marpan@webrtc.org c35f1d26c5 FEC: Fix to coverity issue 14448: unintended sign extension. ...

Review URL: https://webrtc-codereview.appspot.com/647004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2400 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-06-12 20:12:13 +00:00

..

audio_coding

Use new fileutil functions for trace in ACM

2012-06-12 07:16:24 +00:00

audio_conference_mixer

Move audio_frame_operations to the utility module.

2012-05-29 22:13:14 +00:00

audio_device

Added gyp variable to include/exclude all tests.

2012-05-24 13:23:35 +00:00

audio_processing

Refactored Neon code for AECM module, by using pure assembly code.

2012-06-07 16:17:17 +00:00

bitrate_controller

Attempt to fix broken encoding.

2012-06-04 11:04:05 +00:00

interface

Re-added ChangeUniqueId temporary for chrome.

2012-05-24 09:52:19 +00:00

media_file

Added gyp variable to include/exclude all tests.

2012-05-24 13:23:35 +00:00

remote_bitrate_estimator

abs() was used instead of fabsf(), which returns int and not float and therefore truncated the return value.

2012-06-07 13:48:04 +00:00

rtp_rtcp

FEC: Fix to coverity issue 14448: unintended sign extension.

2012-06-12 20:12:13 +00:00

udp_transport

Fix compilation errors on ChromeOS

2012-05-30 16:46:09 +00:00

utility

Fixing gyp bug in https://webrtc-codereview.appspot.com/599006

2012-05-30 14:32:42 +00:00

video_capture

Fix for the alignment problems/mismatch in ViECapture and VP8Encoder.

2012-06-05 23:52:59 +00:00

video_coding

Fix for the alignment problems/mismatch in ViECapture and VP8Encoder.

2012-06-05 23:52:59 +00:00

video_processing/main

Check return value of fwrite. [Video Module]

2012-05-29 17:33:13 +00:00

video_render

Refactored IncomingVideoStream and VideoRenderFrame, to get code in better shape when hunting BUG=481.

2012-05-30 10:45:18 +00:00

modules.gyp

Refactoring the receive-side bandwidth estimation into its own module.

2012-06-07 08:10:14 +00:00