Ownership of EglBase is moved to PeerConnectionClient.
BUG=webrtc:8135
Review-Url: https://codereview.webrtc.org/3007893002
Cr-Commit-Position: refs/heads/master@{#19634}
directives in our DEPS files are not needed anymore.
Includes from webrtc/rtc_base are also whitelisted in webrtc/DEPS
so we don't have to whitelist it in all the others DEPS files.
BUG=webrtc:7634
NOTRY=True
Review-Url: https://codereview.webrtc.org/3006583002
Cr-Commit-Position: refs/heads/master@{#19601}
This CL replaces:
namespace webrtc_jni {
with:
namespace webrtc {
namespace jni {
The main benefit is that we don't have to use the webrtc:: qualifier
inside the jni namespace so we can reduce some clutter.
BUG=None
Review-Url: https://codereview.webrtc.org/3009613002
Cr-Commit-Position: refs/heads/master@{#19569}
It would leave a trailing carriage return character in the payload
list, causing the preferred codec to appear twice if it was at the
back of the list originally. This causes problems down the line and
results in that codec not being negotiated successfully.
BUG=webrtc:8129
Review-Url: https://codereview.webrtc.org/3001363002
Cr-Commit-Position: refs/heads/master@{#19552}
The field trials enables producing new VideoFrames in camera classes.
This field trial should be enabled if VideoSinks are used.
BUG=webrtc:7749, webrtc:7760
Review-Url: https://codereview.webrtc.org/2984633002
Cr-Commit-Position: refs/heads/master@{#19467}
This resolves an issue where setting field trials from AppRTCMobile
would not affect WebRTC Core as the two are linked with different
instances of the field_trials binary.
BUG=webrtc:8106
Review-Url: https://codereview.webrtc.org/2997023002
Cr-Commit-Position: refs/heads/master@{#19372}
This flag enables support for Android Studio 3.0 which allows us to use
Java 8 features. Gradle is updated to version 4.1.0.
BUG=webrtc:8084
Review-Url: https://codereview.webrtc.org/2994123002
Cr-Commit-Position: refs/heads/master@{#19319}
The existing unity plugin (an example in webrtc codebase) does not support camera access on Android platform. This CL implements such functionality.
TBR=gyzhou@chromium.org
BUG=webrtc:8067
Review-Url: https://codereview.webrtc.org/2993273002
Cr-Commit-Position: refs/heads/master@{#19277}
This CL was modified from work of sharifferdous@ (intern supervised by lliuu@)
BUG=webrtc:7389
Review-Url: https://codereview.webrtc.org/2987723002
Cr-Commit-Position: refs/heads/master@{#19146}
We currently don't close the peerconnection before deallocing. That
could potentially cause race conditions if it's still being processed on
other threads.
BUG=webrtc:7976
Review-Url: https://codereview.webrtc.org/2976983002
Cr-Commit-Position: refs/heads/master@{#19121}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2976293002
Cr-Commit-Position: refs/heads/master@{#19094}
The surface view renderer size was set to match parent so it couldn't
adjust based on the frame size. The size is now set to wrap_content
which allows the renderer to adjust. The root element of the call
activity is changed to FrameLayout to allow the renderer to center.
requestLayout is added to SurfaceView setScalingType so onMeasure gets
called again.
BUG=webrtc:7901
Review-Url: https://codereview.webrtc.org/2978173002
Cr-Commit-Position: refs/heads/master@{#19073}
Reason for revert:
Seems to be causing new crashes, possibly because of changes to the "Destroy(false)" behavior. Will re-land after investigating these crashes more and addressing the root cause.
Original issue's description:
> Delete SignalThread class.
>
> Rewrite AsyncResolver to use PlatformThread directly, not
> SignalThread, and update includes of peerconnection client to not
> depend on signalthread.h.
>
> BUG=webrtc:6424,webrtc:7723
>
> Review-Url: https://codereview.webrtc.org/2915253002
> Cr-Commit-Position: refs/heads/master@{#18833}
> Committed: bc8feda1dbTBR=tommi@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
NOPRESUBMIT=true
NOTRY=true
BUG=webrtc:6424,webrtc:7723
Review-Url: https://codereview.webrtc.org/2979733002
Cr-Commit-Position: refs/heads/master@{#18980}
Suppressing lint errors using comments is an undocumented feature of the
linter, and suppressing using the tools:ignore attribute should be
preferred.
Suppressing using comments becomes a problem when using the manifest
merger introduced in
6ada47bc79
as it reformats the comments slightly:
<!--suppress MissingPrefix -->
becomes
<!-- supress MissingPrefix -->
which causes the linter to disregard the suppression.
Bug: 740657
Change-Id: I8e365744d089271c390254e7c958b24b81043766
Reviewed-on: https://chromium-review.googlesource.com/566860
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Ingemar Ådahl <ingemara@opera.com>
Cr-Commit-Position: refs/heads/master@{#18971}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2973183002
Cr-Commit-Position: refs/heads/master@{#18948}
This change extends the definition of wired headset to also include USB
devices. The effect is that audio will now be routed to USB audio devices
when used in combination with AppRTCMobile.
BUG=webrtc:7931
Review-Url: https://codereview.webrtc.org/2971613003
Cr-Commit-Position: refs/heads/master@{#18889}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
Leaving compatibility script in webrtc/tools/compare_videos.py to
avoid breaking our video quality tests in Chromium.
Forwarding GN targets are left in webrtc/tools/BUILD.gn.
BUG=webrtc:7855
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2965593002
Cr-Commit-Position: refs/heads/master@{#18848}
Rewrite AsyncResolver to use PlatformThread directly, not
SignalThread, and update includes of peerconnection client to not
depend on signalthread.h.
BUG=webrtc:6424,webrtc:7723
Review-Url: https://codereview.webrtc.org/2915253002
Cr-Commit-Position: refs/heads/master@{#18833}
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}
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}
After returning from the call the AVAudioSession was configured to
use the receiver instead of the speaker for audio output. The
configuration was only restored if the sound loop was previously
playing, this change makes sure that the configuration is always
reset so the sound can be played audibly after a call has been
finished.
Bug: webrtc:7792
Change-Id: Idabf6fadc8041b18722cb8f5e89e0c8c36b1b74d
Reviewed-on: https://chromium-review.googlesource.com/544819
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18754}
All setting switches except "Loopback mode" is now in the Settings
screen instead of the main screen. They are also persisted across app
launches.
Bug: webrtc:7748
Change-Id: Iafd84e5e39639770118e2503148d1bf7fb9c3d8d
Reviewed-on: https://chromium-review.googlesource.com/527034
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18626}
For devices with multiple cameras, all supported resolutions from both
the front-facing and back cameras are listed.
Bug: webrtc:7783
Change-Id: I228eda28ea48181c86d344413dda9f3a71b0864f
Reviewed-on: https://chromium-review.googlesource.com/529045
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18533}
We want the example app to only link agains the framework. This ensures
that we are actually testing the framework, and that AppRTCMobile
doesn't require any other parts of WebRTC not included in the framework.
Bug: webrtc:7759
Change-Id: Ib04aae0bc3ab2a1a508eaf4a4f15c2d37f521598
Reviewed-on: https://chromium-review.googlesource.com/522722
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Kári Tristan Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18470}
ScheduledExecutorService silently ignores exceptions thrown by the
runnable. This makes debugging issues unnecessarily difficult.
Bug: None
Change-Id: I7deb43b96e5639c096b9aed9c6ff9b197b62f59f
Reviewed-on: https://chromium-review.googlesource.com/521084
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18378}
