The caller can set a negative or zero file size to avoid using a limit.
BUG=
Review-Url: https://codereview.webrtc.org/1974453002
Cr-Commit-Position: refs/heads/master@{#12730}
Reason for revert:
I was too quick to judge, this CL does not cause the problem.
Original issue's description:
> Revert of Android GlDrawer: Add frame size as argument to draw functions (patchset #2 id:20001 of https://codereview.webrtc.org/1948473002/ )
>
> Reason for revert:
> Causes errors on Google3 import.
>
> Original issue's description:
> > Android GlDrawer: Add frame size as argument to draw functions
> >
> > BUG=b/28544933
> >
> > Committed: https://crrev.com/71af75dc3ca8516017dca9de2ebe582145ecad14
> > Cr-Commit-Position: refs/heads/master@{#12623}
>
> TBR=glaznev@webrtc.org,magjed@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=b/28544933
>
> Committed: https://crrev.com/172683173dd84a72659ad494962245445eb2a353
> Cr-Commit-Position: refs/heads/master@{#12627}
TBR=glaznev@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=b/28544933
Review-Url: https://codereview.webrtc.org/1947073002
Cr-Commit-Position: refs/heads/master@{#12628}
Java objects in the API should be allowed to be null in some cases.
Specifically, a null value for maxBitrateBps in RtpParameters.java
has a specific meaning and doesn't imply an error has occurred.
NOTRY=True
Review URL: https://codereview.webrtc.org/1853523002
Cr-Commit-Position: refs/heads/master@{#12221}
The track state should be implicitly set by the underlying source.
This removes the public method and cleans up how AudioRtpReceiver is created. Further more it cleans up how the RtpReceivers are destroyed.
Note that this cl depend on https://codereview.webrtc.org/1790633002.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1816143002
Cr-Commit-Position: refs/heads/master@{#12115}
Also removes unused track states kLive and kFailed.
Since this also required a Video source to exist in all unit tests that create a track, a FakeVideoTrackSource is added and used in tests.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1790633002
Cr-Commit-Position: refs/heads/master@{#12098}
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}
Soft reset can be used when input frame resolution changes
to avoid re creating MediaCodec instance.
Instead MediaCodec is flushed and some variables are reset.
R=pbos@webrtc.org, perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1732533002 .
Cr-Commit-Position: refs/heads/master@{#11878}
This CL replaces the function SurfaceTextureHelper.setListener() that
could only be called once with the functions startListening() and
stopListening() that can be called multiple times. This is necessary
when the SurfaceTextureHelper will be passed to the VideoCapturerAndroid
in startCapture(). startListening() will be called in startCapture() and
stopListening() in stopCapture().
BUG=webrtc:5519
Review URL: https://codereview.webrtc.org/1755573002
Cr-Commit-Position: refs/heads/master@{#11855}
This change is done to remove abnormally high decode time measurements for H264 decoding. H264 decoding sometimes keeps a few frames as reference before outputting a new decoded frame. This pipeline causes some frames to get stuck when the source stops sending new frames. When the source starts sending frames again, the decode time measurements for the frames that were stuck will include the pause time, which can be arbitrary high. This CL is a simple fix for this problem by constraining the decode time values to a "reasonable" range.
BUG=b/27306053
Review URL: https://codereview.webrtc.org/1725243007
Cr-Commit-Position: refs/heads/master@{#11792}
This CL simplifies the VideoCapturer interface from 'String getSupportedFormatsAsJson() throws JSONException' to 'List<CaptureFormat> getSupportedFormats()'. The intermediate conversion to/from a JSON string is removed, and AndroidVideoCapturerJni converts the Java list to a C++ vector directly instead.
BUG=webrtc:5519
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1702603002 .
Cr-Commit-Position: refs/heads/master@{#11669}
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}
Also fixing an issue with the Java PeerConnection unit test.
It wasn't correctly waiting for 10 video frames to be received.
And fixed an issue with the video engine, where generated
black frames don't get any rotation.
BUG=webrtc:5128
Review URL: https://codereview.webrtc.org/1639583003
Cr-Commit-Position: refs/heads/master@{#11583}
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}
