asapersson@webrtc.org 9ffd8fe96b Indentation changes. ...

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8107 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-01-21 08:22:50 +00:00

..

audio_coding

Setting Opus target application.

2015-01-20 16:01:50 +00:00

audio_conference_mixer

Move system_wrappers.gyp files to the proper directory.

2015-01-14 09:30:52 +00:00

audio_device

Move system_wrappers.gyp files to the proper directory.

2015-01-14 09:30:52 +00:00

audio_processing

Revert 8080 "Support 48kHz in AEC"

2015-01-20 10:22:49 +00:00

bitrate_controller

Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps.

2015-01-19 15:44:47 +00:00

desktop_capture

Remove COMPILE_ASSERT and use static_assert everywhere

2015-01-14 10:51:54 +00:00

interface

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

media_file

Remove COMPILE_ASSERT and use static_assert everywhere

2015-01-14 10:51:54 +00:00

pacing

Move system_wrappers.gyp files to the proper directory.

2015-01-14 09:30:52 +00:00

remote_bitrate_estimator

Remove unnecessary remote bitrate estimator build rule which serves no purpose.

2015-01-16 07:50:17 +00:00

rtp_rtcp

Indentation changes.

2015-01-21 08:22:50 +00:00

utility

Move system_wrappers.gyp files to the proper directory.

2015-01-14 09:30:52 +00:00

video_capture

Move internal capture+render to build_with_chromium==0 condition

2015-01-20 11:40:45 +00:00

video_coding

Implement SimulcastEncoderAdapter support.

2015-01-14 16:26:23 +00:00

video_processing

Move system_wrappers.gyp files to the proper directory.

2015-01-14 09:30:52 +00:00

video_render

Move internal capture+render to build_with_chromium==0 condition

2015-01-20 11:40:45 +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_java_chromium.gyp

Fixes build breakage when building WebRTC in Chromium and having include_tests=1.

2013-07-08 18:14:58 +00:00

modules_java.gyp

Build all java files into jar for each module on Android

2013-07-01 17:52:39 +00:00

modules_tests.isolate

Update isolate files for Android APK tests.

2014-11-13 08:35:05 +00:00

modules_unittests.isolate

Removes unused test files by audio_processing/transient

2014-12-15 16:13:05 +00:00

modules.gyp

Remove unnecessary remote bitrate estimator build rule which serves no purpose.

2015-01-16 07:50:17 +00:00

OWNERS

Add kjellander@webrtc.org as OWNER for *.isolate

2014-06-10 05:42:53 +00:00