stefan@webrtc.org 4b6f747373 Fixes a newly introduced bug in the jitter buffer where buffer reallocation ...

causes corrupt pointers.
Review URL: http://webrtc-codereview.appspot.com/186003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@688 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-10-04 06:58:39 +00:00

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audio_coding

To fix warnings in test files.

2011-09-23 15:45:48 +00:00

audio_conference_mixer

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audio_device

Add kPlatformAndroid to platform check function

2011-09-19 17:39:05 +00:00

audio_processing

Adds possibility to log delay estimates in AEC.

2011-10-03 08:18:10 +00:00

interface

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media_file

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rtp_rtcp

Added support for new RTCP message REMB (remote estimated max bitrate)

2011-09-20 13:52:04 +00:00

udp_transport

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utility

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video_capture

Return the number of /dev/video* without trying to open it.

2011-09-21 16:57:15 +00:00

video_coding

Fixes a newly introduced bug in the jitter buffer where buffer reallocation

2011-10-04 06:58:39 +00:00

video_processing/main

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video_render

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modules.gyp

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