Similar to PipeWire implementation of desktop capture, we have to avoid
CFI check for calls of dlopened PipeWire library. This avoid crashing
PipeWire camera backend when "is_official_build=true" option is used as
this turns on "is_cfi=true" enabling control flow integrity.
Bug: chromium:354776214
Change-Id: I7a9fc1c2d77c4ee0e8fe0586369b7246e0bb9180
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358103
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#42706}
To allow delete old signature of the AudioEncoderOpus::MakeAudioEncoder function and thus guarantee Opus AudioEncoder always has an Environment
Bug: webrtc:343086059
Change-Id: Ib660678aeb5a549dddd1dffa3d8c28b2ec6b9d0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356981
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42690}
Old copy of the header and some previous usage is kept around
for compatibility with downstream projects for now.
Bug: chromium:345101934
Change-Id: Icbe42fb8450d3a4115799438d209da4eda127bab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357441
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42681}
This is a reland of commit 844225a76a98aa3be5aca09c19ab72a5e7b6c38a
Original change's description:
> Fix 'Image will be cropped if WindowCapturerWinGdi used'
>
> Bug: webrtc:15719
> Change-Id: I7daf8ee5b90fbe9f1246f1d99211ffa0d8a19f73
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330780
> Reviewed-by: Alexander Cooper <alcooper@chromium.org>
> Commit-Queue: Alexander Cooper <alcooper@chromium.org>
> Cr-Commit-Position: refs/heads/main@{#41503}
Bug: webrtc:15719
Change-Id: Idbb2f4dcc8811d3b2b763a49adc7a57535b3d1b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334380
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42666}
Libvpx was adjusted to support scenarios test verifies, but WebRTC tests were forgotten.
Bug: webrtc:42223649
Change-Id: I19a10c939d844d00dd564bc0a16fe21844cc7cfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357680
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42665}
Currently this class assumed that if the same RTP sequence number is unwrapped again result would be the same.
That might not be true when several packets were inserted in between these two calls and unwrapper changed its state
This CL propose instead to unwrap once, and save the result in the intermediate struct.
To minimize the change and the risk, only redundant unwrapping is replaced to use unwrapped sequence number
Bug: webrtc:353565743
Change-Id: I8a18c8c206a0e16010951cabcf81dd9cb1588eda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357660
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42662}
a frame must be (or should be) first when it contains either SPS (but not just PPS),
is an IDR or is a slice with first_mb_in_slice == 0.
Fixes an edge case where a STAP-A with SPS, PPS and multiple slices of an IDR fit
into a single RTP packet which can happen with small 320x196 frames
BUG=webrtc:352379280,webrtc:346608838
Change-Id: Ic6dea6c81db759d0d7ddd4054407103fd791f6c5
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357121
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42652}
In preparation for upcoming changes in GetSimulcastConfig(), which will require a vector of stream resolutions instead of just the max resolution as an input, switch tests to use CreateEncoderStreams() instead of calling GetSimulcastConfig() directly.
Bug: webrtc:351644568, b/352504711
Change-Id: I541dd54a21a8b75028cff07a250f858a47898223
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357400
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42648}
All known users are updated to use ntp_time_util.h directly
Bug: webrtc:343076000
Change-Id: I7229b9e5dd72d83bfd98ba4050ae7583d792575b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357300
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42645}
This is a cleanup of simulcast.cc. max_qp is not needed to decide simulcast config. Move setting of max QP in VideoStream one level up, to EncoderStreamFactory::CreateEncoderStreams(), where it can be set per stream.
Bug: webrtc:351644568, b/352504711
Change-Id: Ia0e3e9d90032383574dc8867b30d362e9c5df7e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357102
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42634}
This is a cleanup of simulcast.cc. bitrate_priority is not needed to decide simulcast config. Move setting of bitrate priority in VideoStream one level up, to EncoderStreamFactory::CreateEncoderStreams().
Bug: webrtc:351644568
Change-Id: I002d728ccf8d141fe4bbb32b390129ce57c830cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357101
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42629}
ScreenCapturerSCK uses some fields that were not available in macOS 13
but the code compiles with the older SDK because of missing annotations
that were added in the macOS 15 SDK.
Bug: chromium:351843815
Change-Id: Ic1a89b4cab43d6ee81d447ccc33ef94439752c45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356860
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Lambros Lambrou <lambroslambrou@chromium.org>
Cr-Commit-Position: refs/heads/main@{#42624}
Current thresholds were tuned to guarantee no buffer overshoot in an extreme scenario (encoding a high complexity video in a low bitrate).
Bug: b/337757868, webrtc:351644568
Change-Id: I832b2564af6f18f06550338cc9b3618f8acdf831
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356580
Reviewed-by: Dan Tan <dwtan@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42620}
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}
