This reverts commit d2b885fd91909f1b17fb11292a8c989d5d883b22.
Reason for revert: Speculative revert for Chromium importer
Original change's description:
> Fix bug where we assume new m= sections will always be bundled.
>
> 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}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12906, webrtc:12999
Change-Id: I00179d7573f322ad539ff16cad1f85320cfb2270
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227081
Reviewed-by: Björn Terelius <terelius@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34578}
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}
