This CL adds a way to extract the underlying android.opengl.EGLContext
and javax.microedition.khronos.egl.EGLContext for EglBase14 and
EglBase10 respectively. The reason is that clients can't be expected to
use only WebRTC's OpenGL code and might need to integrate with their
own GL code.
Bug: None
Change-Id: Ie00a564de45a090683542a52005da7e43c586ced
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127888
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27205}
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}
These changes simplify the code, and also fix the issue where the peerconnectionstate would sometimes return to "new" during connection setup.
Bug: webrtc:9308
Change-Id: I895cd2f94a2b9688c821cca64d1a077317b99d44
Reviewed-on: https://webrtc-review.googlesource.com/c/111964
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25942}
This reverts commit 1e87b4f32b73526f9caaae2a7bccfbd0cd84dcb9.
Reason for revert: Breaks internal project
Original change's description:
> Replace the IceConnectionState implementation.
>
> PeerConnection::ice_connection_state() used to return a value based on both DTLS and ICE transports.
> Now that we have PeerConnection::peer_connection_state() to fill that role we can change the implementation of ice_connection_state over to match the spec.
>
> Bug: webrtc:6145
> Change-Id: Ia4f348f728f24faf4b976c63dea2187bb1f01ef0
> Reviewed-on: https://webrtc-review.googlesource.com/c/108780
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25773}
TBR=kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,jonasolsson@webrtc.org
Change-Id: Icc4368d120a4167286fa6ba2e884a3650b453eff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6145
Reviewed-on: https://webrtc-review.googlesource.com/c/111925
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25775}
PeerConnection::ice_connection_state() used to return a value based on both DTLS and ICE transports.
Now that we have PeerConnection::peer_connection_state() to fill that role we can change the implementation of ice_connection_state over to match the spec.
Bug: webrtc:6145
Change-Id: Ia4f348f728f24faf4b976c63dea2187bb1f01ef0
Reviewed-on: https://webrtc-review.googlesource.com/c/108780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25773}
Some imports of classes in the same package are a bit silly.
Removing = false for booleans is safe because Java guarantees that
an uninitialized bool will always be false.
Tbr: sakal@chromium.org
Bug: None
Change-Id: I04baa78a6e21b1c4fc74c5e46665e66481da2495
Reviewed-on: https://webrtc-review.googlesource.com/c/111243
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25678}
This support is needed if there is a big delay between the creation of
frames and the time they are delivered to the WebRTC C++ layer in
AndroidVideoTrackSource. This is the case if e.g. some heavy video
processing is applied to the frames that takes a couple of hundred
milliseconds. Currently, timestamps coming from Android video sources
are aligned to rtc::TimeMicros() once they reach the WebRTC C++ layer in
AndroidVideoTrackSource. At this point, we "forget" any latency that
might occur before this point, and audio/video sync consequently
suffers.
Bug: webrtc:9991
Change-Id: I7b1aaca9a60a978b9195dd5e5eed4779a0055607
Reviewed-on: https://webrtc-review.googlesource.com/c/110783
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25654}
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}
Support Injectable Audio Codecs from the Java SDK.
The PeerConnectionFactory.Builder defaults to
BuiltinAudio(Encoder|Decoder)Factory, but other implementations are
permitted via the Audio(Encoder|Decoder)FactoryFactory interface.
Bug: webrtc:9916
Change-Id: I61ad4a6e57666bc1be79daf5f40b129e0eacad84
Reviewed-on: https://webrtc-review.googlesource.com/c/107711
Commit-Queue: Lennart Kolmodin <kolmodin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25478}
This is implemented by allowing users to set two different aspect
ratios, one for landscape input and one for portrait input. This extra
control might be useful in other scenarios as well.
Bug: webrtc:9903
Change-Id: I91676737f4aa1f5d94cfe79ac51d5f866779945b
Reviewed-on: https://webrtc-review.googlesource.com/c/108086
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25387}
With the expanding use cases for webrtc::CryptoOptions it makes more sense for
it to be be available per peer connection instead of only as a factory option.
To support backwards compatability for now this code will support the factory
method of setting crypto options by default. However it will completely
overwrite these settings if an RTCConfiguration.crypto_options is provided.
Got LGTM offline from Sami, adding him to TBR if he has any further comments.
TBR=sakal@webrtc.org
Bug: webrtc:9891
Change-Id: I86914cab69284ad82afd7285fd84ec5f4f2c4986
Reviewed-on: https://webrtc-review.googlesource.com/c/107029
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25375}
The JS API supports two operations which have never been implemented in
the Android counterpart:
- generate a new certificate
- use this certificate when creating a new PeerConnection
Both functions are illustrated in the generateCertificate example code:
- https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/generateCertificate
Currently, on Android, a new certificate is automatically generated for
every PeerConnection with no programmatic way to set a specific
certificate.
A twin of this feature is already underway for iOS here:
- https://webrtc-review.googlesource.com/c/src/+/87303
Work sponsored by |pipe|
Bug: webrtc:9546
Change-Id: Iac221517df3ae380aef83c18c9e59b028d709a4f
Reviewed-on: https://webrtc-review.googlesource.com/c/89980
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25090}
Configuring different number of temporal layers per simulcast layer is not supported.
Bug: webrtc:9785
Change-Id: I5709b2235233420e22e68fb0ae512305ae87e36c
Reviewed-on: https://webrtc-review.googlesource.com/c/102120
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24942}
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}
Also enables support for all hardware implementations. Renames
HardwareVideoDecoderFactory to MediaCodecVideoDecoderFactory. Renames
HardwareVideoDecoder to AndroidVideoDecoder.
Bug: webrtc:8538
Change-Id: I9b351f387526af4da61fb07c07fb4285bd833e19
Reviewed-on: https://webrtc-review.googlesource.com/97680
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24586}
Limited range seems to be more used than full range and many Android
components already use limited range. This includes FileVideoCapturer,
VideoFileRenderer and HW codecs.
Bug: webrtc:9638
Change-Id: Iadd9b2f19020c6a25bde5e43a28e26a6230dde42
Reviewed-on: https://webrtc-review.googlesource.com/94543
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24576}
This is in line with the new C++ VideoCodecFactory interface.
Bug: webrtc:7925
Change-Id: Ice51cab61b6498fef1b0483ce1bd4835ef550231
Reviewed-on: https://webrtc-review.googlesource.com/88368
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23966}
Since the legacy video codecs seem to be around for some time more, we
need to make them injectable and provide a migration path for clients
that still use them so that we can clean up PeerConnectionFactory.
This CL moves the creation of EglContexts into the legacy codec
factories. Clients can then migrate to setEGLContext() instead of using
setVideoHwAccelerationOptions().
Bug: webrtc:9502
Change-Id: I608607b32db73ce3df7704a061e66d9d53946af5
Reviewed-on: https://webrtc-review.googlesource.com/87941
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23934}
All usage of these functions should be migrated to use
android.opengl.Matrix instead.
Bug: webrtc:9487
Change-Id: I023761b31cae7e7af9b537928b849657baa5bb8b
Reviewed-on: https://webrtc-review.googlesource.com/87263
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23853}
This is a reland of 21219a0e43446701810236fb9fdd59be072c12df
The default implementation of OnLogMessage(msg, sev, tag) discarded
the tag, resulting in FileRotatingLogSink not receiving tags.
Since the revert the default implementation of
OnLogMessage(msg, sev, tag) has been updated to add the tag to the log
message. A more efficient implementation of it has also been added for
FileRotatingLogSink.
Unit tests are added for the default implementation and for Loggable
injection.
Original change's description:
> Reland "Injectable logging"
>
> Any injected loggable or NativeLogger would be deleted if PCFactory
> was reinitialized without calling setInjectableLogger. Now native
> logging is not implemented as a Loggable, so it will remain active
> unless a Loggable is injected.
>
> This is a reland of 59216ec4a4151b1ba5478c8f2b5c9f01f4683d7f
>
> Original change's description:
> > Injectable logging
> >
> > 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}
>
> Bug: webrtc:9225
> Change-Id: I2fe3fbc8c323814284bb62e43fe1870bdab581ee
> TBR: kwiberg
> Reviewed-on: https://webrtc-review.googlesource.com/77140
> Commit-Queue: Paulina Hensman <phensman@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23310}
Bug: webrtc:9225
Change-Id: I67a5728fe772f0bedc9509713ed8b8ffdc31af81
TBR: kwiberg
Reviewed-on: https://webrtc-review.googlesource.com/80860
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23711}
Also adds api in PCF for specifying which library to load.
This is in preparation for a CL adding a native function
to be used only in tests.
Bug: webrtc:9225
Change-Id: I72eff272350404729424176758bfa81f7da81836
Reviewed-on: https://webrtc-review.googlesource.com/84125
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23663}
There is no need to hold on to the render thread after the OES texture
buffer has been created. toI420() is handled by the thread in the
SurfaceTextureHelper.
Bug: webrtc:9391
Change-Id: Ide081bc083db72bb991f1deba74d3cecf3e1fee6
Reviewed-on: https://webrtc-review.googlesource.com/84121
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23652}
This CL adds tests that are primarily targeting
VideoFrame.Buffer.toI420() and cropAndScale(), but includes the whole
chain for YuvConverter, GlRectDrawer, and VideoFrameDrawer.
It also includes a couple of fixes to bugs that were exposed by the new
tests.
Bug: webrtc:9186, webrtc:9391
Change-Id: I5eb62979a8fd8def28c3cb2e82dcede57c42216f
Reviewed-on: https://webrtc-review.googlesource.com/83163
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23611}
In particular:
* Trying to remove a VideoSink that was not attached should be a no-op.
* Trying to remove a null VideoSink should be a no-op.
* Adding the same VideoSink multiple times should a no-op, and should
not create duplicate native VideoSinks.
* Trying to add a null VideoSink should throw an exception in the Java
layer instead of crashing in native code.
This CL also adds tests that verify these behaviors.
Bug: webrtc:9403
Change-Id: I928b7bb7f683634e287d7fec9e26f4179f73c150
Reviewed-on: https://webrtc-review.googlesource.com/83322
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23602}
The proper closing procedure is:
1. Alice resets outgoing stream.
2. Bob receives incoming stream reset, resets his outgoing stream.
3. Alice receives incoming stream reset; channel closed!
4. Bob receives acknowledgement of reset; channel closed!
https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.7
However, up until now we've been sending both an incoming and outgoing reset
from the side initiating the closing procedure, and doing nothing on the remote
side.
This means that if you call "Close" and the remote endpoint is using an old
version of WebRTC, the channel's state will be stuck at "closing" since the
remote endpoint won't send a reset. Which is already what happens when Firefox
is talking to Chrome.
This CL also fixes an issue where the DataChannel's state prematurely went to
"closed" before the closing procedure was complete. Which could result in a
new DataChannel attempting to re-use the ID and failing.
TBR=magjed@webrtc.org
Bug: chromium:449934, webrtc:4453
Change-Id: Ic1ba813e46538c6c65868961aae6a9780d68a5e2
Reviewed-on: https://webrtc-review.googlesource.com/79061
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23478}
This CL adds the field but does not implement any functionality using it.
Bug: webrtc:9341
Change-Id: I533fc7f8bc1e40207aa16b834e0d7daa60709614
Reviewed-on: https://webrtc-review.googlesource.com/78741
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23466}
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}
Let the test expect calls to onRenegotiationNeeded(), as introduced by
https://codereview.webrtc.org/2977493002.
Bug: webrtc:7761
Change-Id: If8e3c484236f6599cc225a0398bbbc9cf6c356a5
Reviewed-on: https://webrtc-review.googlesource.com/48364
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23165}
This is a reland of 3e0dee26603cdc3a2653c225398f55dd8ca0d8c1
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 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}
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}
Currently this warnings prevernt chromium roll into webrtc, because we
consider them as errors. So to unblock roll all warning are suppressed.
All places are documented into bug and will be fixed later.
TBR=henrika@webrtc.org
Bug: webrtc:9175
Change-Id: I0bf5a4b65eb49308e28f71a92d42b5fad6a99b74
Reviewed-on: https://webrtc-review.googlesource.com/71420
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22956}
