webrtc_m130

History

andresp@webrtc.org f7eb75be1a Split VideoCodingModuleImpl into VideoSender and VideoReceiver.

Only implmentation is changed the interface to the module is unchanged for now.

R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2200008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4746 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-09-14 00:25:28 +00:00

build

Recognize armv7 target_arch for ios support in webrtc common.gyp

2013-09-05 16:46:36 +00:00

common_audio

Reduce cost of PushSincResampler::Resample().

2013-09-06 21:15:55 +00:00

common_video

Removing some TODO's from libyuv

2013-09-03 20:06:18 +00:00

modules

Split VideoCodingModuleImpl into VideoSender and VideoReceiver.

2013-09-14 00:25:28 +00:00

system_wrappers

OpenSL (not default): Enables low latency audio on Android.

2013-09-10 18:24:07 +00:00

test

Make unittest log printouts opt-in with a --logs flag.

2013-09-05 21:16:29 +00:00

tools

Fix format string in video_quality_analysis.cc.

2013-09-10 15:23:50 +00:00

video_engine

Implement 'abs-send-time' extension in VideoSendStream.

2013-09-11 19:00:39 +00:00

voice_engine

This issue is related to

2013-09-12 17:03:00 +00:00

.gitignore

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

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

common_types.h

Remove redundant STR_CASE_CMP macro definitions.

2013-09-09 17:50:10 +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

Make PCM16 available in Chromium builds.

2013-09-12 01:30:30 +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.