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webrtc_m130/webrtc/modules
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marpan@webrtc.org 38d11b8529 Enable encoder multi-threading for VP9. 
R=stefan@webrtc.org
TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/41489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8150 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 15:21:36 +00:00
..
audio_coding
Introduce rtc::CheckedDivExact
2015-01-26 11:08:53 +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
Fix an unitialized variable warning.
2015-01-21 22:05:12 +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
GN: Prepare to remove webrtc_base target
2015-01-21 20:22:33 +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
Switch to use range based loops in the BWE simulation framework.
2015-01-23 08:29:52 +00:00
rtp_rtcp
Split packets/bytes in StreamDataCounter into RtpPacketCounter struct.
2015-01-22 09:39:59 +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
Enable encoder multi-threading for VP9.
2015-01-26 15:21:36 +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