5 Commits

Author SHA1 Message Date
Alex Glaznev
6a4a03c59c Add an option to soft reset HW decoder.
Soft reset can be used when input frame resolution changes
to avoid re creating MediaCodec instance.
Instead MediaCodec is flushed and some variables are reset.

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

Review URL: https://codereview.webrtc.org/1732533002 .

Cr-Commit-Position: refs/heads/master@{#11878}
2016-03-04 22:11:02 +00:00
Per
fd22e6cf2d Change PeerConnectionFactory.setVideoHwAccelerationOptions to be able to replace Egl context.
BUG=b/27222102
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1707933003 .

Cr-Commit-Position: refs/heads/master@{#11666}
2016-02-18 10:36:01 +00:00
perkj
461121c67b Replaced eglbase_jni with with holding a EglBase in PeerConnectionFactory.
Review URL: https://codereview.webrtc.org/1695763002

Cr-Commit-Position: refs/heads/master@{#11627}
2016-02-15 14:28:40 +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