mikhal@webrtc.org 9e9cc72b53 Relanding r3071 - updates for i420: Making sure that decoded frame is complete and buffer size is sufficient. Re-landing is possible following r3094 - which disabled a problematic test. ...

Review URL: https://webrtc-codereview.appspot.com/939019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3097 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-11-14 17:56:46 +00:00

..

audio_coding

Refactoring acm_generic_codec

2012-11-13 14:09:35 +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

Clarifies the bandwidth estimation interfaces.

2012-11-13 15:02:13 +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

Clarifies the bandwidth estimation interfaces.

2012-11-13 15:02:13 +00:00

rtp_rtcp

Enable paced sender.

2012-11-13 21:12:39 +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

Relanding r3071 - updates for i420: Making sure that decoded frame is complete and buffer size is sufficient. Re-landing is possible following r3094 - which disabled a problematic test.

2012-11-14 17:56:46 +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