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}
This eliminates some instances rtc:Optional and makes the code
simpler. No changes in defaults or other behaviour are intended.
BUG=webrtc:4906
Review URL: https://codereview.webrtc.org/1818033002
Cr-Commit-Position: refs/heads/master@{#12326}
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}
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}
This CL also adds control flag in webrtcsession_unittests
that says whether to prefer constraints APIs or non-constraints APIs, and uses it in the test that was needed
to uncover the bug.
BUG=webrtc:4906
Review URL: https://codereview.webrtc.org/1775033002
Cr-Commit-Position: refs/heads/master@{#11947}
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}
all interfaces that formerly took constraints parameters
in name=value form.
This is in preparation for making Chrome only use these
explicit interfaces.
BUG=webrtc:4906
Review URL: https://codereview.webrtc.org/1717583002
Cr-Commit-Position: refs/heads/master@{#11870}
RFC 5245 allows an ICE restart to occur on only one media section.
However, before this CL, if an endpoint attempted to do this, we would
change our local ICE ufrag/pwd in every media section.
Also did some refactoring, turning the transport options from
mediasesion.h into a map.
Review URL: https://codereview.webrtc.org/1671173002
Cr-Commit-Position: refs/heads/master@{#11728}
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}
