generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
933d88af58
webrtc/webrtc
History
pbos@webrtc.org c3091a6c26 Remove the 'webrtc_test_video_render_dependencies' target. 
This target is no longer needed and is causing linking errors on XCode.

R=andresp@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28519004

Patch from Alexandre Gouaillard <agouaillard@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7226 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18 17:22:18 +00:00
..
base Do not require synchronization access on the thread if called from rtc::Thread::WrapCurrent. 2014-09-18 16:45:21 +00:00
build Split video_render_module implementation into default and internal implementation. 2014-09-18 08:58:15 +00:00
common_audio gn: Fix cflags usage 2014-09-16 17:57:02 +00:00
common_video Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" 2014-09-17 09:02:25 +00:00
examples Split video engine android initialization into each internal module initialization. 2014-09-17 11:44:51 +00:00
libjingle/xmllite Put base tests in webrtc_tests.gyp 2014-09-10 17:28:19 +00:00
modules Trying to fix Chrome FYI bots. 2014-09-18 15:50:05 +00:00
overrides webrtc/overrides: add OWNERS-file. 2014-09-17 08:04:28 +00:00
sound Put base tests in webrtc_tests.gyp 2014-09-10 17:28:19 +00:00
system_wrappers Add a target for the approved subset of rtc_base. 2014-09-16 01:03:29 +00:00
test Remove the 'webrtc_test_video_render_dependencies' target. 2014-09-18 17:22:18 +00:00
tools Fix printing of error stack in rtcbot when a test fails via test.fail(). 2014-09-10 14:35:35 +00:00
video Expose VP8/H264 defaults through video_encoder.h. 2014-09-18 12:42:28 +00:00
video_engine Split video_render_module implementation into default and internal implementation. 2014-09-18 08:58:15 +00:00
voice_engine Calculating round-trip-time in send-only channel in VoE. 2014-09-11 07:51:53 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
BUILD.gn Fix the RTC+Chromium GN build. 2014-09-09 19:15:33 +00:00
call.h Network up/down signaling in Call. 2014-09-03 16:17:12 +00:00
common_types.h Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. 2014-07-11 13:44:02 +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 Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
config.h Check before send/receive rtp header extensions. 2014-07-20 15:27:35 +00:00
engine_configurations.h Add boilerplate code for H.264. 2014-07-04 12:42:07 +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
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 Add CHECK and friends from Chromium. 2014-08-28 16:28:26 +00:00
video_encoder.h Expose VP8/H264 defaults through video_encoder.h. 2014-09-18 12:42:28 +00:00
video_engine_tests.isolate Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
video_frame.h Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" 2014-09-17 09:02:25 +00:00
video_receive_stream.h Change return value for number of discarded packets to be int. 2014-09-04 07:07:44 +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 Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. 2014-07-11 13:44:02 +00:00
webrtc_examples.gyp Split video_render_module implementation into default and internal implementation. 2014-09-18 08:58:15 +00:00
webrtc_perf_tests.isolate Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
webrtc_tests.gypi Remove the 'webrtc_test_video_render_dependencies' target. 2014-09-18 17:22:18 +00:00
webrtc.gyp Split video_render_module implementation into default and internal implementation. 2014-09-18 08:58:15 +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.