generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
795d003770
webrtc/webrtc
History
kwiberg@webrtc.org b8425bc4f3 Don't use DCHECK when you need the side effects... 
R=pbos@webrtc.org
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7625 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 22:10:18 +00:00
..
base Only configure the SSL library in one place. 2014-10-27 18:13:40 +00:00
build Add VP9 codec to VCM and vie_auto_test. 2014-11-01 06:10:48 +00:00
common_audio Add format members to AudioConverter for DCHECKing. 2014-11-03 21:32:14 +00:00
common_video Update all .isolate files for the new format. 2014-10-31 18:08:09 +00:00
examples Update Android projects to API level 21. 2014-10-31 23:26:10 +00:00
libjingle move xmpp and p2p to webrtc 2014-10-28 22:20:11 +00:00
modules Don't use DCHECK when you need the side effects... 2014-11-04 22:10:18 +00:00
overrides webrtc/overrides: add OWNERS-file. 2014-09-17 08:04:28 +00:00
p2p move xmpp and p2p to webrtc 2014-10-28 22:20:11 +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 Add stats for video: 2014-11-03 14:40:38 +00:00
test Add support for VP9 in webrtc::Call and video_loopback. 2014-11-04 19:41:15 +00:00
tools Update all .isolate files for the new format. 2014-10-31 18:08:09 +00:00
video Add support for VP9 in webrtc::Call and video_loopback. 2014-11-04 19:41:15 +00:00
video_engine Reworked paced sender queue 2014-11-04 16:27:16 +00:00
voice_engine Update all .isolate files for the new format. 2014-10-31 18:08:09 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
BUILD.gn arm64 iOS build. 2014-10-31 00:14:39 +00:00
call.h Remove -1 from Call::Config::start_bitrate_bps. 2014-10-14 11:52:10 +00:00
common_types.h Add VP9 codec to VCM and vie_auto_test. 2014-11-01 06:10:48 +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 Implement conference-mode temporal-layer screencast. 2014-10-31 13:08:10 +00:00
config.h Implement conference-mode temporal-layer screencast. 2014-10-31 13:08:10 +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 Add CHECK and friends from Chromium. 2014-08-28 16:28:26 +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 Add VP9 codec to VCM and vie_auto_test. 2014-11-01 06:10:48 +00:00
video_engine_tests.isolate Update all .isolate files for the new format. 2014-10-31 18:08:09 +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 Configure A/V sync in WebRtcVideoEngine2. 2014-10-31 12:59:34 +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 min transmit bitrate to VideoEncoderConfig. 2014-10-24 09:23:21 +00:00
webrtc_examples.gyp Add macros and APIs for webrtc histograms. 2014-10-23 12:57:56 +00:00
webrtc_perf_tests.isolate Update all .isolate files for the new format. 2014-10-31 18:08:09 +00:00
webrtc_tests.gypi Adds support for finch experiments to video_loopback. 2014-11-04 14:57:14 +00:00
webrtc.gyp Add VP9 codec to VCM and vie_auto_test. 2014-11-01 06:10:48 +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.