webrtc

Files

pbos@webrtc.org a5f1787f63 WebRtc_Word32 -> int32_t in test/

BUG=314

Review URL: https://webrtc-codereview.appspot.com/1302004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3794 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-04-09 11:10:21 +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

WebRtc_Word32 -> int32_t in audio_device/

2013-04-09 10:30:35 +00:00

system_wrappers

WebRtc_Word32 -> int32_t in system_wrappers

2013-04-09 09:06:11 +00:00

test

WebRtc_Word32 -> int32_t in test/

2013-04-09 11:10:21 +00:00

tools

Fix frame_editing_unittest reference file handling.

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

video_engine

Always set render delay in ViEChannel::RegisterExternalDecoder.

2013-04-09 00:34:42 +00:00

voice_engine

WebRtc_Word32 -> int32_t in voice_engine/

2013-04-09 10:09:10 +00:00

.gitignore

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

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

common_types.h

Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested.

2013-03-28 09:14:36 +00:00

engine_configurations.h

Remove WEBRTC_*_ENGINE_NETWORK_API use

2013-04-04 19:51:42 +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.