pbos@webrtc.org 315669939a Fix typo from RtpPacketizerH264. ...

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27609004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7295 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-09-25 07:31:42 +00:00

..

audio_coding

WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t

2014-09-24 10:31:02 +00:00

audio_conference_mixer

Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.

2014-09-11 06:20:28 +00:00

audio_device

Skip dlclose() on AddressSanitizer.

2014-09-11 17:29:11 +00:00

audio_processing

Enable render downmixing to mono in AudioProcessing.

2014-09-24 20:06:23 +00:00

bitrate_controller

Remove former team members from OWNERS and WATCHLISTS

2014-08-26 06:12:08 +00:00

desktop_capture

Add missing DesktopConfigurationMonitor Unlock in webrtc::ScreenCapturerMac

2014-09-24 17:23:46 +00:00

interface

Add H.264 packetization.

2014-07-31 14:59:24 +00:00

media_file

Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.

2014-09-11 06:20:28 +00:00

pacing

Move thread_annotations.h to webrtc/base/.

2014-09-24 06:05:00 +00:00

remote_bitrate_estimator

Change gflags and gmock includes to be full paths.

2014-09-04 09:55:40 +00:00

rtp_rtcp

Fix typo from RtpPacketizerH264.

2014-09-25 07:31:42 +00:00

utility

Mark all virtual overrides in the hierarchies of RtpDump and

2014-09-11 22:45:54 +00:00

video_capture

gn: Hide modules/video_capture:video_capture_internal_impl behind an arg

2014-09-23 12:37:06 +00:00

video_coding

Reduce jitter delay for low fps streams.

2014-09-24 14:06:56 +00:00

video_processing

Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.

2014-09-11 06:20:28 +00:00

video_render

Clean directx_sdk_path as it is already defined in base/common.gypi

2014-09-19 08:14:12 +00:00

module_common_types_unittest.cc

Fix issues with incorrect wrap checks when having big buffers and high bitrate.

2013-04-11 17:48:02 +00:00

modules_java_chromium.gyp

Fixes build breakage when building WebRTC in Chromium and having include_tests=1.

2013-07-08 18:14:58 +00:00

modules_java.gyp

Build all java files into jar for each module on Android

2013-07-01 17:52:39 +00:00

modules_tests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

modules_unittests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

modules.gyp

Reduce jitter delay for low fps streams.

2014-09-24 14:06:56 +00:00

OWNERS

Add kjellander@webrtc.org as OWNER for *.isolate

2014-06-10 05:42:53 +00:00