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11d8176cb3383a2f96e118ff054e92e97a8d9db4
platform-external-webrtc/webrtc/modules
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stefan@webrtc.org 11d8176cb3 Move updating nack bitrate inside UpdateNACKBitRate. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7960 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 09:52:24 +00:00
..
audio_coding
Remove obsolete target_arch == armv7.
2014-12-18 21:36:18 +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
Remove or rename typedefs with _t prefixes.
2014-12-17 13:43:55 +00:00
audio_processing
Merge beamformer
2014-12-18 22:22:04 +00:00
bitrate_controller
Suppress REMB in bitrate ctrl if it seems lika a short network glitch.
2014-12-18 11:53:59 +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
Move updating nack bitrate inside UpdateNACKBitRate.
2014-12-19 09:52:24 +00:00
utility
Rename rtpDumpPktHdr_t to RtpDumpPacketHeader.
2014-12-18 09:18:42 +00:00
video_capture
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
video_coding
Refactor some receive-side stats.
2014-12-18 13:50:16 +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
Merge beamformer
2014-12-18 22:22:04 +00:00
OWNERS
Add kjellander@webrtc.org as OWNER for *.isolate
2014-06-10 05:42:53 +00:00