This CL removes an unused ScopedMessageData ctor and introduces
ScopedMessageData::Release which is the first step in order to remove
the data() methods that return a reference to a std::unique_ptr (which
is an anti-pattern).
Bug: None
Change-Id: I8f3c3fcfebd127c07fe0b667ca3442a20f458f0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226870
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34563}
This allows to get encoder implementation name and other properties
without the need of initializing encoder.
Bug: none
Change-Id: I263a358d562a65a31c420ddb7c4b195316fa5ec8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226867
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34559}
Need someone from video team devs to be in the list. Working on projects
related to Android media codecs for couple of years and have enough
experience to review the changes. A concrete short-term motivation is
the need to land https://webrtc-review.googlesource.com/c/src/+/226867
Bug: none
Change-Id: I1d0a672f6b497bbe1e2d446386284d568f84664a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226951
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34556}
Acts as a compile time annotation, with corresponding run-time check
only when DCHECKs are enabled, and built using absl or pthreads mutexes.
Bug: None
Change-Id: Ie044c1ea1e576df71d634301f7df9d75cdf10b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226328
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34555}
Uppercase constants are more likely to conflict with macros (for
example rtc::SRTP_AES128_CM_SHA1_80 and OpenSSL SRTP_AES128_CM_SHA1_80).
This CL renames some constants and follows the C++ style guide.
Bug: webrtc:12997
Change-Id: I2398232568b352f88afed571a9b698040bb81c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226564
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34553}
-Winconsistent-missing-override is part of -Wall so there is no need
to explicitly set it.
-Wthread-safety and -Wimplicit-fallthrough are set by default by
Chromium's toolchain and there is no need to duplicated it in WebRTC
GN files:
gn desc out/Debug/ //rtc_base cflags | grep "Wthread-safety"
-Wthread-safety
-Wthread-safety
gn desc out/Debug/ //rtc_base cflags | grep "implicit"
-Wimplicit-fallthrough
-Wimplicit-fallthrough
Bug: None
Change-Id: Ie5104f7c6d508c7b45788420bf111a17b8b10939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226868
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#34549}
WebRTC can successfully build for arm64 Catalyst, but x64 Catalyst
still needs some work. Nevertheless, the build script can now support
it along with the existing 'simulator' and 'device' environments.
Bug: webrtc:11516
Change-Id: Ic2ce8db32142a5a0a2e50f2d8a672710b283fac3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226900
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34548}
coverity flags two issues on this:
* no copy constructor (and assignment operator) for things that free in the destructor
* missing return value check on setsockopt
BUG=None
Change-Id: I0671bf5f9bc0ede968f80c3686bf7bbd8eb63e98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226743
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34547}
Just like the C++ API, add a method in Java VideoFrame.Buffer that
describes the underlying implementation.
Use this method to properly select AndroidVideoBuffer
or AndroidVideoI420Buffer in Java -> C++ Video Frame Conversion.
Also, add a test case for WrappedNativeI420Buffer
in VideoFrameBufferTest for consistency.
Bug: webrtc:12602
Change-Id: I4c0444e8af6f6a1109bc514e7ab6c2214f1f6d60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223080
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34545}
A recent change [1] assumes that all new m= sections will share the
first BUNDLE group (if one already exists), which avoids generating
ICE candidates that are ultimately unnecessary. This is fine for JSEP
endpoints, but it breaks the following scenarios for non-JSEP endpoints:
* Remote offer adding a new m= section that's not part of any BUNDLE
group.
* Remote offer adding an m= section to the second BUNDLE group.
The latter is specifically problematic for any application that wants
to bundle all audio streams in one group and all video streams in
another group when using Unified Plan SDP, to replicate the behavior of
using Plan B without bundling. It may try to add a video stream only
for WebRTC to bundle it with audio.
This is fixed by doing some minor re-factoring, having BundleManager
update the bundle groups at offer time.
Also:
* Added some additional validation for multiple bundle groups in a
subsequent offer, since that now becomes relevant.
* Improved rollback support, because now rolling back an offer may need
to not only remove mid->transport mappings but alter them.
[1]: https://webrtc-review.googlesource.com/c/src/+/221601
Bug: webrtc:12906, webrtc:12999
Change-Id: I4c6e7020c0be33a782d3608dee88e4e2fceb1be1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225642
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34544}
As announced in the google groups [1], the pre-built Android aar is no
longer distributed and last update was August 2020. [2]
So we can remove the code that uploads aar to bintray in release_aar.py.
Still, the ability to create an Android aar and use it in a gradle
project (examples/aarproject) is useful. It can also be used to validate
aar by running PeerConnectionClientTest from examples/androidtests.
So I renamed release_aar.py to test_aar.py and make it working without
releasing the aar to an external hosting server.
This makes it easy to verify further changes to the aar.
[1] https://groups.google.com/g/discuss-webrtc/c/Ozvbd0p7Q1Y/m/TtQyRI1KAgAJ
[2] https://mvnrepository.com/artifact/org.webrtc/google-webrtc?repo=bt-google-webrtc
Bug: webrtc:11962
Change-Id: Ibe066a3a770569924e3b57805986808e1dd19df6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220622
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34540}
The alternative new name proposed, NackTracker, is already in
use in audio_coding.
Fixed: webrtc:11594
Change-Id: I6a05fafc05fa7ddb18ea4f64886a135e5ef59f7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226744
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34539}
as there are encryption schemes that preserve the payload structure
well enough and do not require those extensions.
This improves consistency as the webrtc-encoded-transform API
(which does not use this synchronous codepath) does not require those
header extensions either.
BUG=webrtc:12995
Change-Id: If237ca5d92e8871ac71c3d48fdd05127206395e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226741
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34537}
Example of error reporting:
Usage of assert() has been detected in the following files, please use
RTC_DCHECK() instead.
Files:
rtc_base/thread.cc
Bug: webrtc:6779
Change-Id: Iae08c3d7ddcc0449073752cadca19b3cf662892c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225549
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34532}
As part of go/coil update code search links to not point to the
"master" branch.
Bug: chromium:1226942
Change-Id: I0ae9e84ecc660f789a69fe0b226f93bbc39a8a66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226081
Commit-Queue: Tony Herre <toprice@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34531}
This CL completes the removal of assert() and relative headers from
the codebase (excluded
//examples/objc/AppRTCMobile/third_party/SocketRocket which is in a
third_party sub-directory).
Bug: webrtc:6779
Change-Id: I93ed57168d2c0e011626873d66529488c5f484f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225546
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34528}
