sprang@webrtc.org 43c883954f Allow rtp packet history to dynamically expand in size.
When using the paced sender, packets will be put into the rtp packet
history and then retreived from there again when it is time to send.

In some cases (low send bitrate and very large frames created) this
may overflow, causing packets to be overwritten in the packet history
before they have been sent.

Check this condition and expand history size if needed.

This is primarily triggered during screenshare, when
switching to a large picture with lots of high frequency
details in it.

BUG=4171
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34879004

Cr-Commit-Position: refs/heads/master@{#8195}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8195 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 09:09:41 +00:00
2015-01-08 21:22:01 +00:00
2014-12-15 07:03:04 +00:00
2015-01-27 19:15:48 +00:00
2014-06-17 08:54:03 +00:00
2014-11-04 17:06:31 +00:00
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
BSD-3-Clause 446 MiB
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%