considerably improving performance.
The change happens inside libvpx_vp9_encoder and is invisible to other parts of webrtc.
This CL includes unit tests, an E2E test already exists: StandardPath/PeerConnectionEncodingsIntegrationParameterizedTest.Simulcast/VP9 in peerconnection_unittests.
Bug: webrtc:347737882
Change-Id: Ic48316ad597700ed07e594d592413cf84b6b20d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355003
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42554}
Set cursor position to (-1,-1) to indicate it's not valid when it goes
off the screen or it gets hidden by the compositor. Compositors indicate
invalid or hidden cursor by unsetting the cursor id in cursor metadata
and using spa_meta_cursor_is_valid() will tell us the needed information
for this.
Bug: chromium:346608851
Change-Id: I71b3222ca161b7fd8e964f4f4e12b9983179beba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355080
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#42548}
This sets the correct frame DPI according to the pixels/DIPs ratio.
It also sets the capture_time_ms for consistency with ScreenCapturerMac.
Bug: chromium:327458809
Change-Id: Ibb0074756e262dd1ce6f2897f60f0d939ddb7fd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355442
Commit-Queue: Lambros Lambrou <lambroslambrou@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Lambros Lambrou <lambroslambrou@chromium.org>
Cr-Commit-Position: refs/heads/main@{#42534}
This option will allow clients to control which ScreenCapturer is used,
for versions of macOS that support ScreenCaptureKit. The default is to
use the previous code, to avoid breaking current users of the module.
Bug: chromium:327458809
Change-Id: Ib0f9390c85d726016a39eea4fda9b8bd14a094c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355020
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Lambros Lambrou <lambroslambrou@chromium.org>
Cr-Commit-Position: refs/heads/main@{#42518}
This is to make sure that the two encoders are "in sync" (the CNG
encoder can be created from an existing speech encoder).
This is a speculative fix for a crash in the CNG encoder where a packet
is unexpectedly emitted from the speech encoder.
Bug: webrtc:42225071
Change-Id: I42571e56e032897f7f083f04d785f6a08ebfb813
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355160
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#42516}
First batch of applying iwyu to the repo.
Done with:
> ./tools_webrtc/iwyu/apply-iwyu api
> git add api/[a-s]*
> python3 gn_autodeps.py ~/local/webrtc/src out/Default
Last step is a custom script I wrote to automatically apply new required
dependencies for target in gn, which saved tons of time manually going
over the files and fixing.
If this is something that interest others, I can submit it as well.
Bug: webrtc:42226242
Change-Id: Id109e77f50835827495bc4512880c4ec9ae175f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42512}
This supports:
* Full-screen capture from any display, via SelectSource().
* Changing the display, via SelectSource(), while capture is running.
* Handling screen-resolution changes while capture is running.
* Capturing from high-DPI displays at their native resolution.
* Basic damage-tracking: the frame's updated-region is either set to
empty, or the full frame area.
It currently does not support:
* Window capture.
* Excluded windows.
* Full-desktop capture across all displays.
* More detailed damage-tracking.
The capturer is not yet enabled. Followup CLs will add a
DesktopCaptureOption to enable this capturer on supported versions of
macOS.
Bug: chromium:327458809
Change-Id: Ie619f6c6c1d6edf0fb9320d4fece578754a732dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352544
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Lambros Lambrou <lambroslambrou@chromium.org>
Cr-Commit-Position: refs/heads/main@{#42510}
OpenH264 cannot be usually used everywhere as it's proprietary and for
that reason it's usually disabled or apps using it are not allowed to be
available in default installations. Using system OpenH264 option allows
us to use e.g. noopenH264, that can be present in default installations
and later replaced by OpenH264 installed from 3rd party repository.
Bug: webrtc:14717
Change-Id: I015aacdb48c0636935f611459f0c9a6aa74a8f94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349301
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#42509}
Using DeinterleavedView<> simplifies these two classes, so now the
classes are arguably thin wrappers on top of DeinterleavedView<> and
AudioFrameView<> can be replaced with DeinterleavedView<>.
The changes are:
* Make VectorFloatFrame not use a vector of vectors but rather
just hold a one dimensional vector of samples and leaves the mapping
into the buffer up to DeinterleavedView<>.
* Remove the `channel_ptrs_` vector which was required due to an
issue with AudioFrameView.
* AudioFrameView is now a wrapper over DeinterleavedView<>. The most
important change is to remove the `audio_samples_` pointer, which
pointed into an externally owned pointer array (in addition to
the array that holds the samples themselves). Now AudioFrameView
can be initialized without requiring such a long-lived array.
Bug: chromium:335805780
Change-Id: I8f3c23c0ac4b5a337f68e9161fc3a97271f4e87d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352504
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42498}
This is a forward fix for https://webrtc-review.googlesource.com/c/src/+/354622 that breaks client code using nalus_length.
No-Try: true
Change-Id: Ic0fc41696e408adefe4eb8792150a64b1eab49da
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354840
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42493}
Some DMAbuf types don't properly implement MAP_PRIVATE as it requires
copy-on-write support. As we don't need to write to these buffers, we
can switch to MAP_SHARED instead, making it work reliably on current
kernels without having any drawbacks in this context.
Tested and confirmed with libcamera software ISP on Thinkpad X13 with
an arm processor.
Bug: webrtc:42225999
Change-Id: Ic47b8c90456cccf3742e8274945dbd64fb8aac6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354623
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42484}
Do not add offset to the "length" argument for mmap call as it should be
passed as the last argument instead. This was not causing any problems
since the offset is usually 0, but it's still better to do it correctly.
Bug: webrtc:42225999
Change-Id: If1dbe7dfd2fb22c53493c0fafd23d782f0683a11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354521
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#42481}
This simplifies the following steps:
* FrameCombiner infers the sample rate from channel size
* Sends the inferred sample rate to FixedDigitalLevelEstimator
and Limiter.
* Those classes then convert the sample rate to channel size.
Along the way perform checks that the derived channel size value
is a legal value (which has already been done by FrameCombiner).
To:
* FrameCombiner sends channel size to FixedDigitalLevelEstimator and
Limiter.
Bug: chromium:335805780
Change-Id: I6d2953ba5ee99771f3ff5bf4f4a049a8a29b5577
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352581
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42480}
Adds support for 32 bits formats needed for libcamera software ISP. This
is needed, because libcamera enforces 8 byte alignment and we only
support 3 byte alignment for RGB. This will make it work with 32 bits
aligned output formats recently added to libcamera.
Relevant libcamera patch: https://patchwork.libcamera.org/patch/20253/
This has been verified on an snapdragon device using libcamera and software ISP and on my machine using "vivid" virtual camera from libcamera and enforcing specific format.
Bug: webrtc:346808586
Change-Id: I8d89120660b2304b880d952c5acd7f5cd09b611e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354400
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42474}
Set layer QP limits equal to QP limits in the main encoder config. This reduces number of nodes to modify if you need to change the settings.
Bug: b/337757868
Change-Id: Id7f6f9d6527903e8e22ff4fad2c974bee6e87cb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353982
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42466}
Along the way slightly simplify the class interface since views
carry audio properties. Also, now allocating FrameCombiner allocates
the mixing buffer in the same allocation.
Bug: chromium:335805780
Change-Id: Id7a76b040c11064e1e4daf01a371328769162554
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352502
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42465}
This avoids having duplicate camera entries presented to the user when
PipeWire camera is being used.
Bug: webrtc:346350844
Change-Id: I423db7fe0654cc1b1c91ee5264c6ba5dc4e24100
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354320
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#42462}
