webrtc_m130

History

phoglund@webrtc.org 8400246fce Improved error messages when binaries are missing. Also stderr = stdout now.

Now that these scripts are called from browser tests, we need to print everything on stdout since the tests will throw away stderr when invoking programs. I chose to assign sys.stderr to sys.stdout. Otherwise I would have missed stuff like parser.error, which print to stderr.

The error message will get improved because the old code did not catch the case when the binary was missing, which lead to a very confusing error when that was the case. This gets fixed now.

BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1886004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4416 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-07-29 11:01:03 +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

To fix a bug in InverseFFTAndWindow() function in AECM.

2013-07-26 23:43:33 +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

Improved error messages when binaries are missing. Also stderr = stdout now.

2013-07-29 11:01:03 +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.