stefan@webrtc.org 94a8c03141 Slightly increased bandwidth adaptation at both receive- and send-side. ...

The send-side increase factor is increased to better follow the pace
of the receive-side estimate, while the receive-side factor is
increased to speed up adaptation.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/297002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1030 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-11-25 14:09:37 +00:00

..

audio_coding

Cleaning up neteq_unittest

2011-11-25 13:43:42 +00:00

audio_conference_mixer

Added empty unit tests to get code coverage measured.

2011-11-24 07:20:00 +00:00

audio_device

Add support to 96kHz sampling rate to Windows CoreAudio interface.

2011-11-25 02:45:39 +00:00

audio_processing

Fix sample rate printout in process_test.

2011-11-23 20:37:12 +00:00

interface

Updating to VP8 RTP spec rev -02

2011-11-24 12:52:40 +00:00

media_file

Added empty unit tests to get code coverage measured.

2011-11-24 07:20:00 +00:00

rtp_rtcp

Slightly increased bandwidth adaptation at both receive- and send-side.

2011-11-25 14:09:37 +00:00

udp_transport

Added empty unit tests to get code coverage measured.

2011-11-24 07:20:00 +00:00

utility

Added empty unit tests to get code coverage measured.

2011-11-24 07:20:00 +00:00

video_capture

Included modules in webrtc.gyp and fixed build errors.

2011-11-16 15:36:44 +00:00

video_coding

Remove WebRTC dependency on libvpx_lib and libvpx_include.

2011-11-25 12:12:43 +00:00

video_processing/main

Fixing Release compilation errors

2011-11-23 12:20:35 +00:00

video_render

Included modules in webrtc.gyp and fixed build errors.

2011-11-16 15:36:44 +00:00

modules.gyp

Restructuring and adding dummy unit test target.

2011-11-17 13:56:54 +00:00