Reason for revert:
The openmax_dl include change breaks downstream projects.
Original issue's description:
> Add check_deps rules in DEPS files.
>
> Add fine-grained check_deps rules for all of WebRTC.
> This will help both maintaining sane dependencies and provides a way
> to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
>
> Example:
> buildtools/checkdeps/graphdeps.py --root=. --format=png \
> --out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
> --excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
>
> will produce a neat webrtc.png image showcasing the dependencies
> (according to the DEPS file) for the bitrate_controller module.
> Some dependencies are filtered out for readability.
>
> BUG=webrtc:5623
> TESTED=Passing runs using:
> buildtools/checkdeps/checkdeps.py --root=. talk
> buildtools/checkdeps/checkdeps.py --root=. webrtc
>
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/086f851b7b9b4bcbd4fe507c3bf83b760bd7f4d9
> Cr-Commit-Position: refs/heads/master@{#12008}
TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5623
Review URL: https://codereview.webrtc.org/1808573002
Cr-Commit-Position: refs/heads/master@{#12009}
Add fine-grained check_deps rules for all of WebRTC.
This will help both maintaining sane dependencies and provides a way
to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
Example:
buildtools/checkdeps/graphdeps.py --root=. --format=png \
--out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
--excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
will produce a neat webrtc.png image showcasing the dependencies
(according to the DEPS file) for the bitrate_controller module.
Some dependencies are filtered out for readability.
BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1796413002 .
Cr-Commit-Position: refs/heads/master@{#12008}
Instead of using a raw pointer output parameter. This is a good idea
in general, but will also be very convenient when scoped_ptr is gone,
since unique_ptr doesn't have an .accept() method.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1799233002
Cr-Commit-Position: refs/heads/master@{#12004}
This CL adds the width and height to the createPbufferSurface exception
message. Also, when a Callable.call() fails in
ThreadUtils.invokeUninterruptibly() we rethrow a new exception, but that
excludes the callstack from Callable.call(). This CL adds the callstack
from Callable.call() to make debugging easier.
BUG=b/27581640,b/27516991
Review URL: https://codereview.webrtc.org/1780183005
Cr-Commit-Position: refs/heads/master@{#11996}
Devices exist (specifically the 16GB 5th generation iPod Touch) that do not have
a rear-facing camera.
This CL:
- Adjusts RTCAVFoundationVideoCapturerInternal so initialization doesn't fail
because a rear-facing camera doesn't exist, but still logs a warning
- Provides a check for whether or not a rear-facing camera can be used (useful
for toggling UI elements)
- Turns an attempt to switch to the rear-facing camera into a no-op with a
warning.
BUG=
Review URL: https://codereview.webrtc.org/1416303003
Cr-Commit-Position: refs/heads/master@{#11992}
This CL changes the interface by adding a SurfaceTextureHelper argument
to VideoCapturer.startCapture(). This removes the need for the
VideoCapturer to create the SurfaceTextureHelper itself. This also means
that it is no longer necessary to send an EGLContext to the
VideoCapturerAndroid.create() function.
The SurfaceTextureHelper is now created in AndroidVideoCapturerJni, and
the EGLContext is passed from PeerConnectionFactory in
nativeCreateVideoSource().
Another change in this CL is that the C++ SurfaceTextureHelper creates
the Java SurfaceTextureHelper instead of getting it passed as an
argument in the ctor.
BUG=webrtc:5519
Review URL: https://codereview.webrtc.org/1783793002
Cr-Commit-Position: refs/heads/master@{#11977}
and signaling the remote side to remove its remote candidate by setting the candidate priority to 0.
BUG=
Review URL: https://codereview.webrtc.org/1648813004
Cr-Commit-Position: refs/heads/master@{#11958}
The old code insists on exact cipher suite matches with hardwired expectations. It does this matching parameterized with key type (RSA vs ECDSA) and TLS version (DTLS vs TLS and version 1.0 vs 1.2).
This CL changes things to check against a white-list of cipher suites, with the check parameterized with key type (again RSA vs ECDSA). Then separately checks TLS version since the old implicit check of TLS version by means of resulting cipher suite was too blunt.
Using a white list for cipher suites isn't perfect, but it is safe and requires minimal maintenance. It allows compatibility with not just one exact version of underlying crypto lib, but any version with reasonable defaults.
The CL also re-enables critical tests which had to be disabled recently to allow a boringssl roll.
BUG=webrtc:5634
Review URL: https://codereview.webrtc.org/1774583002
Cr-Commit-Position: refs/heads/master@{#11951}
This CL also adds control flag in webrtcsession_unittests
that says whether to prefer constraints APIs or non-constraints APIs, and uses it in the test that was needed
to uncover the bug.
BUG=webrtc:4906
Review URL: https://codereview.webrtc.org/1775033002
Cr-Commit-Position: refs/heads/master@{#11947}
This enabled us to be able to remove VideoTrack::GetSink and RemoteVideoCapturer.
Since video frames from the decoder is delivered on a media engine internal thread, VideoBroadCaster must be made thread safe.
BUG=webrtc:5426
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1765423005 .
Cr-Commit-Position: refs/heads/master@{#11944}
Reason for revert:
Frame rate and freeze detection not working properly after switchCamera(). This is because the previous cameraObserver is not removed before posting a new one.
Original issue's description:
> VideoCapturerAndroid: Use one thread per startCapture()/stopCapture() session
>
> Currently, VideoCapturerAndroid sets the thread and handler in the ctor
> and clears them in dispose(). This CL sets the handler in startCapture()
> instead and clears it in stopCapture(). The purpose is to prepare for
> sending in the SurfaceTextureHelper in startCapture() instead of letting
> VideoCapturerAndroid create it in the ctor.
>
> All access to the handler is now synchronized by a lock, and all
> Runnables are posted with a token so that they can be removed all at
> once in stopCapture() to guarantee that no pending operation will be
> executed after stopCapture().
>
> BUG=webrtc:5519
>
> Committed: https://crrev.com/9cbebee523dbd280a4f67ad414a432ed730f241f
> Cr-Commit-Position: refs/heads/master@{#11939}
TBR=perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5519
Review URL: https://codereview.webrtc.org/1777253002
Cr-Commit-Position: refs/heads/master@{#11941}
Currently, VideoCapturerAndroid sets the thread and handler in the ctor
and clears them in dispose(). This CL sets the handler in startCapture()
instead and clears it in stopCapture(). The purpose is to prepare for
sending in the SurfaceTextureHelper in startCapture() instead of letting
VideoCapturerAndroid create it in the ctor.
All access to the handler is now synchronized by a lock, and all
Runnables are posted with a token so that they can be removed all at
once in stopCapture() to guarantee that no pending operation will be
executed after stopCapture().
BUG=webrtc:5519
Review URL: https://codereview.webrtc.org/1763673002
Cr-Commit-Position: refs/heads/master@{#11939}
This involves changing a few verification functions for frames
received so that they always accept the result if there's no stream.
BUG=
Review URL: https://codereview.webrtc.org/1772353002
Cr-Commit-Position: refs/heads/master@{#11937}
Adds setConfiguration back and renames statsId back to reportId.
BUG=
Review URL: https://codereview.webrtc.org/1778033002
Cr-Commit-Position: refs/heads/master@{#11936}
It may be null if the network is unknown.
Also revised the logging to replace network id with network.toString(). They are pretty much the same for logging but network.toString does not need to parse the int value.
BUG=
Review URL: https://codereview.webrtc.org/1774343002
Cr-Commit-Position: refs/heads/master@{#11925}
At the top level, setting a track on an RtpSender is equivalent to
setting a source (previously called a renderer)
on a voice send stream. An RtpSender without a track
is not supposed to send data (not even muted data), so a send stream without
a source shouldn't send data.
Also replacing SendFlags with a boolean and implementing "Start"
and "Stop" methods on AudioSendStream, which was planned anyway
and simplifies this CL.
R=pthatcher@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1741933002 .
Cr-Commit-Position: refs/heads/master@{#11918}
switchCamera() only calls stopCaptureOnCameraThread(), not
stopCapture(), so the stopListening() call must be placed there.
BUG=webrtc:5519,b/27497950
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1770423002 .
Cr-Commit-Position: refs/heads/master@{#11917}
VideoTrackSource will be implemented in an upcoming cl but is needed to be included in libjingle.gyp in Chrome before the cl can be landed.
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1769343003 .
Cr-Commit-Position: refs/heads/master@{#11897}
Moved VideoSourceInterface to MediaStreamInterface.h
Renamed VideoSourceTest to VideoCapturerTrackSourceTest
Renamed VideoSource to VideoCaptureTrackSource and cl lint and cl format.
BUG=webrtc:5426
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1770003002 .
Cr-Commit-Position: refs/heads/master@{#11893}
Changing from:
virtual void RequestIdentity(
rtc::KeyParams key_params,
rtc::Optional<uint64_t> expires,
const rtc::scoped_refptr<DtlsIdentityRequestObserver>& observer);
to:
virtual void RequestIdentity(
const rtc::KeyParams& key_params,
const rtc::Optional<uint64_t>& expires_ms,
const rtc::scoped_refptr<DtlsIdentityRequestObserver>& observer);
Making FakeDtlsIdentityStore DCHECK that |expires_ms| is not set, since it does not support that parameterization.
In a follow-up chromium CL the new signature will be used.
BUG=webrtc:5092, chromium:544902
Review URL: https://codereview.webrtc.org/1766673002
Cr-Commit-Position: refs/heads/master@{#11892}
Also change the type of "time interval" to int from uint32.
Fixed a few TODO therein. I think we should have the following convention:
1. All time delay/intervals should have type int although the time instant should have time uint32_t.
2. "interval" is preferred to "delay" if the delay will be repeated (like rescheduling).
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1762863002 .
Cr-Commit-Position: refs/heads/master@{#11888}
Soft reset can be used when input frame resolution changes
to avoid re creating MediaCodec instance.
Instead MediaCodec is flushed and some variables are reset.
R=pbos@webrtc.org, perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1732533002 .
Cr-Commit-Position: refs/heads/master@{#11878}
This temporarily fixes a probem where renderers causes VideoSinkWants.rotation_applied=true.
The problem was introduced by https://codereview.webrtc.org/1759473003/ where VideTrackRenderes are registered to the cricket::VideoCapturer with default VideoSinkWants.
BUG=webrtc:5621
Review URL: https://codereview.webrtc.org/1764693004
Cr-Commit-Position: refs/heads/master@{#11871}
all interfaces that formerly took constraints parameters
in name=value form.
This is in preparation for making Chrome only use these
explicit interfaces.
BUG=webrtc:4906
Review URL: https://codereview.webrtc.org/1717583002
Cr-Commit-Position: refs/heads/master@{#11870}
This CL replaces the function SurfaceTextureHelper.setListener() that
could only be called once with the functions startListening() and
stopListening() that can be called multiple times. This is necessary
when the SurfaceTextureHelper will be passed to the VideoCapturerAndroid
in startCapture(). startListening() will be called in startCapture() and
stopListening() in stopCapture().
BUG=webrtc:5519
Review URL: https://codereview.webrtc.org/1755573002
Cr-Commit-Position: refs/heads/master@{#11855}
This feature is off by default and can be turned on by setting IceConfig. When turned on, we'll choose a Turn/Turn (UDP takes higher priroity) over the other types of connections while no any connection is writable. However, when there is best connection or there is pending triggered check, those will take higher priority.
BUG=webrtc:4591
Review URL: https://codereview.webrtc.org/1577233006
Cr-Commit-Position: refs/heads/master@{#11850}
This is a preparation CL. The expires param will be used in
a follow-up CL. Initially it will only be used by the
chromium implementation. Then we will either update the
webrtc implementation (DtlsIdentityStoreImpl) to use it or
we will remove that store completely as part of clean-up
work.
There are currently two versions of RequestIdentity, one
that takes KeyType and one that takes KeyParams.
The KeyType version is removed in favor of the new
KeyParams + expires version. The KeyParams version without
expires is kept as to not break chromium which currently
implements that. This is the version that can be removed in
a follow-up CL.
BUG=webrtc:5092, chromium:544902
Review URL: https://codereview.webrtc.org/1749193002
Cr-Commit-Position: refs/heads/master@{#11846}
Multiple sources with the same names forces ugly GYP hacks in
Chromium's libjingle.gyp. Rename the sources in WebRTC to
enable cleaning this up in Chromium.
To summarize:
webrtc/media/base/constants.{cc,h} -> mediaconstants.{cc,h}
webrtc/p2p/base/constants.{cc,h} -> p2pconstants.{cc,h}
This CL will require coordinating landing a roll in Chromium.
BUG=webrtc:4256
NOTRY=True
Review URL: https://codereview.webrtc.org/1750593002
Cr-Commit-Position: refs/heads/master@{#11842}
This cl copies the value of cricket::VideoCapturer::IsScreencast into
a flag in VideoOptions. It is passed on via the chain
VideortpSender::SetVideoSend
WebRtcVideoChannel2::SetVideoSend
WebRtcVideoChannel2::SetOptions
WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions
Where it's used, in
WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame, we can look it up
in parameters_, instead of calling capturer_->IsScreencast().
Doesn't touch screencast logic related to cpu adaptation, since that
code is in flux in a different cl.
Also drop the is_screencast flag from the Dimensions struct, and drop separate options argument from ConfigureVideoEncoderSettings and SetCodecAndOptions, instead always using the options recorded in VideoSendStreamParameters::options.
In the tests, changed FakeVideoCapturer::is_screencast to be a construction time flag. Generally, unittests of screencast have to both use a capturer configured for screencast, and set the screencast flag using SetSendParameters. Since the automatic connection via VideoSource and VideoRtpSender isn't involved in the unit tests.
Note that using SetSendParameters to set the screencast flag doesn't make sense, since it's not per-stream. SetVideoSend would be more appropriate. That should be fixed if/when we drop VideoOptions from SetSendParameters.
BUG=webrtc:5426
R=pbos@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1711763003 .
Cr-Commit-Position: refs/heads/master@{#11837}
Rename SetCodecAndOptions to SetCodec, it no longer sets or uses the
VideoOptions. In MediaConfig, collect the video-related flags into a
struct.
As a followup, it should be possible to delete VideoOptions from
VideoSendParameters and VideoSendStreamParameters.
TBR=pthatcher@webrtc.org
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1745003002
Cr-Commit-Position: refs/heads/master@{#11828}
