mikhal@webrtc.org d6bd7cd2b1 removing redundant calls to cleanframes ...

Review URL: https://webrtc-codereview.appspot.com/1318004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3844 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-04-12 17:09:51 +00:00

..

audio_coding

Replace legacy G_CONST with const.

2013-04-10 18:06:57 +00:00

audio_conference_mixer

WebRtc_Word32 -> int32_t in audio_conference_mixer/

2013-04-09 17:40:15 +00:00

audio_device

Removing remaining WebRtc_Word32 not in typedefs.h

2013-04-10 17:59:17 +00:00

audio_processing

WebRtc_Word32 -> int32_t in audio_processing/

2013-04-10 07:50:54 +00:00

bitrate_controller

Fix webrtc compilation errors for Chrome Win64

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

interface

Fix issues with incorrect wrap checks when having big buffers and high bitrate.

2013-04-11 17:48:02 +00:00

media_file

WebRtc_Word32 -> int32_t in media_file/

2013-04-09 13:31:37 +00:00

pacing

Add interface to signal a network down event.

2013-03-27 16:36:01 +00:00

remote_bitrate_estimator

Change capture interface to use NTP capture time.

2013-04-12 11:56:23 +00:00

rtp_rtcp

Adding a payload type for RTX.

2013-04-12 14:55:46 +00:00

utility

WebRtc_Word32 -> int32_t in utility/

2013-04-09 13:32:55 +00:00

video_capture

Change capture interface to use NTP capture time.

2013-04-12 11:56:23 +00:00

video_coding

removing redundant calls to cleanframes

2013-04-12 17:09:51 +00:00

video_processing/main

WebRtc_Word32 -> int32_t in video_processing/

2013-04-09 13:38:10 +00:00

video_render

WebRTCDemo: handle stride!=width from first frame.

2013-04-10 23:21:10 +00:00

module_common_types_unittest.cc

Fix issues with incorrect wrap checks when having big buffers and high bitrate.

2013-04-11 17:48:02 +00:00

modules.gyp

Fix issues with incorrect wrap checks when having big buffers and high bitrate.

2013-04-11 17:48:02 +00:00