andresp@webrtc.org 523f93729b Re-write the build of the nacklist. ...

Review URL: https://webrtc-codereview.appspot.com/1304008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3822 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-04-11 11:30:39 +00:00

..

audio_coding

Replace legacy G_CONST with const.

2013-04-10 18:06:57 +00:00

audio_conference_mixer

WebRtc_Word32 -> int32_t in audio_conference_mixer/

2013-04-09 17:40:15 +00:00

audio_device

Removing remaining WebRtc_Word32 not in typedefs.h

2013-04-10 17:59:17 +00:00

audio_processing

WebRtc_Word32 -> int32_t in audio_processing/

2013-04-10 07:50:54 +00:00

bitrate_controller

Fix webrtc compilation errors for Chrome Win64

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

interface

Removing remaining WebRtc_Word32 not in typedefs.h

2013-04-10 17:59:17 +00:00

media_file

WebRtc_Word32 -> int32_t in media_file/

2013-04-09 13:31:37 +00:00

pacing

Add interface to signal a network down event.

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

remote_bitrate_estimator

Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps."

2013-04-09 14:56:29 +00:00

rtp_rtcp

Re-write the build of the nacklist.

2013-04-11 11:30:39 +00:00

utility

WebRtc_Word32 -> int32_t in utility/

2013-04-09 13:32:55 +00:00

video_capture

WebRTCDemo: no-op out instead of NPEing on destroyed camera.

2013-04-10 17:11:51 +00:00

video_coding

Reapply the reverted r3747.

2013-04-09 21:21:32 +00:00

video_processing/main

WebRtc_Word32 -> int32_t in video_processing/

2013-04-09 13:38:10 +00:00

video_render

WebRTCDemo: handle stride!=width from first frame.

2013-04-10 23:21:10 +00:00

modules.gyp

Remove the old unused udp_transport

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