Added a new sub-module 'GainApplier'. The build target is
'modules/audio_processing/agc2:gain_applier'. A small refactoring
makes the GainApplier used in adaptive-digital AGC2.
The AGC2 now multiplies samples with a gain in 3 places. It's the
GainApplier, the GainCurveApplier, and the FixedGainController. The
GainApplier is used in AdaptiveDigitalGainApplier and will be used as
a pre-amplifier.
Bug: webrtc:9138
Change-Id: Ibc4c0ea109c6757f159d4adb6e3d8614179c9bc6
Reviewed-on: https://webrtc-review.googlesource.com/69321
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22849}
This reverts commit e3d522dd6b52025191bacfab241f130e9870941f.
Reason for revert: Disabling test failing in downstream projects.
Original change's description:
> Revert "Floating-point exception observer for unit tests"
>
> This reverts commit 3fb3939896f6270d48aff34eee2946bd7661bd63.
>
> Reason for revert: Downstream projects failures.
>
> Original change's description:
> > Floating-point exception observer for unit tests
> >
> > This CL adds a simple tool that let a unit test fail if a floating
> > point exception occurs. It is possible to focus on specific exceptions.
> > Note that FloatingPointExceptionObserver is only effective in debug
> > mode. For this reason, the related unit tests only run in debug mode.
> > Plus, due to some platform-specific limitations, not all the floating
> > point exceptions are available on Android.
> >
> > Bug: webrtc:8948
> > Change-Id: I0956e27f2f3aa68771dd647169fba7968ccbd771
> > Reviewed-on: https://webrtc-review.googlesource.com/58097
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22768}
>
> TBR=phoglund@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I0fd3d114ab4a348fd46339e98273e19c1ac1c6dc
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8948
> Reviewed-on: https://webrtc-review.googlesource.com/67380
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22769}
TBR=phoglund@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8948
Change-Id: I7584d941b227277a271323b47bc70945af999758
Reviewed-on: https://webrtc-review.googlesource.com/69060
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22848}
The issue is visible when reconfiguring the screen arrangement while
sharing the displays. Can sometimes be seen right after starting the
screen sharing.
Indeed CaptureFrame can be called at any time so TakeLatestFrameForDisplay
should always return a valid frame and the call should not empty the
internal container.
Also add missing teardown in the provider on failure case.
Bug: webrtc:8652
Change-Id: Ice151c1da92b9ad2b86ca9368d30d9d21114e53e
Reviewed-on: https://webrtc-review.googlesource.com/69420
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22846}
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.
Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
This CL fixes issues when applying a description with an empty BUNDLE
group (previously it would fail, after recent refactoring it started
crashing).
This CL also will cause an empty BUNDLE group to be generated when it
should be. Namely, when responding to an offer that had a BUNDLE group,
rejecting everything in it.
Bug: chromium:831996
Change-Id: I4e705a328daef4e81f8f1ace6aa73ddfa13c0107
Reviewed-on: https://webrtc-review.googlesource.com/69720
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22844}
Since Windows 10, Windows starts to support virtual desktops. The
problem is when one virtual desktop is not the current one, we can still
enumerate the windows on it, which are still marked as visible by OS.
This causes troubles to decide if a window is on top to be cropped out.
This cl is to utilize a COM API, IsWindowOnCurrentVirtualDesktop of
VirtualDesktopManager, to make sure only the windows on current desktop
will be enumerated.
Bug: chromium:796112
Change-Id: I6e0546e90fbdb37365a8d98694ded0e30791628e
Reviewed-on: https://webrtc-review.googlesource.com/65882
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22842}
This makes it easier to add new test cases without modifying the actual test class.
Bug: None
Change-Id: I48e4f14e26cd6610678ffb07ce9fd56e6bc1ac4e
Reviewed-on: https://webrtc-review.googlesource.com/69600
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22840}
The tests are a combination of the old audio_device_unittest.cc and
audio_manager_unittest.cc, with the exception of a few that were no
longer relevant.
RunPlayoutAndRecordingInFullDuplex remains disabled according to its
comment, but has been verified to pass on at least one device.
MeasureLoopbackLatency also remains disabled, but has not been tested due
to lack of necessary hardware.
Bug: webrtc:7452
Change-Id: Ie361bc8f5e1990729d7b4699faf2a73abe3cbe8d
Reviewed-on: https://webrtc-review.googlesource.com/69340
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22836}
This CL makes it possible to create a GlTextureFrameBuffer from any
thread. The actual GL resources will be allocated the first time
setSize() is called. The purpose is to be able to use 'final' variables
more often for this class and avoid @Nullable annotations.
Bug: None
Change-Id: I350304bcd33fd674990254df37a615995972f322
Reviewed-on: https://webrtc-review.googlesource.com/69241
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22835}
Per-packet info is now signaled in SentPacket to provide useful stats
for bandwidth consumption and overhead analysis in the network stack.
Bug: webrtc:9103
Change-Id: I2b8f6491567d0fa54cc559fc5a96d7aac7d9565e
Reviewed-on: https://webrtc-review.googlesource.com/66281
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22834}
This was previously not working because the answerer wasn't generating
ICE credentials when it should have been.
This was fixed inadvertently by:
https://webrtc-review.googlesource.com/c/src/+/46380
But we should really also have a PeerConnection-level regression test
for this.
Bug: webrtc:6023
Change-Id: I3da900edcc8db8034ed61a7bb981d9c0e616254e
Reviewed-on: https://webrtc-review.googlesource.com/69403
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22832}
This was working before, but somewhat by accident (because an error
wasn't being surfaced).
This CL also starts surfacing that error, from
JsepTransportController::AddRemoteCandidates to PeerConnection.
Bug: None
Change-Id: Ib48c9c00ea2a5baa5f7e3210c5dc7a339498b2d0
Reviewed-on: https://webrtc-review.googlesource.com/69015
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22830}
This changes the PeerConnection when in Unified Plan mode to reject
SDP applied with SetLocalDescription or SetRemoteDescription if the
SDP has multiple "Plan B tracks" (a=ssrc lines) in a media section.
The error is to inform developers that the given SDP will not be
interpreted as they might expect.
Bug: None
Change-Id: I7a0e11282fbf63dac06038cd22a66683517a87d0
Reviewed-on: https://webrtc-review.googlesource.com/68764
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22829}
This reverts commit 4feb2044db4b3563323c24e2372885cbf7708c73.
Reason for revert: Landscape video was not showing as aspect fit as before. .
Original change's description:
> Fix rendering on an iPhone X's tall screen.
>
> Bug: webrtc:8884
> Change-Id: I850e4ea1919837e15a78c90968a4879a1ccbd22c
> Reviewed-on: https://webrtc-review.googlesource.com/52761
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22011}
TBR=magjed@webrtc.org,kthelgason@webrtc.org,jtteh@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8884
Change-Id: I17bcdaf945d74540538162934cd3265240cc9302
Reviewed-on: https://webrtc-review.googlesource.com/68841
Commit-Queue: JT Teh <jtteh@webrtc.org>
Reviewed-by: JT Teh <jtteh@webrtc.org>
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22828}
OnRatesUpdated() is called every time the bitrate estimate, or once per
second. However, since we don't want to reconfigure libvpx too often,
just in case it interferes with the rate controller, so
ScreenshareLayers contains a boolean |bitrate_update_| which indicate
if the configuration should be updated on a call to
UpdateConfiguration().
However, it two rate updates happened between two frames, the first of
which changes the rates and second one does not, |bitrate_update_| will
be reset to false and the encoder won't get the desired config.
This CL makes sure we update the configuration iff the rate has changed
at any time since the last call to UpdateConfiguration().
Bug: webrtc:9012
Change-Id: I62af36cffe20ecb7d3f403b3eb11f23a9692d719
Reviewed-on: https://webrtc-review.googlesource.com/69040
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22826}
Not including <iterator> creates problems on some build systems because
std::back_inserter is defined there.
Bug: None
Change-Id: I27180f72dd327e3a0caab35d3a33907f1e0c4296
Reviewed-on: https://webrtc-review.googlesource.com/69323
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22825}
Contents of the template class VideoMediaChannelTest copied into
VideoMediaChannelTest, the only user.
Bug: None
Change-Id: Ie43a7c4bc1e85e2df77361f43776d4d902b74cae
Reviewed-on: https://webrtc-review.googlesource.com/67400
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22822}
The code which triggered the memcheck failure is gone.
Bug: webrtc:5989
Change-Id: I35decfae1af4e988724ababa58c41fca7a2c4a67
Reviewed-on: https://webrtc-review.googlesource.com/69261
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22821}
Mainly remove CHECKinitialized_ macro and AGC functionality. Also make
actual behavior clearer in some functions.
Bug: webrtc:7452
Change-Id: I1eac86f4eaff7b14820d3e4192b15c20ab6acb45
Reviewed-on: https://webrtc-review.googlesource.com/69161
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22820}
Currently there are several checks against _lastDrawnFrame in RTCEAGLVideoView.mm but this variable is not assigned anywhere. Seems like it was missed in 13941912b1 during work on injecting custom shaders.
Bug: webrtc:9133
Change-Id: Ie979a63de343e7253e4b4e70e3b98ffb0880af04
Reviewed-on: https://webrtc-review.googlesource.com/68720
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22819}
This reverts commit fc43d11717e16dd427ac84fee614e5511e43cefd.
Reason for revert: Crashes downstream tests
Original change's description:
> Add thread checker to PortAllocator and its subclasses and fix a bug
> causing memory contention by threads.
>
> PortAllocator and its subclasses assume all of their methods except the
> constructor must be called on the same thread (the network thread in
> practice). This CL adds a thread checker to PortAllocator and its
> subclasses for thread safety, and fixes bugs of invoking some of their
> methods in PeerConnection on the signaling thread.
>
> Bug: webrtc:9112
> Change-Id: I33ba9bae72ec09a45ec70435962f3f25cd31583c
> Reviewed-on: https://webrtc-review.googlesource.com/66945
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22814}
TBR=deadbeef@webrtc.org,pthatcher@google.com,pthatcher@webrtc.org,qingsi@google.com,honghaiz@webrtc.org
Change-Id: I2db6561d5d6366d38caa58c3e719d0d48eda70c2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9112
Reviewed-on: https://webrtc-review.googlesource.com/69200
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22818}
A part of the Audio Processing Module interface is GetStatistics. The
call collects stats from submodules. We make sure these calls are made
by the fuzzer to cover that code path.
Bug: webrtc:7820
Change-Id: Ia8f89d9838602dcb2599f676bd5c43e815bbf791
Reviewed-on: https://webrtc-review.googlesource.com/68980
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22817}
causing memory contention by threads.
PortAllocator and its subclasses assume all of their methods except the
constructor must be called on the same thread (the network thread in
practice). This CL adds a thread checker to PortAllocator and its
subclasses for thread safety, and fixes bugs of invoking some of their
methods in PeerConnection on the signaling thread.
Bug: webrtc:9112
Change-Id: I33ba9bae72ec09a45ec70435962f3f25cd31583c
Reviewed-on: https://webrtc-review.googlesource.com/66945
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22814}
Reject the local/remote description trying to change the pre-negotiated
BUNDLE tag.
Reject an answer containing a BUNDLE group that's not a subset of the offered group.
Reject an offer/answer with a BUNDLE group containing a MID that no m= section has.
Reject an answer removes an m= section from an established BUNDLE group without
rejecting it.
Bug: chromium:827917
Change-Id: If334eefb00b1c1c1e24f9afba0cb00b5867f5590
Reviewed-on: https://webrtc-review.googlesource.com/67190
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22813}
These two new histograms relate to the packet-loss concealment that
happens when audio packets are lost or late for decoding, and the
NetEq must resort to extrapolating audio from the previously
decoded data.
Bug: webrtc:9126
Change-Id: I99cc97e653169fb742da0092653ab99fd10e5d7b
Reviewed-on: https://webrtc-review.googlesource.com/67861
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22812}
This is a reland of dcc7e88cc79ab4f7aeb87c13f402e007e1320fd8
Original change's description:
> Storing frame if encoder is paused.
>
> Adds a pending frame to VideoStreamEncoder that is used to store frames
> that are not sent because the encoder is paused. If the encoder is
> resumed within 200 ms, the pending frame will be encoded and sent. This
> ensures that resuming a stream instantly starts sending frames if it is
> possible.
>
> This also protects against a race between submitting the first frame
> and enabling the encoder that caused flakiness in end to end tests
> when using the task queue based congestion controller.
>
> Bug: webrtc:8415
> Change-Id: If4bd897187fbfdc4926855f39503230bdad4a93a
> Reviewed-on: https://webrtc-review.googlesource.com/67141
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22781}
Bug: webrtc:8415
Change-Id: I0ea7d4d679e7845907cfbe9a120f128ff2180e4b
Reviewed-on: https://webrtc-review.googlesource.com/68580
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22810}
This CL changes the handling of saturated microphone signals in AEC3.
Some of the changes included are
-Make the detection of saturated echoes depend on the echo path gain
estimate.
-Remove redundant code related to echo saturation.
-Correct the computation of residual echoes when the echo is saturated.
-Soften the echo removal during echo saturation.
Bug: webrtc:9119
Change-Id: I5cb11cd449de552ab670beeb24ed8112f8beb734
Reviewed-on: https://webrtc-review.googlesource.com/67220
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22809}
The ostream operators does not work consistently with gtest since the
definitions would have to be included before the first include of gtest.
This is tricky to ensure and the end result is that the operators will
sometimes work and sometimes not without obvious explanations.
To avoid causing confusing behavior, this CL removes the operators
pending a better solution.
Bug: None
Change-Id: I66bead0efb7246d368359ddf9e9bfad9d67c05da
Reviewed-on: https://webrtc-review.googlesource.com/68640
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22808}
The static initializers caused issues with chromium build. This should
fix it for now.
Bug: None
Change-Id: I4592df0e2bd02980421bb6319c24e7b6983d2252
Reviewed-on: https://webrtc-review.googlesource.com/69020
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22807}
Set the flag based on coded length of buffered frame which is reset
after picture is encoded and, thus, is equal to zero when encoder
delivers first frame of next picture.
Before this change first_frame_in_picture was set based on index of
spatial layer of encoded frame. This is not right anymore since encoder
can drop base layer but deliver upper layers.
Bug: chromium:828350
Change-Id: I12c7534240de8bc4905f04ff368cc3704720a70b
Reviewed-on: https://webrtc-review.googlesource.com/68561
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22805}
Recent Clang versions fixed a bug which had previously allowed some casts that
removed qualifiers to go undiagnosed.
This fixes the following kind of error:
./../third_party/webrtc/api/optional.h:41:35: error: reinterpret_cast from
'const int *' to 'void *' casts away qualifiers
FunctionThatDoesNothingImpl(reinterpret_cast<void*>(x)));
^~~~~~~~~~~~~~~~~~~~~~~~~~
../../third_party/webrtc/api/optional.h:280:45: note: in instantiation of
function template specialization
'rtc::optional_internal::FunctionThatDoesNothing<const int>' requested here
return has_value_ ? *optional_internal::FunctionThatDoesNothing(&value_)
^
../../third_party/webrtc/call/rtp_bitrate_configurator.cc:70:53:
note: in instantiation of member function 'rtc::Optional<int>::value_or'
requested here
std::max(bitrate_config_mask_.min_bitrate_bps.value_or(0),
^
Bug: chromium:831081
Change-Id: I032ebd1f052fa2a50548e984febb7fa462df42ea
Reviewed-on: https://webrtc-review.googlesource.com/68941
Commit-Queue: Hans Wennborg <hans@chromium.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22804}
instead of relying on optional.h to included these 2 headers.
Bug: webrtc:9078
Change-Id: I7a4b3facd81690b8f107640487e129986c1f5ff6
Reviewed-on: https://webrtc-review.googlesource.com/68602
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22803}
The given IOSurfaceRef was ignored until now. Wrap it into the new
DesktopFrameIOSurface. The new DesktopFrameProvider object is there
to manage them as it has to be done per display id.
From initial measurement this speed-up the frame capture by 2.
Disabled by default for now but it can be enabled by calling
options.set_use_iosurface. This CL will allow to do some advanced
tests.
Bug: webrtc:8652
Change-Id: Ia9ac0b69b30098774941cb378804b45cb1710119
Reviewed-on: https://webrtc-review.googlesource.com/33014
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22801}
This reverts commit a8f13ccad400eb8ff84a379042c0595951ca9658.
Reason for revert: It's causing no video to be shown after the 1st call.
Original change's description:
> Improve thread-safety of MTL Renderer.
>
> Bug: b/77579859
> Change-Id: I427d0f41593155dc5cbf98a09d7ec826497b803c
> Reviewed-on: https://webrtc-review.googlesource.com/67040
> Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22795}
TBR=andersc@webrtc.org,kthelgason@webrtc.org
Change-Id: Ia8f33995e087178f1c3be7753f70be8ba18447f8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/77579859
Reviewed-on: https://webrtc-review.googlesource.com/68860
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22800}
