This is a reland of 5faf36ef3c
The issue in Chrome has been fixed and this should be safe to reland.
TBR=deadbeef
Original change's description:
> Implement RtpParameters.transaction_id for PC RtpSenderInterface
>
> The transaction_id field should be refreshed for every getParameters()
> call and checked at each setParameters() call.
> This also checks that getParameters() was ever called to return a proper
> error code.
>
> Bug: webrtc:7580
> Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
> Reviewed-on: https://webrtc-review.googlesource.com/70820
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23120}
Bug: webrtc:7580
Change-Id: Iabd41fb21afdf452c039d5513824ae334f8d1d3f
Reviewed-on: https://webrtc-review.googlesource.com/76980
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23247}
Allows passing a Loggable to PCFactory.initializationOptions, which
is then injected to Logging.java and logging.h. Future log messages
in both Java and native will then be passed to this Loggable.
Bug: webrtc:9225
Change-Id: I2ff693380639448301a78a93dc11d3a0106f0967
Reviewed-on: https://webrtc-review.googlesource.com/73243
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23241}
Before this CL, there would be an out-of-bounds write in the ByteBuffer
copying when a decoded frame had height != sliceHeight.
Bug: webrtc:9194
Change-Id: Ibb80e5555e8f00d9e1fd4cb8a73f5e4ccd5a0b81
Tested: 640x360 loopback with eglContext == null in AppRTCMobile on Pixel.
Reviewed-on: https://webrtc-review.googlesource.com/74120
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23184}
This reverts commit 5faf36ef3c.
Reason for revert: fast/peerconnection/RTCRtpSender-setParameters.html
failing in webrtc roll, probably this CL? https://chromium-review.googlesource.com/c/chromium/src/+/1045889.
Original change's description:
> Implement RtpParameters.transaction_id for PC RtpSenderInterface
>
> The transaction_id field should be refreshed for every getParameters()
> call and checked at each setParameters() call.
> This also checks that getParameters() was ever called to return a proper
> error code.
>
> Bug: webrtc:7580
> Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
> Reviewed-on: https://webrtc-review.googlesource.com/70820
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23120}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,orphis@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:7580
Change-Id: I86da108227f8fc8d235bb2e9559377c800595b8c
Reviewed-on: https://webrtc-review.googlesource.com/74740
Reviewed-by: Max Morin <maxmorin@webrtc.org>
Commit-Queue: Max Morin <maxmorin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23134}
The transaction_id field should be refreshed for every getParameters()
call and checked at each setParameters() call.
This also checks that getParameters() was ever called to return a proper
error code.
Bug: webrtc:7580
Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
Reviewed-on: https://webrtc-review.googlesource.com/70820
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23120}
This is a reland of 3e0dee2660
Original change's description:
> Android: Remove deprecated PeerConnectionFactory ctors
>
> This CL removes deprecated PeerConnectionFactory ctors as well as some
> deprecated comments and functions left from the
> PeerConnectionFactory.initialize work.
>
> Bug: webrtc:9158
> Change-Id: I757f85b52cbfdbe15bf2570c394202b898892550
> Reviewed-on: https://webrtc-review.googlesource.com/70400
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23085}
TBR=sakal
Bug: webrtc:9158
Change-Id: Idb3628be85cc3268a7a4cf6990af5ed2f406ab07
Reviewed-on: https://webrtc-review.googlesource.com/74400
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23114}
This CL is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/71666 where a lot of code
was removed. Accidentally, the code that called
SurfaceTextureHelper.dispose() was removed. This code used to reside in
surfacetexturehelper.cc. This CL reintroduces the call to dispose in the
VideoSource.java backwards compatibility path.
Bug: webrtc:9181
Change-Id: I3e439dbf97965d806d238f7697561ac5ee9e79f1
Reviewed-on: https://webrtc-review.googlesource.com/73180
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23087}
This CL removes deprecated PeerConnectionFactory ctors as well as some
deprecated comments and functions left from the
PeerConnectionFactory.initialize work.
Bug: webrtc:9158
Change-Id: I757f85b52cbfdbe15bf2570c394202b898892550
Reviewed-on: https://webrtc-review.googlesource.com/70400
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23085}
The VPN adapter type is not effectively supported in WebRTC Android for
1) the network monitor may not obtain the VPN adapter type from the OS,
e.g. via NetworkInfo.getType, 2) and VPN adapter type is replaced
by the adapter type of an underlying network by the network monitor in
the current implementation. Specifically, WebRTC Android would
previously classify VPNs as either type ADAPTER_TYPE_UNKNOWN, or the
type of the currently active network (which we assume the VPN is
using).
In this CL, VPNs are classified as ADAPTER_TYPE_VPN whenever possible,
and the underlying network type, if available from the VPN, is
separately stored and used to prioritize ICE candidates in network path
selection.
This allows ADAPTER_TYPE_VPN to be used in networkIgnoreMask to ignore
VPNs when gathering ICE candidates.
Bug: webrtc:9168
Change-Id: I9513c76a114ba967437b699e71223a4a2f13f34a
Reviewed-on: https://webrtc-review.googlesource.com/70960
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23061}
There was an attempt to add MediaRecording functionality to the camera
classes, but it was never finished and never worked properly. This CL
removes the code for it. In the future, if offline video recording is
needed we should add it as a VideoSink instead of inside the camera
classes.
Bug: webrtc:9144
Change-Id: I74b70d4b128aa212d84e70da01e5e19133c5af24
Reviewed-on: https://webrtc-review.googlesource.com/69642
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23050}
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}
The new ADM code removed some redundancies, which led to a decrease in
log output. This especially affected NS and AEC logs. This change
reintroduces these log messages, making debugging easier. "Acoustic
Echo Canceler" has been changed to AEC for easier grepping.
Some new logging is also added.
Bug: webrtc:7452
Change-Id: I9bfb91895931d73d92f3187c8c7c5b7524ac05ba
Reviewed-on: https://webrtc-review.googlesource.com/71401
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23003}
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}
Any native call before PeerConnectionFactory.initialize() will fail.
This means creation of JavaAudioDeviceModule will fail if it's created
before PeerConnectionFactory.initialize(). Clients should technically
always call PeerConnectionFactory.initialize() first, but we can help
the situation by deferring creation of the native ADM until it's
actually needed.
Bug: webrtc:7452
Change-Id: I53df2bdb980a8bdc413975f1cea6bcf297b453d5
Reviewed-on: https://webrtc-review.googlesource.com/70763
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22936}
Logging the OpenGL shader source code makes it easier to debug problems.
Bug: None
Change-Id: Ie4724b1353511eae3806e98270b04e5daa4c11fc
Reviewed-on: https://webrtc-review.googlesource.com/69322
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22900}
This reverts commit 64051d4975.
Reason for revert: Fix applied.
Original change's description:
> Revert "Android: Generalize and make TextureBufferImpl public"
>
> This reverts commit 28111d7fa0.
>
> Reason for revert: Crashes video_quality_loopback_test.
>
> Original change's description:
> > Android: Generalize and make TextureBufferImpl public
> >
> > This CL generalizes TextureBufferImpl so it's useful from other contexts than
> > from a SurfaceTextureHelper, and fixes a bug in cropAndScale(). It also exposes
> > the class in the api so that clients don't have to duplicate the logic.
> >
> > Bug: None
> > Change-Id: Ib82aa8bee025ec14de74a7be9d91fd4e5298a248
> > Reviewed-on: https://webrtc-review.googlesource.com/69819
> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22875}
>
> TBR=magjed@webrtc.org,sakal@webrtc.org
>
> Change-Id: Ica7fc181fec70b8b79f39f0e114eef81a03aa116
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: None
> Reviewed-on: https://webrtc-review.googlesource.com/70240
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22878}
TBR=magjed@webrtc.org,sakal@webrtc.org
Change-Id: I173d1ccfe0baa80460f796ebaedc51731233108f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/70183
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22883}
This reverts commit 28111d7fa0.
Reason for revert: Crashes video_quality_loopback_test.
Original change's description:
> Android: Generalize and make TextureBufferImpl public
>
> This CL generalizes TextureBufferImpl so it's useful from other contexts than
> from a SurfaceTextureHelper, and fixes a bug in cropAndScale(). It also exposes
> the class in the api so that clients don't have to duplicate the logic.
>
> Bug: None
> Change-Id: Ib82aa8bee025ec14de74a7be9d91fd4e5298a248
> Reviewed-on: https://webrtc-review.googlesource.com/69819
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22875}
TBR=magjed@webrtc.org,sakal@webrtc.org
Change-Id: Ica7fc181fec70b8b79f39f0e114eef81a03aa116
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/70240
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22878}
This CL generalizes TextureBufferImpl so it's useful from other contexts than
from a SurfaceTextureHelper, and fixes a bug in cropAndScale(). It also exposes
the class in the api so that clients don't have to duplicate the logic.
Bug: None
Change-Id: Ib82aa8bee025ec14de74a7be9d91fd4e5298a248
Reviewed-on: https://webrtc-review.googlesource.com/69819
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22875}
This CL makes it possible to create a GlTextureFrameBuffer from any
thread. The actual GL resources will be allocated the first time
setSize() is called. The purpose is to be able to use 'final' variables
more often for this class and avoid @Nullable annotations.
Bug: None
Change-Id: I350304bcd33fd674990254df37a615995972f322
Reviewed-on: https://webrtc-review.googlesource.com/69241
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22835}
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 flag (added to CryptoOptions) will allow applications to opt-in to
use of this suite, before it's disabled by default later. See bug for
more details.
TBR=magjed@webrtc.org
Bug: webrtc:7670
Change-Id: I800bedd4b26d807b6b7ac66b505d419c3323e454
Reviewed-on: https://webrtc-review.googlesource.com/64390
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22586}
Add support for creating java PeerConnectionFactory from native one
by adding:
1. Constructor from native pointer in java PeerConnectionFactory
2. Method NativeToJavaPeerConnectionFactory in
sdk/android/native/api/peerconnection/peerconnectionfactory.h that
provides ability to convert native factory to java one.
Bug: webrtc:8946
Change-Id: Ibe8b019bd0d45849e2b16d74663d054784526746
Reviewed-on: https://webrtc-review.googlesource.com/62344
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22564}
This CL performs some simplifications and cleanups of the moved audio code.
* All JNI interaction now goes from the C++ audio manager calling into
the Java audio manager. The calls back from the Java code to the C++
audio manager are removed (this was related to caching audio parameters).
It's simpler this way because the Java code is now unaware of the C++
layer and it will be easier to make this into a Java interface.
* A bunch of state was removed that was related to caching the audio parameters.
* Some unused functions from audio manager was removed.
* The Java audio manager no longer depends on ContextUtils, and the context has
to be passed in externally instead. This is done because we want to get rid of
ContextUtils eventually.
* The selection of what AudioDeviceModule to create (AAudio, OpenSLES
input/output is now exposed in the interface. The reason is that client should
decide and create what they need explicitly instead of setting blacklists
in static global WebRTC classes. This will be more modular long term.
* Selection of what audio device module to create (OpenSLES combinations) no
longer requires instantiating a C++ AudioManager and is done with static
enumeration methods instead.
Bug: webrtc:7452
Change-Id: Iba29cf7447a1f6063abd9544d7315e10095167c8
Reviewed-on: https://webrtc-review.googlesource.com/63760
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@{#22542}
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}
Add configurable parameters in RTCConfiguration with the default value
given by the constants CONNECTION_WRITE_CONNECT_TIME and
CONNECTION_WRITE_CONNECT_FAILURES in the ICE implementation. These two
parameters define the time period for which a candidate pair must wait
for ping response and the minimum number of connectivity checks that
the pair must send without response before its state becomes unreliable
from writable as defined in the current ICE implementation.
Bug: webrtc:8988
Change-Id: I484599b7d776489a87741ffea8926df766095da9
Reviewed-on: https://webrtc-review.googlesource.com/60704
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22411}
The connectivity check intervals for candidate pairs with strong and
weak connectivity are currently constants in the ICE implementation. A
set of suboptimal value of these constants for a given application may
result in undesirable behavior including excessive network switching
latency. This CL adds these intervals to RTCConfiguration that is
available to applications to configure, while maintaining the original
constants as their default value for compatibility with existing
applications.
Bug: webrtc:8988
Change-Id: I804b0f4cf7881be7d3c8aec2776bc9596de72482
Reviewed-on: https://webrtc-review.googlesource.com/60585
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22351}
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}
This adds wrappers to the following native APIs:
- SdpSemantics enum added to the RTCConfiguration
- RtpTransceiver
- PeerConnection.addTrack
- PeerConnection.removeTrack
- PeerConnection.addTransceiver
- PeerConnection.getTransceivers
These APIs are used with the new Unified Plan semantics.
Bug: webrtc:8869
Change-Id: I19443f3ff7ffc91a139ad8276331f09e57cec554
Reviewed-on: https://webrtc-review.googlesource.com/57800
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@{#22317}
This change makes the class thread-safe.
Bug: b/73773043
Change-Id: I1ad13e4f15907e3dd1fef1307f9c654e53b69b22
Reviewed-on: https://webrtc-review.googlesource.com/57040
Commit-Queue: Honghai Zhang <honghaiz@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22238}
Allows passing in the application context to NetworkMonitor in
startMonitoring. The audio code will refactored once it is moved under
sdk/android.
Bug: webrtc:8937
Change-Id: I50c917a845fc4f711899a97d34c04813cc68b68c
Reviewed-on: https://webrtc-review.googlesource.com/58091
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22231}
This ensures memory is released timely and avoids problems with garbage
collection.
Native buffers don't support array operation, so FileVideoCapturer had
to be update to use FileChannel to write ByteBuffers directly.
Bug: None
Change-Id: I3f63d2adc159e9f39f0c68dd0bd6b1747686584e
Reviewed-on: https://webrtc-review.googlesource.com/55262
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22118}
STUN candidates use STUN binding requests to keep NAT bindings open. The
interval at which the STUN keepalive pings are sent is configurable now
via RTCConfiguration.
TBR=sakal@webrtc.org
Bug: None
Change-Id: I5f99ea3fe1e9042fa2bf7dcab0aace78f57739e6
Reviewed-on: https://webrtc-review.googlesource.com/54180
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22109}
This reverts commit 00733015fa.
Reason for revert: The reason for a downstream test failure on the original commit and a workaround has been found. Solution is to keep a PeerConnectionFactory constructor implementation as the same as before.
Original change's description:
> Revert "Enables PeerConnectionFactory using external fec controller"
>
> This reverts commit 4f07bdb255.
>
> Reason for revert: Speculatively reverting, because downstream test is now hitting "PeerConnectionFactory.initialize was not called before creating a PeerConnectionFactory" error, even though it did call initialize. I don't see how any change in this CL could cause that, but it's the only CL on the blamelist, and it does modify PeerConnectionFactory.java
>
> Original change's description:
> > Enables PeerConnectionFactory using external fec controller
> >
> > Bug: webrtc:8799
> > Change-Id: Ieb2cf6163b9a83844ab9ed4822b4a7f1db4c24b8
> > Reviewed-on: https://webrtc-review.googlesource.com/43961
> > Commit-Queue: Ying Wang <yinwa@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22038}
>
> TBR=sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org
>
> Change-Id: I95868c35d6f9973e0ebf563814cd71d0fcbd433d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8799
> Reviewed-on: https://webrtc-review.googlesource.com/54080
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22040}
TBR=deadbeef@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org
Bug: webrtc:8799
Change-Id: If9f3292bfcc739782967530c49f006d0abbc38a8
Reviewed-on: https://webrtc-review.googlesource.com/55400
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22100}
