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jmarusic@webrtc.org 2acec4cc32 Enhanced documentation. Replaced DCHECK with CHECK. 
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34309004

Cr-Commit-Position: refs/heads/master@{#8461}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8461 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 15:28:14 +00:00
..
base
Add functions rtc::AtomicOps::Load and rtc::RefCountedObject::HasOneRef
2015-02-21 13:23:46 +00:00
build
Roll chromium_revision 601e6f3..b0c3ed3 (315263:316737)
2015-02-18 10:38:11 +00:00
common_audio
WebRtc_GetCPUFeaturesARM is only available on android
2015-02-11 17:03:24 +00:00
common_video
I420VideoFrame: Remove functions set_width, set_height, and ResetSize
2015-02-19 15:35:50 +00:00
examples
Refactoring WebRTC Java/JNI audio track in C++ and Java.
2015-02-23 11:54:41 +00:00
libjingle
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
2015-02-12 11:55:32 +00:00
modules
Enhanced documentation. Replaced DCHECK with CHECK.
2015-02-23 15:28:14 +00:00
overrides
Add arraysize to overrides to avoid macros redefinitions in Chromium
2015-02-14 02:46:13 +00:00
p2p
Change default from GICE to ICE5245 for SDP offers
2015-02-20 02:20:19 +00:00
sound
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
2015-02-12 11:55:32 +00:00
system_wrappers
Copy SetThreadName from webrtc/base/thread.cc into thread_win.cc
2015-02-19 19:18:54 +00:00
test
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
tools
Re-land "Remove <(webrtc_root) from source file entries."
2015-01-29 14:30:41 +00:00
video
Enable NACK under SendsAndReceivesH264.
2015-02-23 11:12:18 +00:00
video_engine
Move SetTargetSendBitrates logic from default module to payload router.
2015-02-23 07:45:45 +00:00
voice_engine
Adding two new stats to VoiceReceiverInfo
2015-02-18 15:24:39 +00:00
.gitignore
…
BUILD.gn
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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 RtcpPacketTypeCounter stats to new API.
2015-02-19 12:47:45 +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
Add RtcpPacketTypeCounter stats to new API.
2015-02-19 12:47:45 +00:00
engine_configurations.h
Delete all codec-specific subclasses of ACMGenericCodec
2015-02-23 09:26:51 +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 601e6f3..b0c3ed3 (315263:316737)
2015-02-18 10:38:11 +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
I420VideoFrame: Remove functions set_width, set_height, and ResetSize
2015-02-19 15:35:50 +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
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
webrtc_tests.gypi
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
webrtc.gyp
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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.