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9334ac2d78f760b37f512ef6c12bff220d1654c1
platform-external-webrtc/webrtc/modules
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pbos@webrtc.org 9334ac2d78 Use vector of CSRCs for DeliverFrame & SetCSRCs. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28029004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7734 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-24 08:25:50 +00:00
..
audio_coding
Build fix for MIPS Android Webview build.
2014-11-21 16:28:32 +00:00
audio_conference_mixer
GN: Add common configs to all targets.
2014-09-28 17:37:22 +00:00
audio_device
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
audio_processing
Build fix for MIPS Android Webview build.
2014-11-21 16:28:32 +00:00
bitrate_controller
Put send-side bwe probing under finch experiment.
2014-11-10 13:55:16 +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 size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
media_file
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
pacing
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
remote_bitrate_estimator
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
rtp_rtcp
Use vector of CSRCs for DeliverFrame & SetCSRCs.
2014-11-24 08:25:50 +00:00
utility
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
video_capture
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
video_coding
Update mock_frame_dropper.h to use size_t
2014-11-21 09:40:57 +00:00
video_processing
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
video_render
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +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
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
modules.gyp
Use RtpFileSource in NetEqDecodingTest
2014-11-17 09:08:38 +00:00
OWNERS
Add kjellander@webrtc.org as OWNER for *.isolate
2014-06-10 05:42:53 +00:00