webrtc_m130

History

sprang@webrtc.org fe5d36b6fe Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.

We will do some refactoring of video engine and would like to use the
same rtcp stats struct there. Both video and audio seem to use 8bit
fraction lost, so that is changed in the struct as well.

BUG=
R=henrik.lundin@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5039 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-10-28 09:21:07 +00:00

build

Roll chromium_revision 228675:229708

2013-10-22 12:09:48 +00:00

common_audio

Upgrade scoped_ptr to Chromium's latest version.

2013-10-22 12:50:00 +00:00

common_video

Open file in binary in CreateFromYuvFile().

2013-10-16 08:46:06 +00:00

modules

Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.

2013-10-28 09:21:07 +00:00

system_wrappers

Prefix MOVE_ONLY_TYPE_FOR_CPP_03 with WEBRTC_.

2013-10-24 15:16:53 +00:00

test

Upgrade scoped_ptr to Chromium's latest version.

2013-10-22 12:50:00 +00:00

tools

Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.

2013-10-28 09:21:07 +00:00

video_engine

Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals.

2013-10-23 23:59:45 +00:00

voice_engine

Fixing broken tests in voe_auto_test extended

2013-10-23 10:39:56 +00:00

.gitignore

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

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

common_types.h

Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.

2013-10-28 09:21:07 +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

Adds a new voice engine warning for the typing noise off state.

2013-09-17 18:09:20 +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

Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined.

2013-10-23 19:11:32 +00:00

webrtc.gyp

Reorganize GYP targets to make webrtc.gyp more usable.

2013-10-10 08:48:16 +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.