pwestin@webrtc.org 6faf71d27b Remove the old unused udp_transport ...

Review URL: https://webrtc-codereview.appspot.com/1272009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3788 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-04-08 23:25:25 +00:00

..

audio_coding

Fixed a bug in isac-fix's entropy coding function: out of bounds acces to array.

2013-04-08 16:37:53 +00:00

audio_conference_mixer

Replace AudioFrame's operator= with CopyFrom().

2013-01-22 04:44:30 +00:00

audio_device

Resolves TSan v2 reports data races in voe_auto_test.

2013-04-05 14:34:57 +00:00

audio_processing

Resolves TSan v2 reports data races in voe_auto_test.

2013-04-05 14:34:57 +00:00

bitrate_controller

Fix webrtc compilation errors for Chrome Win64

2013-01-29 06:45:22 +00:00

interface

Removed redundant VP8 width/height and made sure the generic width/height is set.

2013-03-12 09:59:27 +00:00

media_file

Fix webrtc compilation errors for Chrome Win64

2013-01-29 06:45:22 +00:00

pacing

Add interface to signal a network down event.

2013-03-27 16:36:01 +00:00

remote_bitrate_estimator

WebRtc_Word32 => int32_t remote_bitrate_estimator/

2013-04-08 10:04:37 +00:00

rtp_rtcp

WebRtc_Word32 => int32_t for rtp_rtcp/

2013-04-08 11:08:41 +00:00

utility

Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizer

2013-04-02 11:59:11 +00:00

video_capture

WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14.

2013-03-27 16:48:34 +00:00

video_coding

Reduce execution time of rate control test.

2013-04-08 20:32:48 +00:00

video_processing/main

Remove '<(library)' in gyp files.

2013-01-18 23:42:21 +00:00

video_render

Thread safety issue fix in incoming_video_stream.cc. See issue 1465.

2013-03-20 16:57:09 +00:00

modules.gyp

Remove the old unused udp_transport

2013-04-08 23:25:25 +00:00