8 Commits

Author SHA1 Message Date
kjellander
a96e2d77cb Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.

The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.

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 in order to not
break Git history.

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.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
  webrtc/base/testutils.cc
  webrtc/base/testutils.h

The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.

I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media 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/1604303002/

BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1587193006

Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-05 07:52:35 +00:00
magjed@webrtc.org
f09e7b8a4f WebRtcVideoFrame: DCHECK exclusive ownership for non-const pixel access
Add some const safety by DCHECK(HasOneRef()) in non-const GetYPlane. This CL also replaces all incorrect non-const calls with const calls for pixel data access in cricket::VideoFrame. It's easy to call the non-const version of e.g. GetYPlane by mistake, even if only const-access is needed. For example:
const scoped_ptr<cricket::VideoFrame> foo;
const uint8_t* y = foo->GetYPlane();
will actually call the non-const version of GetYPlane.

R=mflodman@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39079004

Cr-Commit-Position: refs/heads/master@{#8507}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8507 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 14:50:19 +00:00
jlmiller@webrtc.org
5f93d0a140 Update libjingle license statements at top of talk files for consistency
BUG=2133
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 21:36:13 +00:00
tkchin@webrtc.org
8125744a5f Cleanup RTCVideoRenderer interface.
RTCVideoRenderer should be a protocol not a class. This change includes
an adapter for use with the C++ apis. The video views have been refactored
to implement that protocol.

BUG=3795
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7626 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 23:06:15 +00:00
buildbot@webrtc.org
a09a99950e (Auto)update libjingle 73222930-> 73226398
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13 17:26:08 +00:00
buildbot@webrtc.org
d4e598d57a (Auto)update libjingle 72097588-> 72159069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 17:36:52 +00:00
tkchin@webrtc.org
1732a591e7 Add a UIView for rendering a video track.
RTCEAGLVideoView provides functionality to render a supplied RTCVideoTrack using OpenGLES2.

R=fischman@webrtc.org
BUG=3188

Review URL: https://webrtc-codereview.appspot.com/12489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6192 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 23:26:01 +00:00
henrike@webrtc.org
28e2075280 Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk


git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 00:45:36 +00:00