This means there's only one thread hop to the worker thread.
At the video engine level, SetOptions and SetSource
are combined into one method (all within the same critical section)
which ensures that no frame will be encoded while SetVideoSend
is only partially finished.
BUG=webrtc:5691
Review-Url: https://codereview.webrtc.org/1838413002
Cr-Commit-Position: refs/heads/master@{#13022}
This is similar to how a "receive" method is used to apply
RtpParameters to an RtpReceiver in ORTC. Currently, SetParameters
doesn't allow changing the parameters, so the main use of the API is
to retrieve the set of configured codecs. But other uses will likely
be made possible in the future.
R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1917193008 .
Cr-Commit-Position: refs/heads/master@{#12761}
The reference to the old track needs to be kept alive until SetAudioSend/
SetSource is called, because otherwise it could be deleted while the audio/
video engine is still trying to use the track.
BUG=webrtc:5796
Review-Url: https://codereview.webrtc.org/1894283002
Cr-Commit-Position: refs/heads/master@{#12598}
The problem with gmock is worked around by commenting out any other override declarations in classes using gmock.
NOPRESUBMIT=True
BUG=webrtc:3970
Review-Url: https://codereview.webrtc.org/1921653002
Cr-Commit-Position: refs/heads/master@{#12563}
But keep #including scoped_ptr.h in .h files, so as not to break
WebRTC users who expect those .h files to give them rtc::scoped_ptr.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1930463002
Cr-Commit-Position: refs/heads/master@{#12530}
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}
This change allows the application to limit the bitrate of the outgoing
audio and video streams at runtime. The API roughly follows the WebRTC
API draft, defining the RTCRtpParameters structure witn exactly one
encoding (simulcast streams are not exposed in the API for now).
(https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters)
BUG=
Review URL: https://codereview.webrtc.org/1788583004
Cr-Commit-Position: refs/heads/master@{#12025}
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}
At the top level, setting a track on an RtpSender is equivalent to
setting a source (previously called a renderer)
on a voice send stream. An RtpSender without a track
is not supposed to send data (not even muted data), so a send stream without
a source shouldn't send data.
Also replacing SendFlags with a boolean and implementing "Start"
and "Stop" methods on AudioSendStream, which was planned anyway
and simplifies this CL.
R=pthatcher@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1741933002 .
Cr-Commit-Position: refs/heads/master@{#11918}
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}
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}
