webrtc_m130

History

braveyao@webrtc.org b6433b7a1e Access receiving_ under receive_cs critical section

Note: InsertRTPPacket/InsertRTCPPacket could be merged into 
ReceivedRTPPacket, as there are no other callers.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1869005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4410 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-07-26 09:02:46 +00:00

build

Don't set clang_use_chrome_plugins in common.gypi

2013-07-26 00:55:46 +00:00

common_audio

Downstream latest Chromium SincResampler changes.

2013-07-25 22:04:30 +00:00

common_video

Proper spacing for end-of-namespace comments.

2013-07-03 15:12:26 +00:00

modules

Fixes resources and data path in modules_unittests.isolate.

2013-07-25 22:44:04 +00:00

system_wrappers

Fixes: Resolves conflict that will happen when merging libjingle's and WebRTC's supplemental.gyp. By separating build_with_chromium and build_with_libjingle one can now just define build_with_libjingle in libjingle's supplemental.gyp. Once that is done it will be possible to merge the two supplemental.gyp-files. I.e. in WebRTC the supplemental.gyp would only set build_with_chromium to 0 since there is no longer any reason to disable logging and tests as they will be accessible in the same repository as libjingle.

2013-07-16 16:37:22 +00:00

test

Add root_path_android.cc to webrtc/test/Android.mk.

2013-07-09 15:24:16 +00:00

tools

Unreverts revert: Makes it possible to find files used by some unit tests when running them as Chrome native tests.

2013-07-08 14:55:23 +00:00

video_engine

Access receiving_ under receive_cs critical section

2013-07-26 09:02:46 +00:00

voice_engine

Default constructor for RtcpAppHandler.

2013-07-17 14:25:45 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

common_types.h

webrtc/common_types.h: Document bitrate fields' units.

2013-07-23 18:32:10 +00:00

common_unittest.cc

Fix unnamed-type-template-args warnings on clang.

2013-06-27 10:18:09 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

2013-07-25 18:28:29 +00:00

engine_configurations.h

Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing.

2013-06-19 17:03:47 +00:00

LICENSE

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

LICENSE_THIRD_PARTY

Consolidate all third party licenses in LICENSE_THIRD_PARTY.

2013-05-05 18:54:10 +00:00

PATENTS

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

PRESUBMIT.py

Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.

2013-07-22 22:32:50 +00:00

README.chromium

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

typedefs.h

Disable clang C++11 warnings to permit OVERRIDE keyword.

2013-05-08 08:34:34 +00:00

webrtc.gyp

Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle

2013-07-23 18:15:11 +00:00

README.chromium

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.