57 Commits

Author SHA1 Message Date
perkj
a3ede6c510 Renamed VideoSourceInterface to VideoTrackSourceInterface.
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}
2016-03-08 00:28:03 +00:00
kjellander
43942d1f1e Roll chromium_revision 508edd3..35d57a0 (379249:379535)
Change log: 508edd3..35d57a0
Full diff: 508edd3..35d57a0

Changed dependencies:
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/708db16..58218b6
DEPS diff: 508edd3..35d57a0/DEPS

No update to Clang.

TBR=torbjorng@webrtc.org
BUG=webrtc:5634
NOTRY=True

Review URL: https://codereview.webrtc.org/1773543002

Cr-Commit-Position: refs/heads/master@{#11890}
2016-03-07 21:59:15 +00:00
kjellander
f475277547 Rename constants files in webrtc/{media,p2p}
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}
2016-03-02 13:42:35 +00:00
kjellander@webrtc.org
9b8df25c73 Move talk/session/media -> webrtc/pc
The libjingle_p2p target is renamed to rtc_pc.
The libjingle_p2p_unittest test will be renamed in a
separate follow-up CL, to make it possible to run all
trybots successfully for this CL.

BUG=webrtc:5419
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1691463002 .

Cr-Commit-Position: refs/heads/master@{#11592}
2016-02-12 05:48:10 +00:00
kjellander@webrtc.org
5ad129741c Rename webrtc/media/webrtc -> webrtc/media/engine
BUG=webrtc:5420
NOTRY=True
R=pthatcher@google.com, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1684163002 .

Cr-Commit-Position: refs/heads/master@{#11591}
2016-02-12 05:39:50 +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