phoglund@webrtc.org 315d39866e Formatted dtmf_queue. ...

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1398004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3982 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-05-08 10:04:06 +00:00

..

audio_coding

Fix clang errors in non-GYP_DEFINES=clang=1 build

2013-05-07 12:36:21 +00:00

audio_conference_mixer

Remove vim/emacs modelines from .gypi files

2013-04-16 12:40:34 +00:00

audio_device

Refactoring for typing detection

2013-05-07 21:04:24 +00:00

audio_processing

Consolidate common_audio into a single target.

2013-04-30 23:43:26 +00:00

bitrate_controller

Remove vim/emacs modelines from .gypi files

2013-04-16 12:40:34 +00:00

desktop_capture

Add DesktopCapturer interface for desktop capturers.

2013-04-29 20:10:57 +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

Remove vim/emacs modelines from .gypi files

2013-04-16 12:40:34 +00:00

pacing

Fix crash in pacer.

2013-05-06 16:37:22 +00:00

remote_bitrate_estimator

Fix clang errors in non-GYP_DEFINES=clang=1 build

2013-05-07 12:36:21 +00:00

rtp_rtcp

Formatted dtmf_queue.

2013-05-08 10:04:06 +00:00

utility

Consolidate common_audio into a single target.

2013-04-30 23:43:26 +00:00

video_capture

Make sure timestamps are monotonically increasing.

2013-04-24 23:54:31 +00:00

video_coding

Fix VCMProcessTimer::TimeUntilProcess() unsigned-integer underflow problem.

2013-05-08 06:38:53 +00:00

video_processing/main

Consolidate common_audio into a single target.

2013-04-30 23:43:26 +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

Add DesktopCapturer interface for desktop capturers.

2013-04-29 20:10:57 +00:00