aluebs@webrtc.org 0c39e91cc8 Merge beamformer ...

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7958 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-12-18 22:22:04 +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

Store the received report blocks map (mapped per remote ssrc) in a map per source ssrc.

2014-12-18 14:30:32 +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