7 Commits

Author SHA1 Message Date
Sami Kalliomaki
9b0dc622d4 Revert of Combine webrtc/api/java/android and webrtc/api/java/src. (patchset #1 id:1 of https://codereview.webrtc.org/2111823002/ )
Reason for revert:
Breaks downstream dependencies

Original issue's description:
> Combine webrtc/api/java/android and webrtc/api/java/src.
>
> It used to be that there was a Java api for devices not running Android
> but that is no longer the case. I combined the directories and made
> the folder structure chromium style.
>
> BUG=webrtc:6067
> R=magjed@webrtc.org, tommi@webrtc.org
>
> Committed: ceefe20dd6

TBR=magjed@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6067

Review URL: https://codereview.webrtc.org/2106333005 .

Cr-Commit-Position: refs/heads/master@{#13357}
2016-07-01 07:37:49 +00:00
Sami Kalliomaki
ceefe20dd6 Combine webrtc/api/java/android and webrtc/api/java/src.
It used to be that there was a Java api for devices not running Android
but that is no longer the case. I combined the directories and made
the folder structure chromium style.

BUG=webrtc:6067
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2111823002 .

Cr-Commit-Position: refs/heads/master@{#13356}
2016-07-01 07:09:09 +00:00
Alex Glaznev
d57048433c Decrease the amount of maximum outstanding frames for Android HW H.264 decoder.
BUG=b/28150902
R=pbos@webrtc.org, sakal@webrtc.org

Review URL: https://codereview.webrtc.org/2088353002 .

Cr-Commit-Position: refs/heads/master@{#13302}
2016-06-27 18:51:24 +00:00
sakal
1fc4810006 Always on statistics for AndroidMediaEncoder.
Earlier, no statistics were reported if no frames were being delivered
for encoding. This makes statics always be reported regardless of if
there are frames being delivered to the encoder.

Review-Url: https://codereview.webrtc.org/2051403002
Cr-Commit-Position: refs/heads/master@{#13122}
2016-06-14 08:53:44 +00:00
Niels Möller
d28db7fd65 Delete all use of tick_util.h.
Depends on Chrome cl https://codereview.chromium.org/1888003002/, which was landed some time ago.

BUG=webrtc:5740
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1888593004 .

Cr-Commit-Position: refs/heads/master@{#12674}
2016-05-10 14:31:58 +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