For simplicity, camera with index 0 is used. User also has to manually
give the permission to use the camera for the app.
Bug: webrtc:8769
Change-Id: I371f26f94d629411fd299671b4f3202e84556b80
Reviewed-on: https://webrtc-review.googlesource.com/76982
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23284}
Build targets involving files under api/video/ are moved into this
file, from api/BUILD.gn. In addition, drop "_api" part of target
names, and move the header file api/videosinkinterface.h to
api/video/video_sink_interface.h.
Bug: webrtc:9253
Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155
Reviewed-on: https://webrtc-review.googlesource.com/75500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23207}
Bigger buttons, fewer taps makes it less tedious to test loopback calls
locally. See webrtc:9240 for details.
Bug: webrtc:9240
Change-Id: I0dfcbc6020f27f284eae25903b2bdc1f272221b6
Reviewed-on: https://webrtc-review.googlesource.com/74583
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23137}
To prepare for making the software codecs optional and injectable, these
codec factories provide a way to pass in identical factories as were the
default old behaviour.
Bug: webrtc:7925
Change-Id: I0c70fa3c56c999e9d1af6e172eff2fbba849e921
Reviewed-on: https://webrtc-review.googlesource.com/71162
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23096}
with call to RuntimeEnvironment.application which provides the same instance.
Bug: None
Change-Id: I4e318955086dff990cb572a09c116d28a1023a34
Reviewed-on: https://webrtc-review.googlesource.com/73244
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23064}
This CL removes internal support for anything else than Android frames
that are wrapped Java VideoFrames. This allows for a big internal
cleanup and we can remove the internal class AndroidTextureBuffer and
all logic related to that. Also, the C++ AndroidVideoTrackSource no
longer needs to hold on to a C++ SurfaceTextureHelper and we can
remove all JNI code related to SurfaceTextureHelper. Also, when these
methods are removed, it's possible to let VideoSource implement the
CapturerObserver interface directly and there is no longer any need for
AndroidVideoTrackSourceObserver. Clients can then initialize
VideoCapturers themselves outside the PeerConnectionFactory, and a new
method is added in the PeerConnectionFactory to allow clients to create
standalone VideoSources that can be connected to a VideoCapturer outside
the factory.
Bug: webrtc:9181
Change-Id: Ie292ea9214f382d44dce9120725c62602a646ed8
Reviewed-on: https://webrtc-review.googlesource.com/71666
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23004}
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}
Splits out audio_java into audio_api_java and
java_audio_device_module_java.
Makes depending on java_audio_device_module_jni optional for clients
that do not use it. It is only necessary to depend on this target if
depending on java_audio_device_module_java.
Also some cleanup.
Bug: webrtc:7452
Change-Id: Ic6c4dbe11db3ed8330802a8e90203acb8ef18e72
Reviewed-on: https://webrtc-review.googlesource.com/70220
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22981}
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}
This CL splits out the audio device module Java code into a separate
target, and also splits up the audio device module implementations into
three different build targets, one for OpenSLES, AAudio, and the Java
based implementation.
Bug: webrtc:7452, webrtc:9048
Change-Id: I8ec09c73580b468837223ddd420fb29ca61fdea5
Reviewed-on: https://webrtc-review.googlesource.com/66461
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22727}
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}
jsr305 is necessary dependency for Nullable annotations.
Also adds a flag to release_aar.py to specify the build directory
manually. This makes it easier to test the script without full
recompilation.
Bug: webrtc:8881
Change-Id: Ib4b8cd4592ced9c92ca2810928bcbb6173d2164e
Reviewed-on: https://webrtc-review.googlesource.com/65081
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22671}
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}
Demonstrates how to use the iOS native API to wrap components into
C++ classes.
This CL also introduces a native API wrapper for the capturer.
The C++ code is forked from the corresponding CL for Android at
https://webrtc-review.googlesource.com/c/src/+/60540
Bug: webrtc:8832
Change-Id: I12d9f30e701c0222628e329218f6d5bfca26e6e0
Reviewed-on: https://webrtc-review.googlesource.com/61422
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22484}
Downstreams have been updated, and this now updates all uses of label()
to id() within WebRTC code. This change also makes id() pure virtual and
removes label().
Bug: webrtc:8977
Change-Id: Ib045ea4fabba6f14447c64875c7aeba87dc2be24
Reviewed-on: https://webrtc-review.googlesource.com/60382
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22431}
Add a flag to Android perf tests, so we can specify the number of
retries.
Bug: chromium:755660
Change-Id: Ic498373421b7e0fdf779a4659a0c79d47a59fbde
Reviewed-on: https://webrtc-review.googlesource.com/61103
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22390}
The app is a simple loopback demo demonstrating the usage of Android
native API. This is an initial version and I will add support for
HW codecs etc. in the future.
Bug: webrtc:8769
Change-Id: Ifb6209769dabeb8ca3185b969a1ef8afd6d84390
Reviewed-on: https://webrtc-review.googlesource.com/60540
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22385}
1, Let targets libwebrtc_unity and webrtc_unity_plugin built with Ninja -C out/***.
2, Fixed compile issue of libwebrtc_unity.
3, Built libwebrtc_unity classes into Java 7 instead of Java 8 for android.
4, Added an interface to enable peerconnectionFactory for android in Unity.
Bug: webrtc:8986
Change-Id: I2a206a77ab38895ec9ac845ce89507d61076d396
Reviewed-on: https://webrtc-review.googlesource.com/59000
Reviewed-by: Qiang Chen <qiangchen@chromium.org>
Commit-Queue: George Zhou <gyzhou@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22373}
Add native api conversions for video frames and video renderer. This
also requires some changes to sdk/BUILD to avoid cyclic dependencies.
Bug: webrtc:8832
Change-Id: Ibf21e63bdcae195dcb61d63f9262e6a8dc4fa790
Reviewed-on: https://webrtc-review.googlesource.com/57142
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22340}
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}
The naming convention according to the spec is stream id, not stream
labels.Changing things now to be spec compliant, before it is widely
used. This also includes changes to objective C wrapper code to be in
sync with the change.
Bug: webrtc:7932
Change-Id: I5705e6d8a647aaeed860316466a7320132f24b00
Reviewed-on: https://webrtc-review.googlesource.com/59301
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22316}
This removes the routing for the deprecated audio control setting
Bug: none
Change-Id: If7a134ee487b80a653ba982768ba74ce2d539e0a
Reviewed-on: https://webrtc-review.googlesource.com/58941
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22288}
This removes the routing for the deprecated audio control setting
Change-Id: Id83ff548625279d5b34c9e3cadc097c25a00ef05
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/58900
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22279}
This also changes AppRTC to use addTrack instead of addStream and
"early media" using the RtpTransceiver API.
Bug: webrtc:8870
Change-Id: Ie2848a87c71a95adb785367d822c61e1f753d8c6
Reviewed-on: https://webrtc-review.googlesource.com/56440
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22255}
We want to evaluate the impact on battery consumption from using the
fullscreen HW scaling.
Bug: None
Change-Id: If5becf02c6eaf77f9d0877827db39773ae17fc05
Reviewed-on: https://webrtc-review.googlesource.com/59101
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22243}
Uses new WebRtcAudioRecordSamplesReadyCallback which was added recently in
https://webrtc-review.googlesource.com/c/src/+/49981.
This CL:
- Serves as a test of new WebRtcAudioRecordSamplesReadyCallback.
- Useful for debugging purposes since it records the most native raw audio.
Bug: None
Change-Id: I57375cbf237c171e045b0bdb05f7ae1401930fbc
Reviewed-on: https://webrtc-review.googlesource.com/53120
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22128}
It turns out that some headers were not owned by any targets.
These were:
RTCVideoCodec.h
RTCVideoCodecFactory.h
RTCVideoCodecH264.h
RTCVideoEncoderVP8.h
RTCVideoDecoderVP8.h
RTCVideoEncoderVP9.h
RTCVideoDecoderVP9.h
And some were owned by multiple targets, namely:
RTCPeerConnectionFactory+Native.h
RTCPeerConnectionFactory+Private.h
RTCVideoFrameBuffer.h
These have all been moved to their appropriate homes.
This CL also fixes a lot of cyclic interdependencies in the iOS sdk build files.
Bug: webrtc:8855
Change-Id: I1b7ddb6c2a93868d1510ccf0a64bd3dd169ec3e7
Reviewed-on: https://webrtc-review.googlesource.com/49060
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22052}
