pwestin@webrtc.org db4185664c Introduced pause and resume to the pacer ...

Review URL: https://webrtc-codereview.appspot.com/1217007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3717 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-03-22 23:39:29 +00:00

..

audio_coding

WebRtc_Word -> stdint in audio_coding/g711/

2013-03-21 13:38:29 +00:00

audio_conference_mixer

Replace AudioFrame's operator= with CopyFrom().

2013-01-22 04:44:30 +00:00

audio_device

Expose the capture-side AudioProcessing object and allow it to be injected.

2013-03-05 01:12:49 +00:00

audio_processing

Fixed initialization of SPL in echo_control_mobile.

2013-03-19 00:25:01 +00:00

bitrate_controller

Fix webrtc compilation errors for Chrome Win64

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

interface

Removed redundant VP8 width/height and made sure the generic width/height is set.

2013-03-12 09:59:27 +00:00

media_file

Fix webrtc compilation errors for Chrome Win64

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

pacing

Introduced pause and resume to the pacer

2013-03-22 23:39:29 +00:00

remote_bitrate_estimator

Rename webrtc::StatsObserver to webrtc::CallStatsObserver

2013-02-19 22:09:36 +00:00

rtp_rtcp

Generic video-codec support.

2013-03-18 16:39:03 +00:00

udp_transport

Revert r3667 and r3665

2013-03-13 23:20:57 +00:00

utility

Changing non-const reference arguments to pointers, ACM

2013-02-21 10:27:48 +00:00

video_capture

Fix -Wstring-conversion warnings.

2013-03-20 21:22:48 +00:00

video_coding

Remove incorrect asserts.

2013-03-21 12:15:44 +00:00

video_processing/main

Remove '<(library)' in gyp files.

2013-01-18 23:42:21 +00:00

video_render

Thread safety issue fix in incoming_video_stream.cc. See issue 1465.

2013-03-20 16:57:09 +00:00

modules.gyp

Revert r3667 and r3665

2013-03-13 23:20:57 +00:00