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}
Multiple sources with the same names forces ugly GYP hacks in
Chromium's libjingle.gyp. Rename the sources in WebRTC to
enable cleaning this up in Chromium.
To summarize:
webrtc/media/base/constants.{cc,h} -> mediaconstants.{cc,h}
webrtc/p2p/base/constants.{cc,h} -> p2pconstants.{cc,h}
This CL will require coordinating landing a roll in Chromium.
BUG=webrtc:4256
NOTRY=True
Review URL: https://codereview.webrtc.org/1750593002
Cr-Commit-Position: refs/heads/master@{#11842}
Meaning "a=msid:...", instead of "a=ssrc:X msid:...".
An additional option to SdpSerialize determines if the
"a=msid" attribute is used.
Review URL: https://codereview.webrtc.org/1688383002
Cr-Commit-Position: refs/heads/master@{#11644}
For now, the network cost is purely based on the network type (cellular has cost 0xFFFF and everything else has cost 0).
Add cost to the candidate signaling and the stun request signaling (which is needed for peer reflexive candidates).
BUG=webrtc:4325
Review URL: https://codereview.webrtc.org/1668073002
Cr-Commit-Position: refs/heads/master@{#11642}
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}
