mflodman@webrtc.org 47d657b68e Remove Set/Get sending status from the default RTP module. ...

This is now taken care of by the payload router and the calls to set_active.

BUG=769
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42379004

Cr-Commit-Position: refs/heads/master@{#8427}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8427 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-02-19 10:30:19 +00:00

..

audio_coding

Adding two new stats to VoiceReceiverInfo

2015-02-18 15:24:39 +00:00

audio_conference_mixer

Remove ChangeUniqueID.

2015-01-29 12:14:13 +00:00

audio_device

Remove EventWrapper::Reset().

2015-02-13 14:58:38 +00:00

audio_processing

Apply good settings to Beamformer

2015-02-17 19:24:37 +00:00

bitrate_controller

Enable bitrate probing by default.

2015-02-03 12:34:17 +00:00

desktop_capture

GN: Prepare to remove webrtc_base target

2015-01-21 20:22:33 +00:00

interface

Revamp the ProcessThreadImpl implementation.

2015-02-06 09:44:45 +00:00

media_file

Remove ChangeUniqueID.

2015-01-29 12:14:13 +00:00

pacing

Enable bitrate probing by default in PacedSender.

2015-02-16 15:48:29 +00:00

remote_bitrate_estimator

Implement the Nada rmcat proposal within the simulation framework.

2015-02-17 16:04:17 +00:00

rtp_rtcp

Remove Set/Get sending status from the default RTP module.

2015-02-19 10:30:19 +00:00

utility

Don't rely on webrtc/base/scoped_ptr.h to include stuff for you

2015-02-11 14:31:19 +00:00

video_capture

WebRtcVideoCapturer: Remove dead code |OnIncomingCapturedEncodedFrame|

2015-02-17 15:10:41 +00:00

video_coding

Initialize RTPVideoHeader fields to correctly set simulcastIdx for non VP8 codecs.

2015-02-18 20:30:18 +00:00

video_processing

Remove ChangeUniqueID.

2015-01-29 12:14:13 +00:00

video_render

Remove EnableMirroring and MirrorRenderStream

2015-02-18 12:55:17 +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

Re-implementing AcmOpusTest as AcmGenericCodecOpusTest

2015-02-18 13:15:45 +00:00

OWNERS

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

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