This CL adds a new library for the RNN VAD that provides (optimized)
vector math ops. The scheme is the same of the `VectorMath` class of AEC3
to ensure correct builds across different platforms.
Bug: webrtc:10480
Change-Id: I96bcfbf930ca27388ab5f2d52c022ddb73acf8e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194326
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32741}
In preparation for adding AVX2 code, a safe scheme to support
different SIMD optimizations is added.
Safety features:
- AVX2 kill switch to stop using it even if supported by the
architecture
- struct indicating the available CPU features propagated from
AGC2 to each component; in this way
- better control over the unit tests
- no need to propagate individual kill switches but just
set to false features that are turned off
Note that (i) this CL does not change the performance of the RNN VAD
and (ii) no AVX2 optimization is added yet.
Bug: webrtc:10480
Change-Id: I0e61f3311ecd140f38369cf68b6e5954f3dc1f5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193140
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32739}
AudioAttributes::getAllowedCapturePolicy was added in API Level 29.
Update WebRtcAudioTrack to add API Level check before using the API.
Bug: webrtc:12250
Change-Id: Ica6604eb1d7fa736a0e64729a022eefcfb7b3020
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195941
Commit-Queue: Gaurav Vaish <gvaish@chromium.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32735}
For some time now, calls to EnumerateCapturableWindows could lead to a
deadlock if an application's main thread is waiting on the thread that
is running EnumerateCapturableWindows. This is because calls to
GetWindowText and GetWindowTextLength send a message to the window if
the window is owned by the current process. Since the main thread is
waiting on us, it will never reply to this message and we will hang.
This happens occasionally in Chromium when tearing down the
NativeDesktopMediaList object, e.g. when a user clicks "cancel" on
the capture target picker.
We can avoid this deadlock by checking if the window we are querying
is owned by the current process, and if it is then we must ensure it
is responding to messages before we call a GetWindowText* API.
This change also adds a unit test for this scenario. We create a
window and force it to be unresponsive by creating a deadlock, and
then call GetWindowList and (with the new changes) we should not
hang. Without the new changes to GetWindowListHandler, this test
would hang.
Change-Id: I2523cd735f96fd7ea60708c30cd22e5b525803f0
Bug: chromium:1152841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195365
Commit-Queue: Austin Orion <auorion@microsoft.com>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32734}
This cl copies modules/remote_bitrate_estimator/inter_arrival.x to inter_arrival.h and interrival_delta.cc in goog_cc in the first patchset.
In the following- this class is modified to use webrtc::Timestamp and webrtc::Timedelta in order to avoid having to use 24 bit time repressentation.
Bug: none
Change-Id: I9befe6e3e283cf7e21efa974ae33e8a83e26cbe6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194004
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32733}
When building WebRTC.framework, building the XCTest test runner is a
problem because it requires Chromium's //base checkout. This workaround
allows to skip that.
No-Presubmit: True
Bug: webrtc:12134
Change-Id: I0d99bd03f27911f46679ee91b0120e7121d1c7d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196081
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32732}
This CL renames webrtc guava dependencies from
third_party/guava:guava_android_java to
//third_party/android_deps:guava_android_java
This is in preparation for deleting third_party/guava:guava_android_java
BUG=chromium:2560401
No-Presubmit: True
Change-Id: If9227f4ac4d24386896c47eeb38142a76a27a4ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195720
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32730}
As requested on bugs.webrtc.org/12096#c2, this CL adds a Chromium
metric OWNERS in order to always have their review when WebRTC's UMA
metrics are updated.
Bug: webrtc:12096
Change-Id: Icd9ab7dda5f7a4ba6ac078f667c1fd39f3314123
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191702
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32728}
Assume bitrate is evenly distributed between frames. This is wrong for
uneven frame sizes and will underestimate the overhead for simulcast.
However, it will be more correct than the current calculation,
especially for low bitrates when each frame is smaller than one packet.
This is also when overhead matters more since it is a larger fraction
of the total bitrate.
It is also unclear what will happen when using FEC.
Bug: b/166341943
Change-Id: I247b9d0fc7a8ad5daa9b577f55ec16c56efa34c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195221
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32725}
This CL breaks out descriptor specific parts into separate classes. All logic in the newly added classes is just copy pasted from the (previously massive) RtpFrameReferenceFinder with the exception of how frames are being returned, which is now done via return value rather than a callback. Basically, all interesting changes have been made in the RtpFrameReferenceFinder.
Bug: webrtc:12221
Change-Id: I5f958d2fbf4b77ba11c3c6c01d8d0d80e325be60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195448
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32717}
After upgrading to xcode 12, some Gtest tests have started to randomly
fail. The solution around this problem is to build and run GTests as
XCTests.
In order to achieve that, the CL sets enable_run_ios_unittests_with_xctest
to true in all iOS builds and adds a dependency on
//base/test:google_test_runner for each Gtest that needs to run as an
XCTest.
Real XCTest don't need the dependency and they are marked with the
rtc_test() argument `is_xctest=true` (apprtcmobile_tests, sdk_unittests
and sdk_framework_unittests).
This CL is based on [1] which passes --xctest to the runner and uses
--undefok to avoid to crash when absl/flags doesn't recognize the
flag --enable-run-ios-unittests-with-xctest (absl/flags cannot have "-"
in flags so WebRTC binaries cannot define that flag). To workaround the
issue, WebRTC tests always behave like
--enable-run-ios-unittests-with-xctest is always set (by linking only
with //base/test:google_test_runner to run iOS tests).
This fixes iOS12 and iOS13 tests but not iOS14 on which some tests
are failing because of restricted access to resources (this will be
addressed in another CL).
Long term, this solution might cause problems when Chromium decides
to update test() GN template and/or the test launcher, so WebRTC should
plan a better integration with Chromium's iOS infra.
[1] - https://chromium-review.googlesource.com/c/chromium/tools/build/+/2550656
Bug: webrtc:12134
Change-Id: I24c731dee0310e02ae1bbe6c23d359d6fcd18f17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193620
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32716}
Needed in order to return different codes for different failures
in initialization.
Sideswipe: Check TURN URL hostnames for illegal characters.
Bug: webrtc:12238
Change-Id: I1af3a37b9654b83b268304f7356049f9f3786b7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195541
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32710}
External reasons here are simulcast configuration and
source resolution change.
Initial frame dropper should be enabled in these cases because the
client can request way too big resolution for available bitrate and
usual quality scaling would take too long.
Bug: none
Change-Id: I02fbbd3c15b53b39672c083c2a1f9a780256c507
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195004
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32707}
Without this change, if the user disables QVGA and VGA streams via |active|
flags in SetParamters, the resulting stream would have too high min bitrate.
This would lead to bad performance and low quality adaptation rate.
Bug: none
Change-Id: I919a30bfb248c06747c989afe6965b3afaef2260
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195325
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32706}
This CL also adds commentary to member variables that couldn't be guarded
because they're accessed from multiple threads.
Bug: webrtc:12230
Change-Id: I5193a7ef36ab25588c76ee6a1863de6a844be1dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195331
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32705}
Delete unused macros BWE_MIN and BWE_MAX.
Move enum RateControlState: Make it a private enum class in
AimdRateControl, the only user.
Change users of the header file that only need BandwidthUsage, to
instead include api/network_state_predictor.h, the file defining this
class. As a result, fewer dependencies on
modules/remote_bitrate_estimator.
Bug: None
Change-Id: I4450c79dd58e1875d64dd74d1ae2cb7d911a14b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195222
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32704}
Instead of flushing all packets, it makes sense to flush down to the target level instead. This CL also initiates a flush when the packet buffer is a multiple of the target level, instead of waiting until it is completely full.
Bug: webrtc:12201
Change-Id: I8775147624536824eb88752f6e8ffe57ec6199cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193941
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32701}
Setting the option after calling connect but before the socket is
connected fails in some circumstances on Windows, while setting it
before connecting always succeeds. That's what Chrome is doing;
TCPClientSocket::OpenSocket calls SetDefaultOptionsForClient (which
sets TCP_NODELAY) right after opening the socket.
Also, start logging errors, and storing last error when setsockopt
fails.
Bug: webrtc:12217
Change-Id: I169d52e31b50e54e5bc93ff3590bae656cacb2b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195060
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32696}
