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jlmiller@webrtc.org ea1c84285c Correct GetDriveType error handling. 
BUG=4020
R=brucedawson@google.com, juberti@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36899005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8127 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 17:44:19 +00:00
..
base Correct GetDriveType error handling. 2015-01-22 17:44:19 +00:00
build Fix searching for DirectX SDK during GN build. 2015-01-14 21:25:25 +00:00
common_audio Add beamforming to audioproc_float utility. 2015-01-15 01:28:36 +00:00
common_video Move system_wrappers.gyp files to the proper directory. 2015-01-14 09:30:52 +00:00
examples Use int64_t more consistently for times, in particular for RTT values. 2015-01-12 21:51:21 +00:00
libjingle Added support for an Origin header in STUN messages. 2015-01-10 00:47:02 +00:00
modules Modify some tests to never use DTX disable mode 2015-01-22 13:30:58 +00:00
overrides Remove COMPILE_ASSERT and use static_assert everywhere 2015-01-14 10:51:54 +00:00
p2p release the turn allocation by sending a refresh request with lifetime 0 2015-01-17 00:58:15 +00:00
sound rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation. 2014-10-01 16:33:03 +00:00
system_wrappers Simplify and guard access to WindowsRealTimeClock. 2015-01-21 12:51:13 +00:00
test Move system_wrappers.gyp files to the proper directory. 2015-01-14 09:30:52 +00:00
tools Move system_wrappers.gyp files to the proper directory. 2015-01-14 09:30:52 +00:00
video Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. 2015-01-22 09:39:59 +00:00
video_engine Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. 2015-01-22 09:39:59 +00:00
voice_engine Modify some tests to never use DTX disable mode 2015-01-22 13:30:58 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
BUILD.gn GN: Prepare to remove webrtc_base target 2015-01-21 20:22:33 +00:00
call.h Use int64_t more consistently for times, in particular for RTT values. 2015-01-12 21:51:21 +00:00
codereview.settings Add codereview.settings to the /webrtc subdirectory 2014-12-05 13:43:35 +00:00
common_types.h Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. 2015-01-22 09:39:59 +00:00
common.gyp Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
common.h Add a Config class interface to AudioProcessing for passing options. 2013-07-25 18:28:29 +00:00
config.cc Revert 8028 "Support associated payload type when registering Rt..." 2015-01-09 20:22:46 +00:00
config.h Revert 8028 "Support associated payload type when registering Rt..." 2015-01-09 20:22:46 +00:00
engine_configurations.h Add VP9 codec to VCM and vie_auto_test. 2014-11-01 06:10:48 +00:00
experiments.h Remove no longer used SkipEncodingUnusedStreams. 2014-07-22 07:17:17 +00:00
frame_callback.h Wire up statistics in video receive stream of new API 2014-02-07 12:06:29 +00:00
LICENSE Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY Consolidate all third party licenses in LICENSE_THIRD_PARTY. 2013-05-05 18:54:10 +00:00
OWNERS GN: Add BUILD.gn files + kjellander to OWNERS 2014-06-23 19:21:07 +00:00
PATENTS Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
PRESUBMIT.py PRESUBMIT.py: accept variants on the copyright message that are present in the codebase. 2014-05-23 17:27:18 +00:00
README.chromium Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
rtc_unittests.isolate Update all .isolate files for the new format. 2014-10-31 18:08:09 +00:00
supplement.gypi Roll chromium_revision 289723:291647 2014-08-25 14:16:32 +00:00
transport.h Rename newapi::Transport::SendRTP()->SendRtp(). 2013-11-20 12:17:04 +00:00
typedefs.h Annotate COMPILE_ASSERT with __attribute__((unused)). 2014-11-17 13:47:38 +00:00
video_decoder.h Add support for VP9 in webrtc::Call and video_loopback. 2014-11-04 19:41:15 +00:00
video_encoder.h Use int64_t more consistently for times, in particular for RTT values. 2015-01-12 21:51:21 +00:00
video_engine_tests.isolate Update isolate files for Android APK tests. 2014-11-13 08:35:05 +00:00
video_frame.h Report encoded frame size in VideoSendStream. 2014-12-01 15:23:21 +00:00
video_receive_stream.h Log configs when creating video streams in Call. 2015-01-15 10:09:39 +00:00
video_renderer.h Separate Call API/build files from video_engine/. 2013-10-28 16:32:01 +00:00
video_send_stream.h Log configs when creating video streams in Call. 2015-01-15 10:09:39 +00:00
webrtc_examples.gyp Move internal capture+render to build_with_chromium==0 condition 2015-01-20 11:40:45 +00:00
webrtc_perf_tests.isolate Update isolate files for Android APK tests. 2014-11-13 08:35:05 +00:00
webrtc_tests.gypi Move system_wrappers.gyp files to the proper directory. 2015-01-14 09:30:52 +00:00
webrtc.gyp Remove unnecessary dependencies from webrtc_all target. 2015-01-21 10:06:55 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.