The build tools is update android sdk to 31, consistent with it.
Bug: None
Change-Id: I873d13481d24009d7b730b7adeeffd2362145ccd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263800
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37050}
WebRTC’s minSdk is 21, so all those checks are dead code.
Change-Id: I26497fd92259b66d9e5ac6afbb393adf4d904c77
Bug: webrtc:13780
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253124
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Linus Nilsson <lnilsson@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36140}
This CL adds the callback on ICE Candidate Error to the Android and
the iOS SDKs.
Spec: https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-onicecandidateerror
Bug: webrtc:13446
Change-Id: I6e511aaa80f1aa8f4310d8518d1144d97470cd7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239460
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35531}
This is part of the removal of support for SDES.
Bug: webrtc:11066
Change-Id: I448d0e0032672c04c87b00550ab4b9d792071a0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234864
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35262}
* Replace "AV1X" with "AV1";
* Keep mapping of "AV1X" payload name to kVideoCodecAv1 to not break
support of injectable "AV1X".
Bug: webrtc:13166
Change-Id: I9a50481209209f3857bbf28f4ed529ee6972377e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231560
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34987}
This change does not affect downstream dependencies as androidx.annotation
is fully compatible with android.support.annotation.
Bug: webrtc:11962
Change-Id: I714b473df8d0fee8000ddf3a9beca7c5613db5ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226881
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34839}
Change-Id: I0880caa77a1097f56c560152e85c9ca29242f825
This PR add support for the `PeerConnectionObserverJni::OnRemoveTrack()`
event on Java, allowing to be notified when a remote track has been
removed. It's a very thing JNI wrapper on top of C++ API, being mostly
similar to other already available events like `track` and `addTrack`.
In Javascript API, tracks are not "removed" explicitly from the
PeerConnection, but instead receiver PeerConnection gets notified that
they have been removed from the streams they are associated to, and when
no `MediaStream` object has that track, it's considered that the track
has been removed from the PeerConnection. In Java and C++ APIs there's no
`MediaStreamObserver` class, so there's no way to listen to the
`removeTrack` event the same way happens in Javascript API, but instead
C++ API has a `removeTrack` event at PeerConnection level. This patchset
just only wraps and expose this `removeTrack` event from the C++ API to
the Java API.
This PR has been sponsored by Atos Research and Innovation
(https://atos.net/en/about-us/innovation-and-research).
Bug: webrtc:12850
Change-Id: I0880caa77a1097f56c560152e85c9ca29242f825
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218847
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34225}
After recently changing .pylintrc (see [1]) we discovered that
the presubmit check always checks all the python files when just
one python file gets updated.
This CL moves all these files one step closer to what the linter
wants.
Autogenerated with:
# Added all the files under pylint control to ~/Desktop/to-reformat
cat ~/Desktop/to-reformat | xargs sed -i '1i\\'
git cl format --python --full
This is part 1 out of 2. The second part will fix function names and
will not be automated.
[1] - https://webrtc-review.googlesource.com/c/src/+/186664
No-Presubmit: True
Bug: webrtc:12114
Change-Id: Idfec4d759f209a2090440d0af2413a1ddc01b841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32530}
renames the RTCSessionDescription object from "ѕdp" to "desc" in a few places.
The term SDP should generally refer to the blob of text described in
RFC 4566 while the RTCSessionDescription specified in
https://w3c.github.io/webrtc-pc/#rtcsessiondescription-class
contains both a type and a sdp.
BUG=None
Change-Id: Iacf332d02b03134e49c2b4147dc5725affa89741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183882
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32080}
I'll hold on to the root OWNER for a bit longer for convenience.
Bug: None
Change-Id: I13303ba726fed612adc74008eeaaeadf9595e084
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170047
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30727}
Some ErrorProne warnings have been enabled by [1], that broke the
Chromium Roll into WebRTC, this CL should have taken care of all the
problems.
[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1935889
Bug: None
Change-Id: I2670e948c320984a122fdb774b891c98e05f582e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160862
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29933}
Those are preventive annotations to prepare for incoming android update
(coming with Chromium roll).
Currently the roll is blocked partly because errorprone complains!
Bug: webrtc:11095, chromium:1003532
Change-Id: If4e2879a522e895ce7fb1f2a9ad36d06f98f2a61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160002
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29830}
Bug: webrtc:10419
Change-Id: I18528bf2526e933568bf052de76a434f012161da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148320
Commit-Queue: Alex Drake <alexdrake@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28838}
Removes CAPTURE_VIDEO_OUTPUT permission since it is not needed
when using MediaProjection API and not allowed in Android Q.
This is a reland of af4f1b4127
Original change's description:
> Target SDK level 27 in AppRTCMobile.
>
> Implements the dynamic permission model required by the newer SDK and
> changes the theme.
>
> Bug: webrtc:8803
> Change-Id: I3ea23a25b27f196fcffd018c7cdd2ff6255b62d9
> Reviewed-on: https://webrtc-review.googlesource.com/44400
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21788}
Bug: webrtc:8803
Change-Id: Ie79d7f7b9e8fea255581d3b56772b5bb6fffff4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139881
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28137}
The Legacy ADM has remained to be available for testing. Now we are
ready to move on to using only the Java ADM.
Bug: webrtc:7452
Change-Id: Ic95b04b933e165f3c16b587a44384a2c965ef16c
Reviewed-on: https://webrtc-review.googlesource.com/c/123921
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26852}
This is a propagation of upstream chromium change needed to
resume DEPS autorolls into WebRTC.
Original comment from upstream change:
> This change is made in preparation for an ErrorProne
> check to catch this at compile time. See bug for details.
Bug: chromium:771683
Change-Id: I56aed15f73a633dcadae7ece6c645cd3596f9257
Reviewed-on: https://webrtc-review.googlesource.com/c/113505
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25951}
This change makes it possible for android apps to use the new standards-compliant PeerConnectionState.
Bug: webrtc:9977
Change-Id: Iad19c38e664a59e86879715ec7a04a59a9894bee
Reviewed-on: https://webrtc-review.googlesource.com/c/109883
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25652}
Removes redundant field initializers such as null, 0 and false.
Bug: webrtc:9742
Change-Id: I1e54f6c6000885cf95f7af8e2701875a78445497
Reviewed-on: https://webrtc-review.googlesource.com/99481
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24676}
Android rules contain `assert is_android`.
This didn't cause any problems only because GN doesn't touch files if they are not referenced from the root BUILD.gn file.
Skipping presubmit because this CL triggers a warning even though it's just adding indentation.
No-Presubmit: True
Bug: None
Change-Id: Ifcb8f0e1d47784ff800507f9d560c68e8f78c717
Reviewed-on: https://webrtc-review.googlesource.com/90040
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24061}
This CL updates the WebRTC code to stop using the old VideoRenderer and
VideoRenderer.I420Frame classes and instead use the new VideoSink and
VideoFrame classes.
This CL is the first step and the old classes are still left in the code
for now to keep backwards compatibility.
Bug: webrtc:9181
Change-Id: Ib0caa18cbaa2758b7859e850ddcaba003cfb06d6
Reviewed-on: https://webrtc-review.googlesource.com/71662
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22989}
PeerConnectionFactory.initialize() should be the first call before
any other call to the Android WebRTC API. The reason this is important
is mainly because PeerConnectionFactory.initialize() loads the native
C++ code, so all other WebRTC calls that rely on native calls will fail
before this has been done.
Bug: webrtc:7474, webrtc:9153
Change-Id: Id0cb78eaf18ea036f39d616d00ac6e32696266bb
Reviewed-on: https://webrtc-review.googlesource.com/70428
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22954}
Fixes a mismatch between "useHardware" and "disableBuiltIn" when
creating JavaAudioDeviceModule.
Bug: webrtc:7452
Change-Id: Ia5572822dc4514ff9a06811af1bdbb8362a2c71c
Reviewed-on: https://webrtc-review.googlesource.com/69987
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22908}
This CL introduces sdk/android/api/org/webrtc/audio/AudioDeviceModule.java,
which is the new interface for audio device modules on Android.
This CL also refactors the main AudioDeviceModule implementation, which
is sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java and makes
it conform to the new interface. The old code used global static methods
to configure the audio device code. This CL gets rid of all that and uses
a builder pattern in JavaAudioDeviceModule instead. The only two dynamic
methods left in the interface are setSpeakerMute() and setMicrophoneMute().
Removing the global static methods allowed a significant cleanup, and e.g.
the file sdk/android/src/jni/audio_device/audio_manager.cc has been
completely removed.
The PeerConnectionFactory interface is also updated to allow passing in
an external AudioDeviceModule. The current built-in ADM is encapsulated
under LegacyAudioDeviceModule.java, which is the default for now to
ensure backwards compatibility.
Bug: webrtc:7452
Change-Id: I64d5f4dba9a004da001f1acb2bd0c1b1f2b64f21
Reviewed-on: https://webrtc-review.googlesource.com/65360
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22765}
The class called AudioDeviceModule today is an implementation of a
future interface. We want to reserve the name AudioDeviceModule for
the actual interface. The implementation class has been renamed to
JavaAudioDeviceModule. 'Java' here refers to the fact that the
implementation is using android.media.AudioRecord as input and
android.media.AudioTrack as output, and this is opposed to native
AudioDeviceModule implementations such as OpenSLES and AAudio.
Bug: webrtc:7452
Change-Id: Ifc243c2e169b12a50128ee3252f06d574aa7b358
Reviewed-on: https://webrtc-review.googlesource.com/65400
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22673}
This CL refactors the way RecordedAudioToFileController is connected to
AudioRecord. Instead of allowing to dynamically set and update the
AudioSamplesCallback, it's set once at start time and then stopping is
implemented in RecordedAudioToFileController by simply ignoring calls to
onWebRtcAudioRecordSamplesReady.
The reason for this CL is to reduce the amount of methods we need to
add to the future AudioDeviceModule interface. The more functionality
we can move to creation time in the ctor, the less methods we need to
have in the interface.
Bug: webrtc:7452
Change-Id: I462df275d8579c848e1d2c86cbd8e881da89cbf3
Reviewed-on: https://webrtc-review.googlesource.com/64988
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22653}
To facilitate testing both the old and new AudioDeviceModule path, a
setting is added to AppRTC. Enable "Use legacy audio device" to use
the old path.
Bug: webrtc:7452
Change-Id: I221378ac7bb0fa4e543c3fd081c7a322621621a0
Reviewed-on: https://webrtc-review.googlesource.com/64760
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22609}
This CL contains some follow-up fixes for
https://webrtc-review.googlesource.com/c/src/+/60541. It removes all use
of the old voiceengine implementation from AppRTCMobile.
Bug: webrtc:7452
Change-Id: Iea21a4b3be1f3cbb5062831164fffb2c8051d858
Reviewed-on: https://webrtc-review.googlesource.com/63480
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22530}
This CL adds a stand-alone Android AudioDeviceModule in the
sdk/android folder. It's forked from modules/audio_device/android/
and then simplified for the Android case. The stand-alone Android
ADM is available both in the native_api and also under a field trial
in the Java API.
Bug: webrtc:7452
Change-Id: If6e558026bd0ccb52f56d78ac833339a5789d300
Reviewed-on: https://webrtc-review.googlesource.com/60541
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22517}
This updates AppRTC to use addTrack instead of addStream, and removes
the use of onAddStream, because we no longer have to wait for this to be
fired to set the remote track's video renderers.
Bug: webrtc:8869
Change-Id: I1ecae684a9bc4b30512e8c5d717e72b52c589831
Reviewed-on: https://webrtc-review.googlesource.com/57840
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22318}
