Many of these interfaces are not intuitive, or are the way they are for
complex historical reasons, so it would be nice to document these things
for future developers.
Also, many nonstandard things (such as RTCConfiguration options) were
not documented at all before this CL.
BUG=webrtc:7131
TBR=pthatcher@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2680273002
Cr-Commit-Position: refs/heads/master@{#16485}
Moves webrtc/common_video/rotation.h and parts of
webrtc/common_video/include/video_frame_buffer.h and
webrtc/video_frame.h, and adds to a new GN target api:video_frame_api.
BUG=webrtc:5880
Review-Url: https://codereview.webrtc.org/2517173004
Cr-Commit-Position: refs/heads/master@{#15993}
Files only making use of size_t from basictypes.h are replaced with
stddef.h, except in cases where they already for instance use stdio.h or
stdlib.h that already provide size_t.
BUG=webrtc:6853
R=tommi@webrtc.org
Review-Url: https://codereview.webrtc.org/2605123002
Cr-Commit-Position: refs/heads/master@{#15865}
Permits overriding the source-default is_screencast option to be able to
treat screencast sources as fluid video, preserving motion at the loss
of individual frame quality (or vice versa).
BUG=chromium:653531
R=deadbeef@webrtc.org
Review-Url: https://codereview.webrtc.org/2579993003
Cr-Commit-Position: refs/heads/master@{#15659}
The stat is currently always set to zero until the residual echo detector has landed.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2431443003
Cr-Commit-Position: refs/heads/master@{#14721}
This cl clean up unused forward declaration of cricket::AudioFrame and cricket::Renderer.
cricket::VideoFrame is used in the interface and thus the header file is included.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2145613006
Cr-Commit-Position: refs/heads/master@{#13477}
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 enabled us to be able to remove VideoTrack::GetSink and RemoteVideoCapturer.
Since video frames from the decoder is delivered on a media engine internal thread, VideoBroadCaster must be made thread safe.
BUG=webrtc:5426
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1765423005 .
Cr-Commit-Position: refs/heads/master@{#11944}
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 patch tries to only change the interface to VideoTrack, with
minimal changes to the implementation. Some points worth noting:
VideoTrackRenderers should ultimately be deleted, but it is kept for
now since we need an object implementing webrtc::VideoRenderer, and
that shouldn't be VideoTrack.
BUG=webrtc:5426
TBR=glaznev@webrtc.org // please look at examples
Review URL: https://codereview.webrtc.org/1684423002
Cr-Commit-Position: refs/heads/master@{#11775}
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}
