104 Commits

Author SHA1 Message Date
hta
a2a49d9d9c This CL provides interfaces that do not use constraints for
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}
2016-03-04 10:51:44 +00:00
guoweis
36f0137fd5 Implement Turn/Turn first logic for connection selection.
This feature is off by default and can be turned on by setting IceConfig. When turned on, we'll choose a Turn/Turn (UDP takes higher priroity) over the other types of connections while no any connection is writable. However, when there is best connection or there is pending triggered check, those will take higher priority.

BUG=webrtc:4591

Review URL: https://codereview.webrtc.org/1577233006

Cr-Commit-Position: refs/heads/master@{#11850}
2016-03-03 02:02:58 +00:00
kjellander
b24317bfda Fix license headers in webrtc/api.
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}
2016-02-10 15:54:53 +00:00
Henrik Kjellander
15583c19d7 Move talk/app/webrtc to webrtc/api
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}
2016-02-10 09:53:26 +00:00