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webrtc_m130/webrtc
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tommi@webrtc.org 11c5db01af Revert 8273 "Temporarily change ThreadPosix to CHECK (crash) if ..." 
> Temporarily change ThreadPosix to CHECK (crash) if we ever spend more than 30 seconds waiting for thread shutdown. There are cases on build bots where it looks like we're hitting this problem, but reproducing locally has been a struggle.
> 
> TBR=pbos@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/41799004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38039004

Cr-Commit-Position: refs/heads/master@{#8300}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8300 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 16:31:54 +00:00
..
base
Clean kForever from basictypes and move it to the interfaces that actually have it.
2015-02-09 14:19:39 +00:00
build
Restructure GYP for vp9, opus and direct trace
2015-02-09 10:01:40 +00:00
common_audio
Move channel_buffer.{h,cc} to common_audio.
2015-01-28 19:57:44 +00:00
common_video
Cleanup: unify rotation to be enum based instead of int for degree.
2015-02-09 01:29:45 +00:00
examples
Add arraysize() macro from Chromium, and make use of it in a few places.
2015-01-28 21:37:13 +00:00
libjingle
Enable Clang warning implicit-fallthrough and annotate the code.
2015-01-28 18:38:13 +00:00
modules
Revert r8297 "Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig."
2015-02-09 15:06:42 +00:00
overrides
Clean kForever from basictypes and move it to the interfaces that actually have it.
2015-02-09 14:19:39 +00:00
p2p
Remove USE_WEBRTC_DEV_BRANCH.
2015-02-09 10:17:12 +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
Revert 8273 "Temporarily change ThreadPosix to CHECK (crash) if ..."
2015-02-09 16:31:54 +00:00
test
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +00:00
tools
Re-land "Remove <(webrtc_root) from source file entries."
2015-01-29 14:30:41 +00:00
video
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +00:00
video_engine
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +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
…
BUILD.gn
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +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
…
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
…
LICENSE_THIRD_PARTY
…
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
…
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
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +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
Re-land "Remove <(webrtc_root) from source file entries."
2015-01-29 14:30:41 +00:00
webrtc.gyp
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +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.