Commit Graph

6 Commits

Author SHA1 Message Date
16032126ed This implementation greatly simplifies Android video capturing stack. The old
stack will be removed soon in a separate CL. Constraints will not be supported
in the new implementation. Apps can request a format directly and the closest
supported format will be selected.

Changes needed from the apps:
1. Use the new createVideoSource without constraints.
2. Call startCapture manually.
3. Don't call videoSource.stop/restart, use startCapture/stopCapture instead.

R=magjed@webrtc.org
TBR=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/2127893002 .

Cr-Commit-Position: refs/heads/master@{#13504}
2016-07-20 14:13:20 +00:00
6ab787964a Adding deadbeef@ as owner of api and p2p, and honghaiz as owner of p2p.
NOTRY=true

Review-Url: https://codereview.webrtc.org/2154543002
Cr-Commit-Position: refs/heads/master@{#13494}
2016-07-16 07:48:11 +00:00
14897d0b7d Add missing dependencies on audio, video and call to the new GN files.
This caused linker failures when depending on the new `api` target.

TBR=henrika@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2029323006
Cr-Commit-Position: refs/heads/master@{#13042}
2016-06-03 20:14:37 +00:00
208d19845d Rename APK tests workaround to make it more generic.
We plan to add junit tests running with Robolectric
so naming these files "apk" is slightly confusing.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2020213002
Cr-Commit-Position: refs/heads/master@{#12971}
2016-05-31 11:01:47 +00:00
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
aa32c3e537 Update API for Objective-C RTCIceServer
BUG=

Review URL: https://codereview.webrtc.org/1499653003

Cr-Commit-Position: refs/heads/master@{#11000}
2015-12-14 03:58:19 +00:00