webrtc_m130

History

pwestin@webrtc.org d35964a1ce Fixing AV sync.

Increased 2 const to allow for a bigger difference in AV sync.

BUG=1711

Re-wrote the ComputeDelays to be readable and remove the possibilities of returning values lower than base_target_delay_ms

R=mflodman@webrtc.org, mikhal@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1367004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3922 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-04-30 16:06:10 +00:00

build

Removing another instance of file api

2013-04-25 23:31:27 +00:00

common_audio

Add a wrapper around PushSincResampler and the old Resampler.

2013-04-29 17:27:29 +00:00

common_video

WebRtc_Word32 -> int32_t in common_video.

2013-04-09 13:36:08 +00:00

modules

Fixing AV sync.

2013-04-30 16:06:10 +00:00

system_wrappers

Make win_support_condition_variables_primitive global to aligned with |library|

2013-04-15 18:06:42 +00:00

test

Revert "Updating test file contents to emmastjernloef"

2013-04-17 14:59:28 +00:00

tools

Fix frame_editing_unittest reference file handling.

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

video_engine

Fixing AV sync.

2013-04-30 16:06:10 +00:00

voice_engine

Add a wrapper around PushSincResampler and the old Resampler.

2013-04-29 17:27:29 +00:00

.gitignore

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

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

common_types.h

Adding extra options to interact with external encoder/decoder.

2013-04-23 22:50:53 +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

Replace legacy G_CONST with const.

2013-04-10 18:06:57 +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.