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webrtc_m130/webrtc/modules
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andrew@webrtc.org 84d84471f5 Minor fixes regarding accumulator usage on MIPS platforms. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33729004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7979 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-29 17:08:44 +00:00
..
audio_coding
Minor fixes regarding accumulator usage on MIPS platforms.
2014-12-29 17:08:44 +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
GN: Fix build for Mac
2014-12-19 13:28:37 +00:00
audio_processing
Wire up Beamformer in AudioProcessing
2014-12-19 19:57:34 +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
GN: Fix build for Mac
2014-12-19 13:28:37 +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
Rtp-Rtcp sender cleanup.
2014-12-19 13:49:55 +00:00
utility
Rename rtpDumpPktHdr_t to RtpDumpPacketHeader.
2014-12-18 09:18:42 +00:00
video_capture
Fix mac video capture leak.
2014-12-19 20:51:02 +00:00
video_coding
Add field trial for screenshare bitrates when using temporal layers.
2014-12-23 15:19:35 +00:00
video_processing
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
video_render
GN: Fix build for Mac
2014-12-19 13:28:37 +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