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webrtc/webrtc
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aluebs@webrtc.org 28b54671cb Make all comments whole sentences in ns_core 
This is done to make the code more readable.
It generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7547 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 20:56:53 +00:00
..
base Only configure the SSL library in one place. 2014-10-27 18:13:40 +00:00
build Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." 2014-10-17 22:03:39 +00:00
common_audio common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16 2014-10-28 13:03:10 +00:00
common_video GN: Add common configs to all targets. 2014-09-28 17:37:22 +00:00
examples Split video engine android initialization into each internal module initialization. 2014-09-17 11:44:51 +00:00
libjingle/xmllite Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." 2014-10-17 22:03:39 +00:00
modules Make all comments whole sentences in ns_core 2014-10-28 20:56:53 +00:00
overrides webrtc/overrides: add OWNERS-file. 2014-09-17 08:04:28 +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 scoped_ptr.h: Renames function and change namespace scope to fix conflicts with Chromium not detected by the FYI bots. 2014-10-28 18:06:42 +00:00
test Move (test) RtpFileReader to a lightweight target. 2014-10-27 18:01:03 +00:00
tools Adding the subtool rtcBot report visualizer 2014-10-24 09:26:16 +00:00
video Move min transmit bitrate to VideoEncoderConfig. 2014-10-24 09:23:21 +00:00
video_engine Add macros and APIs for webrtc histograms. 2014-10-23 12:57:56 +00:00
voice_engine Add a simple AudioConverter class. 2014-10-27 18:18:17 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
BUILD.gn GN: Enable libvpx, add link target and convert some test targets 2014-09-30 18:05:02 +00:00
call.h Remove -1 from Call::Config::start_bitrate_bps. 2014-10-14 11:52:10 +00:00
common_types.h Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..." 2014-10-17 18:54:46 +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 Move min transmit bitrate to VideoEncoderConfig. 2014-10-24 09:23:21 +00:00
config.h Move min transmit bitrate to VideoEncoderConfig. 2014-10-24 09:23:21 +00:00
engine_configurations.h Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..." 2014-10-17 18:54:46 +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 Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest. 2014-09-30 14:21:10 +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 Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..." 2014-10-17 18:54:46 +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 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 Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
webrtc_tests.gypi Add macros and APIs for webrtc histograms. 2014-10-23 12:57:56 +00:00
webrtc.gyp Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." 2014-10-17 22:03:39 +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.