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}
Ensure OnNetworkStateEstimate behaves the same way as internal networks state updates.
Also, ignore OnNetworkStateEstimate if an internal estimator exist.
Bug: webrtc:10742
Change-Id: I7967d202381250c406824fb2d0574bb95d2cd592
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354102
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@google.com>
Cr-Commit-Position: refs/heads/main@{#42456}
and move usages to webrtc::RefCountInterface
This CL also moves more stuff to webrtc:: and adds backwards
compatible aliases for them.
Bug: webrtc:42225969
Change-Id: Iefb8542cff793bd8aa46bef8f2f3c66a1e979d07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353720
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42446}
Allow skipping the deinterleaving steps in PushResampler
before resampling when deinterleaved buffers already exist.
Bug: chromium:335805780
Change-Id: I2080ce2624636cb743beef78f6f08887db01120f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352202
Reviewed-by: Per Åhgren <peah@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42438}
The new version of MSan (rolled by [1]) detects the following:
```
==39908==WARNING: MemorySanitizer: use-of-uninitialized-value
#0 0x5591400a52ef in GetPlayoutDelayMs ./../../modules/audio_coding/neteq/decision_logic.cc:466:35
#1 0x5591400a52ef in webrtc::DecisionLogic::ExpectedPacketAvailable(webrtc::NetEqController::NetEqStatus) ./../../modules/audio_coding/neteq/decision_logic.cc:311:36
#2 0x5591400a39e9 in webrtc::DecisionLogic::GetDecision(webrtc::NetEqController::NetEqStatus const&, bool*) ./../../modules/audio_coding/neteq/decision_logic.cc:0:0
#3 0x55913cf590c9 in webrtc::DecisionLogicTest_PreemptiveExpand_Test::TestBody() ./../../modules/audio_coding/neteq/decision_logic_unittest.cc:139:3
#4 0x55913ef28283 in HandleExceptionsInMethodIfSupported<testing::Test, void> ./../../third_party/googletest/src/googletest/src/gtest.cc:0:3
#5 0x55913ef28283 in testing::Test::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:2710:5
#6 0x55913ef2ab46 in testing::TestInfo::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:2856:11
#7 0x55913ef2da34 in testing::TestSuite::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:3034:30
#8 0x55913ef621e8 in testing::internal::UnitTestImpl::RunAllTests() ./../../third_party/googletest/src/googletest/src/gtest.cc:5964:44
#9 0x55913ef60f54 in HandleExceptionsInMethodIfSupported<testing::internal::UnitTestImpl, bool> ./../../third_party/googletest/src/googletest/src/gtest.cc:0:0
#10 0x55913ef60f54 in testing::UnitTest::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:5543:10
#11 0x55913ee1a944 in RUN_ALL_TESTS ./../../third_party/googletest/src/googletest/include/gtest/gtest.h:2334:73
#12 0x55913ee1a944 in webrtc::(anonymous namespace)::TestMainImpl::Run(int, char**) ./../../test/test_main_lib.cc:203:21
#13 0x55913cbd36b8 in main ./../../test/test_main.cc:72:16
#14 0x7fdb18c73082 in __libc_start_main /build/glibc-LcI20x/glibc-2.31/csu/../csu/libc-start.c:308:16
#15 0x55913cb3e1a9 in _start ??:0:0
```
[1] - https://webrtc-review.googlesource.com/c/src/+/353620
Bug: b/344970813
Change-Id: I9b5d7791e68b4c494168ba9f007a3099ae21fed4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353581
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42433}
- avoid holding a lock across OnCaptureResult() callback to avoid a risk
of a possible deadlock
- annotate damage region as guarded by the same lock as latest frame as
both belong together
- document the acqusition order between locks
Bug: chromium:333945842
Change-Id: I9c65beed720ba54e40b85fb243a07d40524695f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353600
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Cr-Commit-Position: refs/heads/main@{#42432}
and files that broke when I fixed the first set.
Bug: webrtc:42226242
Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42429}
Mac OS X 10.5 was shipped in 2006, and Mac OS X 10.7 was shipped in
2010. Assume that WebRTC is not running on releases older than
those.
Bug: none
Change-Id: Ia7323c2ae7f186602aa972f390ea682bd2d1ff47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353240
Auto-Submit: Avi Drissman <avi@chromium.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42423}
With this cl, sending can be forced with field trial "WebRTC-RFC8888CongestionControlFeedback/force_send:true/"
In the future, ReceiveSideCongestionController::EnablSendCongestionControlFeedbackAccordingToRfc8888 if RFC 8888 has been negotiated.
Bug: webrtc:42225697
Change-Id: Ib09066aa89ca7b3fffc551da541090c69ab8d75f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352720
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42413}
Interleave and Deinterleave now accept two parameters, one for the
interleaved buffer and another for the deinterleaved one.
The previous versions of the functions still need to exist for test
code that uses ChannelBuffer.
Bug: chromium:335805780
Change-Id: I20371ab6408766d21e6901e6a04000afa05b3553
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351664
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42412}
Without this, packets may be sorted in the wrong order.
Bug: webrtc:42225697
Change-Id: Ib9a72cdc7cb8f7ef6ca1571d095a6474215a83f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352821
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42411}
Because it is flaky !?
Bug: webrtc:42225697, b/343600373
Change-Id: I74415a9b97e90c25807b55053fd549f335b863ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352820
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42408}
CongestionControlFeedbackGenerator collect receive time information about received
packets and sends feedback according to RFC8888
Bug: webrtc:42225697
Change-Id: I70b7f7322fd262f99f45fd56b6eb8630a11b30c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351543
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42404}
BWE logging has as far as I know know been used for a long time. RTC event logs are the prefered method of logging.
Removed since it causes some BUILD pain.
For debugging the metrics API https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/test/metrics/ can be used instead.
Bug: webrtc:343347276
Change-Id: I046b58d880faabfadbc22269b0392fdd644155fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352602
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42402}
