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webrtc_m130/webrtc/modules
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pbos@webrtc.org 451a133f44 Add AGC manager tests. 
R=bjornv@webrtc.org
BUG=4098

Review URL: https://webrtc-codereview.appspot.com/35539005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7914 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 14:48:47 +00:00
..
audio_coding
Make an AudioEncoder subclass for RED
2014-12-16 13:41:36 +00:00
audio_conference_mixer
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
audio_device
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
audio_processing
Add AGC manager tests.
2014-12-16 14:48:47 +00:00
bitrate_controller
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
desktop_capture
Fix a platform check to use WEBRTC_WIN instead of OS_WIN.
2014-11-12 20:53:00 +00:00
interface
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
media_file
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
pacing
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
remote_bitrate_estimator
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
rtp_rtcp
Cleanup: Remove 'const' qualifier from OnReceivedEstimatedBitrate().
2014-12-16 12:29:59 +00:00
utility
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
video_capture
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
video_coding
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
video_processing
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
video_render
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +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
Make an AudioEncoder subclass for RED
2014-12-16 13:41:36 +00:00
OWNERS
Add kjellander@webrtc.org as OWNER for *.isolate
2014-06-10 05:42:53 +00:00