fischman@webrtc.org b900721472 Fix OpenGL rendering of WebRTCDemo by accounting for stride != width. ...

BUG=998

Review URL: https://webrtc-codereview.appspot.com/936021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3079 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-11-12 16:39:25 +00:00

..

audio_coding

Optimized function AllpassFilter2FixDec16() in isac fix for Android Neon platforms.

2012-11-09 00:39:45 +00:00

audio_conference_mixer

Move src/ -> webrtc/

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

audio_device

Add Android OWNER files

2012-11-09 17:51:55 +00:00

audio_processing

Fixed and enabled ARM assembly code in AECM and NS.

2012-11-07 22:34:31 +00:00

bitrate_controller

Move src/ -> webrtc/

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

interface

Move src/ -> webrtc/

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

media_file

Move src/ -> webrtc/

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

pacing

Adding pacing module, will replace the transmission_bucket in the RTP module.

2012-11-09 20:56:23 +00:00

remote_bitrate_estimator

Break out unittest helpers for remote_bitrate_estimator.

2012-11-07 18:35:30 +00:00

rtp_rtcp

pre-factor cleanup pre-work.

2012-11-07 17:01:04 +00:00

udp_transport

Move src/ -> webrtc/

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

utility

Fix valgrind issue.

2012-10-30 18:22:02 +00:00

video_capture

Capture module: Fixing size computation for u and v planes

2012-11-09 18:45:12 +00:00

video_coding

Revert 3071 - i420:verify image length

2012-11-12 09:16:08 +00:00

video_processing/main

Verify output frame timestamp in VideoProcessingModuleTest.Resampler.

2012-11-08 01:07:21 +00:00

video_render

Fix OpenGL rendering of WebRTCDemo by accounting for stride != width.

2012-11-12 16:39:25 +00:00

modules.gyp

Adding pacing module, will replace the transmission_bucket in the RTP module.

2012-11-09 20:56:23 +00:00