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kjellander@webrtc.org e01264306b Remove temporary GYP targets 
The Chromium libjingle.gyp has now been updated in
https://codereview.chromium.org/907343002/ and the changes
in https://webrtc-codereview.appspot.com/35099004/ are rolled
into Chromium. Therefore these targets are no longer needed.

BUG=
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41919004

Cr-Commit-Position: refs/heads/master@{#8352}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8352 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-12 13:48:39 +00:00
..
base Use std::min and std::max instead of self-defined functions such as rtc::_min/_max. 2015-02-12 11:55:32 +00:00
build Remove potential deadlock in RTPSenderAudio. 2015-02-12 12:20:50 +00:00
common_audio WebRtc_GetCPUFeaturesARM is only available on android 2015-02-11 17:03:24 +00:00
common_video CVO capturer feature: allow unrotated frame flows through the capture pipeline. 2015-02-11 18:38:53 +00:00
examples Refactoring WebRTC Java/JNI audio recording in C++ and Java. 2015-02-11 08:39:19 +00:00
libjingle Use std::min and std::max instead of self-defined functions such as rtc::_min/_max. 2015-02-12 11:55:32 +00:00
modules Remove temporary GYP targets 2015-02-12 13:48:39 +00:00
overrides Use std::min and std::max instead of self-defined functions such as rtc::_min/_max. 2015-02-12 11:55:32 +00:00
p2p Use std::min and std::max instead of self-defined functions such as rtc::_min/_max. 2015-02-12 11:55:32 +00:00
sound Use std::min and std::max instead of self-defined functions such as rtc::_min/_max. 2015-02-12 11:55:32 +00:00
system_wrappers Fix false positive DHECK in event_posix.cc 2015-02-11 15:19:22 +00:00
test Rename GYP and GN targets for video capture+render. 2015-02-11 07:47:47 +00:00
tools Re-land "Remove <(webrtc_root) from source file entries." 2015-01-29 14:30:41 +00:00
video Use an external-only VideoRenderModule in Call. 2015-02-12 10:48:55 +00:00
video_engine Remove ViENetwork::SetBandwidthEstimationConfig() interface since dynamically changing BWE settings isn't necessary now that AIMD is the default. 2015-02-12 13:21:27 +00:00
voice_engine audio_processing: Now records mic volume level also when using new AGC 2015-02-06 19:44:46 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
BUILD.gn Rename GYP and GN targets for video capture+render. 2015-02-11 07:47:47 +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 Add method for incrementing RtpPacketCounter. Removes duplicate code. 2015-02-04 08:35:21 +00:00
common.gyp Make it easier to use external libyuv + cleanup GYP files. 2015-01-26 19:17:26 +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 Changing include guard in frame_callback.h. 2015-02-03 14:51:39 +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 Enable Clang warning implicit-fallthrough and annotate the code. 2015-01-28 18:38:13 +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 Remove default arguments in EncodedImageCallback. 2015-02-09 09:14:48 +00:00
video_engine_tests.isolate Update isolate files for Android APK tests. 2014-11-13 08:35:05 +00:00
video_frame.h CVO capturer feature: allow unrotated frame flows through the capture pipeline. 2015-02-11 18:38:53 +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 Use VideoReceiveStream as an ExternalRenderer. 2015-02-09 15:15:24 +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 Rename GYP and GN targets for video capture+render. 2015-02-11 07:47:47 +00:00
webrtc.gyp Rename GYP and GN targets for video capture+render. 2015-02-11 07:47:47 +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.