platform-external-webrtc

Files

andrew@webrtc.org 1b31c78e5f Remove VoE's default call in Trace::SetLevelFilter.

This is an application level setting. Applying it here has the potential to override the application's preferences.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1252004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3727 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-03-26 14:09:48 +00:00

build

Use ninja with merge_libs.py on Windows and clean up.

2013-02-26 02:47:50 +00:00

common_audio

Refactored inline assembly code in complex_fft.c, by combining the individual __asm lines into a single block, to avoid potential register usage problems when building with different tools.

2013-03-01 23:01:14 +00:00

common_video

Fixing/disabling Windows x64 warnings

2013-02-04 10:07:17 +00:00

modules

Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust.

2013-03-26 14:02:30 +00:00

system_wrappers

Add trace printouts to all unit tests.

2013-03-22 17:13:23 +00:00

test

Add some VoE and AudioProcessing mocks.

2013-03-25 21:20:38 +00:00

tools

Fix frame_editing_unittest reference file handling.

2013-03-11 15:30:33 +00:00

video_engine

Updated WebRTC version to 3.27

2013-03-22 21:59:19 +00:00

voice_engine

Remove VoE's default call in Trace::SetLevelFilter.

2013-03-26 14:09:48 +00:00

.gitignore

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

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

common_types.h

Add min and target bitrate to VideoCodec.

2013-03-22 17:13:08 +00:00

engine_configurations.h

Revert r3667 and r3665

2013-03-13 23:20:57 +00:00

LICENSE

Move src/ -> webrtc/

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

LICENSE_THIRD_PARTY

Move src/ -> webrtc/

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

PATENTS

Move src/ -> webrtc/

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

README.chromium

Move src/ -> webrtc/

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

typedefs.h

Add some VoE and AudioProcessing mocks.

2013-03-25 21:20:38 +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.