stefan@webrtc.org 9e254133ad Rewrite the jitter buffer statistics test and put make it robust under valgrind. ...

BUG=1158

Review URL: https://webrtc-codereview.appspot.com/1116008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3579 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-02-28 08:45:23 +00:00

..

audio_coding

Changing non-const reference arguments to pointers, ACM

2013-02-21 10:27:48 +00:00

audio_conference_mixer

Replace AudioFrame's operator= with CopyFrom().

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

audio_device

Misc cleanups to webrtc/android code:

2013-02-20 23:13:46 +00:00

audio_processing

AEC Refactoring:

2013-02-27 21:03:41 +00:00

bitrate_controller

Fix webrtc compilation errors for Chrome Win64

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

interface

Rename webrtc::StatsObserver to webrtc::CallStatsObserver

2013-02-19 22:09:36 +00:00

media_file

Fix webrtc compilation errors for Chrome Win64

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

pacing

Remove '<(library)' in gyp files.

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

remote_bitrate_estimator

Rename webrtc::StatsObserver to webrtc::CallStatsObserver

2013-02-19 22:09:36 +00:00

rtp_rtcp

Fix to send a full NACK list at least roughly once every 1.5 x RTT.

2013-02-27 16:23:06 +00:00

udp_transport

Bug fix for webrtc issue 1391. Typo in sin_length for socket address.

2013-02-15 18:42:12 +00:00

utility

Changing non-const reference arguments to pointers, ACM

2013-02-21 10:27:48 +00:00

video_capture

Minor bug fix in maxFPS parameter declaration.

2013-02-21 22:03:39 +00:00

video_coding

Rewrite the jitter buffer statistics test and put make it robust under valgrind.

2013-02-28 08:45:23 +00:00

video_processing/main

Remove '<(library)' in gyp files.

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

video_render

Misc cleanups to webrtc/android code:

2013-02-20 23:13:46 +00:00

modules.gyp

Initial upload of NetEq4

2013-01-29 12:09:21 +00:00