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webrtc_m130/webrtc/modules
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jmarusic@webrtc.org 50604128db Method WebRtc_g722_encode that is eventually called always returns non-negative integer (internal counter) 
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34259004

Cr-Commit-Position: refs/heads/master@{#8428}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8428 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 12:16:49 +00:00
..
audio_coding
Method WebRtc_g722_encode that is eventually called always returns non-negative integer (internal counter)
2015-02-19 12:16:49 +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