This CL changes the interface by adding a SurfaceTextureHelper argument
to VideoCapturer.startCapture(). This removes the need for the
VideoCapturer to create the SurfaceTextureHelper itself. This also means
that it is no longer necessary to send an EGLContext to the
VideoCapturerAndroid.create() function.
The SurfaceTextureHelper is now created in AndroidVideoCapturerJni, and
the EGLContext is passed from PeerConnectionFactory in
nativeCreateVideoSource().
Another change in this CL is that the C++ SurfaceTextureHelper creates
the Java SurfaceTextureHelper instead of getting it passed as an
argument in the ctor.
BUG=webrtc:5519
Review URL: https://codereview.webrtc.org/1783793002
Cr-Commit-Position: refs/heads/master@{#11977}
and signaling the remote side to remove its remote candidate by setting the candidate priority to 0.
BUG=
Review URL: https://codereview.webrtc.org/1648813004
Cr-Commit-Position: refs/heads/master@{#11958}
Moved VideoSourceInterface to MediaStreamInterface.h
Renamed VideoSourceTest to VideoCapturerTrackSourceTest
Renamed VideoSource to VideoCaptureTrackSource and cl lint and cl format.
BUG=webrtc:5426
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1770003002 .
Cr-Commit-Position: refs/heads/master@{#11893}
This CL factors out the interface that AndroidVideoCapturerJni is using to communicate with the Java counterpart. This interface is moved into VideoCapturer. The interface is not touched in this CL, and a follow-up CL is planned to simplify and improve it.
Another change is that the native part of VideoCapturer is created in PeerConnectionFactory.createVideoSource() instead of doing it immediately in the ctor.
BUG=webrtc:5519
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1696553003 .
Cr-Commit-Position: refs/heads/master@{#11606}
Reason for revert:
Breaks downstream compilation. Please reland in a non-breaking fashion.
Original issue's description:
> Android: Remove VideoCapturer
>
> This CL makes PeerConnectionFactory.createVideoSource() and nativeCreateVideoSource work directly with VideoCapturerAndroid instead of going via VideoCapturer. The native part is now created in nativeCreateVideoSource() instead of doing it immediately in VideoCapturerAndroid.create().
>
> BUG=webrtc:5519
> R=perkj@webrtc.org
>
> Committed: https://crrev.com/09eab315fddc3432c19d8f662f4b9360f2a58010
> Cr-Commit-Position: refs/heads/master@{#11582}
TBR=perkj@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5519
Review URL: https://codereview.webrtc.org/1690073002
Cr-Commit-Position: refs/heads/master@{#11586}
This CL makes PeerConnectionFactory.createVideoSource() and nativeCreateVideoSource work directly with VideoCapturerAndroid instead of going via VideoCapturer. The native part is now created in nativeCreateVideoSource() instead of doing it immediately in VideoCapturerAndroid.create().
BUG=webrtc:5519
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1684403002 .
Cr-Commit-Position: refs/heads/master@{#11582}
In addition to the code moved from talk/app/webrtc
there were some files in webrtc/api/objctests that still
had the libjingle license header.
BUG=webrtc:5418
TBR=tkchin@webrtc.org
NOTRY=True
Review URL: https://codereview.webrtc.org/1680293005
Cr-Commit-Position: refs/heads/master@{#11552}
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc
The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.
I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle:
https://codereview.chromium.org/1615433002
BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1610243002 .
Cr-Commit-Position: refs/heads/master@{#11545}
