In most cases we can just return a frame referencing the buffer
returned by the decoder.
Bug: webrtc:7760
Change-Id: I0b42ab9662b39149e42a3c83adfd38a9d80e0e30
Reviewed-on: https://chromium-review.googlesource.com/544299
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18824}
Some .proto files have newline at the end. This CL levels all our .proto
files. A presubmit check will follow.
NOTRY=True
TBR=minyue@webrtc.org
Bug: None
Change-Id: I988fe94c31abf91c85a45b564c488329d677b958
Reviewed-on: https://chromium-review.googlesource.com/552137
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18823}
Reland the base->rtc_base without adding stub headers (will be
done in follow-up CL). This preserves git blame history of all files.
BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org
Change-Id: Iea3bb6f3f67b8374c96337b63e8f5aa3e6181012
Reviewed-on: https://chromium-review.googlesource.com/554611
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18821}
Will reland in two different commits to preserve git blame history.
BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org
Change-Id: I550da8525aeb9c5b8f96338fcf1c9714f3dcdab1
Reviewed-on: https://chromium-review.googlesource.com/554610
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18820}
Reorganized the targets in webrtc/sdk/BUILD.gn so that the applications which use
WebRTC DataChannel only can depend on the "peerconnection_factory_no_media"
instead of "rtc_sdk_objc" to reduce the binary size.
Provided a no-media implementation of RTCPeerConnectionFactory using the macro
"HAVE_NO_MEDIA".
BUG=webrtc:7613
Review-Url: https://codereview.webrtc.org/2944643002
Cr-Commit-Position: refs/heads/master@{#18819}
This failed on the Memcheck bot due to different stack signature.
Widening the suppression should fix that.
BUG=webrtc:5988
TBR=pthatcher@webrtc.org
Change-Id: Ia448d0f157d650e3ab6d4b02b3acbac91c62d1cd
Reviewed-on: https://chromium-review.googlesource.com/553377
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18818}
This is needed after 6776518bea
It wasn't detected since it was a build-only change,
the TSan trybot wasn't run.
BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org
Change-Id: Idb8e71bc302349a55b729174e01c4824f707a8d7
Reviewed-on: https://chromium-review.googlesource.com/553358
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18817}
This refactoring takes a careful approach to avoid rushing the change:
* stub headers are left in all the old locations of webrtc/base
* existing GN targets are kept and now just forward to the moved ones
using public_deps.
The only exception to the above is the base_java target and its .java files,
which were moved to webrtc/rtc_base right away since it's not possible
to use public_deps for android_library.
To avoid breaking builds, a temporary Dummy.java file was added to
the new intermediate target in webrtc/rtc_base:base_java as well to avoid
hitting a GN assert in the android_library template.
The above approach should make the transition smooth without breaking
downstream.
A helper script was created (https://codereview.webrtc.org/2879203002/)
and was run like this:
stub-headers.py -s webrtc/base -d webrtc/rtc_base -i 7634
stub-headers.py -s webrtc/base/numerics -d webrtc/rtc_base/numerics -i 7634
Fixed invalid header guards in the following files:
webrtc/base/base64.h
webrtc/base/cryptstring.h
webrtc/base/event.h
webrtc/base/flags.h
webrtc/base/httpbase.h
webrtc/base/httpcommon-inl.h
webrtc/base/httpcommon.h
webrtc/base/httpserver.h
webrtc/base/logsinks.h
webrtc/base/macutils.h
webrtc/base/nattypes.h
webrtc/base/openssladapter.h
webrtc/base/opensslstreamadapter.h
webrtc/base/pathutils.h
webrtc/base/physicalsocketserver.h
webrtc/base/proxyinfo.h
webrtc/base/sigslot.h
webrtc/base/sigslotrepeater.h
webrtc/base/socket.h
webrtc/base/socketaddresspair.h
webrtc/base/socketfactory.h
webrtc/base/stringutils.h
webrtc/base/testbase64.h
webrtc/base/testutils.h
webrtc/base/transformadapter.h
webrtc/base/win32filesystem.h
Added new header guards to:
sslroots.h
testbase64.h
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
R=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2877023002 .
Cr-Commit-Position: refs/heads/master@{#18816}
Was only working when the nonstandard "renomination" extension to ICE
is enabled, which chromium doesn't use.
BUG=chromium:734094
Review-Url: https://codereview.webrtc.org/2957303002
Cr-Commit-Position: refs/heads/master@{#18814}
Rationale:
1. sinks_ is not properly differentiated from rsid_sinks_.
2. Consistency with RtcpDemuxer.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2958283002
Cr-Commit-Position: refs/heads/master@{#18813}
Having members in a class which only exist when certain compliation flags are turned on (unless relating to the target platform) means that those flags must be the same when compiling the module as when including its headers from other modules. This means that code outside of WebRTC runs the risk of misjudging the size of an rtc::CriticalSection, or any class which has an rtc::CriticalSection as a member. (This rule is applied recursively.) If a mismatch occurs, memory corruption is likely.
Having discussed this a bit, we have decided that the simplest solution is probably the best - always define those members, even when compilation flags (namely, CS_DEBUG_CHECKS) render it unused.
BUG=webrtc:7867
Review-Url: https://codereview.webrtc.org/2957753002
Cr-Commit-Position: refs/heads/master@{#18811}
It allows to use rtc::Buffer in templates that expect std container,
e.g. it can now be used as ::testing::ElementsAreArray parameter
Bug: None
Change-Id: I97d7ffb13393d02850ddb213f7a1d01129b10b05
Reviewed-on: https://chromium-review.googlesource.com/539635
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18809}
A seen in https://codereview.webrtc.org/2956153004/ the CQ
is picky about the ending ;, so this CL removes it.
BUG=None
TBR=ehmaldonado@webrtc.org
NOTRY=True
Change-Id: I5e6359f0966f171c98225a982da042cc7147f765
Reviewed-on: https://chromium-review.googlesource.com/552138
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Edward Lesmes <ehmaldonado@chromium.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18807}
This is in preparation of upgrading to libsrtp 2.1.0.
BUG=webrtc:7856
Review-Url: https://codereview.webrtc.org/2958123002
Cr-Commit-Position: refs/heads/master@{#18805}
The conversion code was wrong because it assumed the 3x3 matrix is a
XYZ-matrix when it really is XYW-matrix. We have to override the matrix
for I420 frames to flip the vertically before rendering.
R=magjed@webrtc.org
Bug: webrtc:7760
Change-Id: I1f08c1a929bf5721706e2a902701100cf7a9c31d
Reviewed-on: https://chromium-review.googlesource.com/541346
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18801}
Adds a timeout to the dequeue input buffer call. This improves stability
because WebRTC quickly queues frames multiple when the call starts. This
might cause the decoder to run out of input buffers. Waiting for
dequeueOutputBuffers call is no longer necessary.
Bug: webrtc:7760
Change-Id: I503ff1cf44042c4d8610077090148d9dfef169f5
Reviewed-on: https://chromium-review.googlesource.com/548357
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18800}
This will make this bot be triggered for autoroll CLs.
BUG=None.
TBR=ehmaldonado@webrtc.org
NOTRY=True
Change-Id: I6d8920a4c10a5dee36e5cb93a1e3cb6e3a2ed7ae
Reviewed-on: https://chromium-review.googlesource.com/551717
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18798}
A recent update of Windows 10 blocks IDXGIAdapter::EnumOutputs() in session 0,
so ScreenCapturerWinDirectx::IsSupported() always returns false in session 0. We
should ensure ScreenCapturerWinDirectx can respond correctly in session 0.
Meanwhile, this change looses the requirement of DirectX capturer: it still
works if some of the video adapters do not support DirectX 11 or
IDXGIOutputDuplication. This issue usually happens when handling a virtual video
adapter.
BUG=webrtc:7809
Review-Url: https://codereview.webrtc.org/2937663003
Cr-Commit-Position: refs/heads/master@{#18797}
This change also wires up the rest of the production code in
webrtc/sdk/android to be built when the directory is a dependency.
BUG=webrtc:7613
NOTRY=True
Change-Id: Ideda181970a5a570c3f8148b033e471e926243d1
Reviewed-on: https://chromium-review.googlesource.com/548038
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18791}
It was used to force a codec to the top of a list, but it resulted in
"a < a" being true, which some C++ runtimes complain about.
BUG=None
TBR=pthatcher@webrtc.org
Review-Url: https://codereview.webrtc.org/2963543002
Cr-Commit-Position: refs/heads/master@{#18786}
Dont do a normal AimdRateControlUpdate update after a probe. Only set result.updated if the bitrate estimate has changed.
BUG=webrtc:7866
Review-Url: https://codereview.webrtc.org/2949203002
Cr-Commit-Position: refs/heads/master@{#18785}
This CL corrects the initialization of the AEC3, as well
as for the other submodules in the whole audio processing module
in the sense that it properly update the submodule states also
for the case when reinitialization is trigger from the render
side of the audio processing module.
Bug: chromium:736889,webrtc:7879
Change-Id: I423e963835d0c3227caa8e186b29031bcb912515
Reviewed-on: https://chromium-review.googlesource.com/549315
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18784}
Reenable hanging tests on Mac.
Deadlock happened because the following locks were grabbed by two threads at the end of a test:
Thread 1:
CapturedFrameForwarder::AddOrUpdateSink() locks CapturedFrameForwarder::crit_ and calls
FrameGeneratorCapturer::AddOrUpdateSink() what tries to lock FrameGeneratorCapturer::lock_.
Thread 2:
FrameGeneratorCapturer::InsertFrame() locks FrameGeneratorCapturer::lock_ and calls
CapturedFrameForwarder::OnFrame() which tries to lock CapturedFrameForwarder::crit_.
So two threads are locking two same locks in different orders which may cause deadlock.
BUG=webrtc:7870
Review-Url: https://codereview.webrtc.org/2955083002
Cr-Commit-Position: refs/heads/master@{#18783}
Patch set 1 is a reland + trivial rebase.
Patch set >= 2 contains bug fixes.
> Original issue's description:
> > Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended.
> >
> > That however exposes a bunch of failed test, so this CL also fixed a few other things:
> > * FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
> > * FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
> > * Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
> > * Fix test
> >
> > BUG=7664
> >
> > Review-Url: https://codereview.webrtc.org/2883963002
> > Cr-Commit-Position: refs/heads/master@{#18473}
> > Committed: 6431e21da6
BUG=webrtc:7664
Review-Url: https://codereview.webrtc.org/2953053002
Cr-Commit-Position: refs/heads/master@{#18782}
This CL corrects the computation of the delay headroom so that
it is only updated when the delay is updated. This is important
as otherwise a too large headroom will be reported, which then
could cause buffer access issues.
Bug: webrtc:7878, chromium:736893
Change-Id: Ib37cb608b064dd5d4df3f8fc423448ee80ed0106
Reviewed-on: https://chromium-review.googlesource.com/549335
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18781}
The iOS 11 SDK adds nullability annotations to several framework functions
and in this it added the _Nonnull specifier to a protocol method that
we implement. We were passing nil to that method in a test.
The warning is now fixed by passing a mock object instead of nil.
Bug: webrtc:7883
Change-Id: I9f64b0c59750629ca3969400aa725729bb10541b
Reviewed-on: https://chromium-review.googlesource.com/549927
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Kári Tristan Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18780}
Tests that still need to be enabled:
* apprtcmobile_tests
* audio_decoder_unittests
* rtc_media_unittests
* rtc_unittests
* webrtc_nonparallel_tests
Bug: webrtc:7155
Change-Id: Ic3bc91224dff8b32dc71e356a4ba04b386c92a0f
Reviewed-on: https://chromium-review.googlesource.com/541368
Commit-Queue: Oleh Prypin <oprypin@chromium.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18775}
