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platform-external-webrtc/webrtc
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kjellander@webrtc.org 665d861115 Restore webrtc_base target until r7140 is rolled into Chromium. 
In r7140 the webrtc_base target was renamed to rtc_base. This
breaks our FYI bots for rolling WebRTC in Chromium's DEPS.
By re-adding a None target named webrtc_base, this transition
should be smoother.

TBR=henrikg@webrtc.org,
TESTED=Passed build/gyp_chromium on a Chromium checkout with src/third_party/webrtc replaced by a mount like this:
cd /path/to/chromium/src
sudo mount --bind /path/to/webrtc/trunk/webrtc third_party/webrtc

Review URL: https://webrtc-codereview.appspot.com/23589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7150 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 09:22:13 +00:00
..
base
Restore webrtc_base target until r7140 is rolled into Chromium.
2014-09-11 09:22:13 +00:00
build
Add schannel webrtc_base build using a new use_schannel gyp variable.
2014-09-10 18:06:47 +00:00
common_audio
Add ctors to ChannelBuffer to enable copying on construction.
2014-09-08 20:27:04 +00:00
common_video
Expose VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +00:00
examples
Fix compile error on JDK 1.7.
2014-09-10 12:35:59 +00:00
libjingle/xmllite
Put base tests in webrtc_tests.gyp
2014-09-10 17:28:19 +00:00
modules
audio_processing_unittests: Enabled ApmTest.Process for all platforms but Android
2014-09-11 08:36:35 +00:00
overrides/webrtc/base
Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h.
2014-05-21 16:52:14 +00:00
sound
Put base tests in webrtc_tests.gyp
2014-09-10 17:28:19 +00:00
system_wrappers
Put base tests in webrtc_tests.gyp
2014-09-10 17:28:19 +00:00
test
Expose VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +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 VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +00:00
video_engine
Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.
2014-09-11 06:20:28 +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
…
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
…
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
…
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 VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +00:00
video_engine_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video_frame.h
Expose VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +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
Fix rm command for class cleanup in r7091
2014-09-06 22:11:28 +00:00
webrtc_perf_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
webrtc_tests.gypi
Put base tests in webrtc_tests.gyp
2014-09-10 17:28:19 +00:00
webrtc.gyp
Put base tests in webrtc_tests.gyp
2014-09-10 17:28:19 +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.