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webrtc/webrtc
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zstein 6dfd53a81e Rename PeerConnection::OnIceConnectionChange to OnIceConnectionStateChange 
for consistency with the WebRTC 1.0 standard as suggested in a TODO.

BUG=None

Review-Url: https://codereview.webrtc.org/2732663004
Cr-Commit-Position: refs/heads/master@{#17077}
2017-03-06 21:49:03 +00:00
..
api Reland "Enable GN check for webrtc/examples" 2017-03-06 08:29:21 +00:00
audio Adding placeholder for low bandwidth audio test 2017-03-06 12:01:16 +00:00
base Fixing race between CallbackCanceled and CancelCallback in AsyncInvoker. 2017-03-03 18:33:18 +00:00
call Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule. 2017-03-03 11:20:24 +00:00
common_audio Enable GN check for webrtc/common_audio 2017-03-01 15:07:10 +00:00
common_video Ignore aud and sei NALus when parsing bitstream. 2017-03-06 10:49:36 +00:00
examples Add kjellander to OWNERS for *.py in examples/android{app,tests} 2017-03-06 11:56:57 +00:00
logging Remove unused include from rtc_event_log_parser.cc 2017-02-28 09:46:19 +00:00
media WebRtcVideoChannel2Test::SetRecvCodecsAcceptsMultipleVideoCodecs passes now. 2017-03-06 20:09:24 +00:00
modules Fix segmentation fault in AudioEncoderOpusTest.EncodeAtMinBitrate. 2017-03-06 14:49:27 +00:00
ortc Revert of Enable GN check for webrtc/{ortc,p2p} (patchset #4 id:60001 of https://codereview.webrtc.org/2714263004/ ) 2017-03-04 23:08:44 +00:00
p2p Replace StunMessage's vector<StunAttribute*>* with a 2017-03-06 21:36:05 +00:00
pc Rename PeerConnection::OnIceConnectionChange to OnIceConnectionStateChange 2017-03-06 21:49:03 +00:00
sdk Android HW decoder: Don't check slice_height for texture output 2017-03-06 13:20:49 +00:00
stats Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect. 2017-03-01 09:02:45 +00:00
system_wrappers Check __GLIBC_PREREQ availability before use. 2017-03-06 15:34:06 +00:00
test Revert of Added large room scenario to full-stack tests. Added thumbnail streams functionality to call test/v… (patchset #8 id:140001 of https://codereview.webrtc.org/2730073002/ ) 2017-03-06 15:35:13 +00:00
tools Rename webrtc::PacketInfo to webrtc::PacketFeedback. This resolves ambiguity with a similarly named RTCPReceiver::PacketInformation and RtpPacketizerVp9::PacketInfo. 2017-03-06 13:32:21 +00:00
video Revert of Added large room scenario to full-stack tests. Added thumbnail streams functionality to call test/v… (patchset #8 id:140001 of https://codereview.webrtc.org/2730073002/ ) 2017-03-06 15:35:13 +00:00
voice_engine Rename webrtc::PacketInfo to webrtc::PacketFeedback. This resolves ambiguity with a similarly named RTCPReceiver::PacketInformation and RtpPacketizerVp9::PacketInfo. 2017-03-06 13:32:21 +00:00
.gitignore
BUILD.gn Remove HAVE_SRTP define and unmaintained code. 2017-03-06 19:32:22 +00:00
call.h Moved call.h and most of api/call/* into call/ 2016-12-07 12:53:04 +00:00
codereview.settings
common_types.cc Reject XR TargetBitrate items with unsupported layer indices 2016-12-06 14:09:00 +00:00
common_types.h Remove the Windows Wave audio device implementation. 2017-02-17 22:48:07 +00:00
config.cc Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. 2017-02-26 02:15:09 +00:00
config.h Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. 2017-02-26 02:15:09 +00:00
DEPS Move VideoFrame and related declarations to webrtc/api/video. 2017-01-10 15:44:26 +00:00
LICENSE
LICENSE_THIRD_PARTY Delete unused spreadsort implementation. 2016-12-05 11:03:26 +00:00
no_op_function.cc Moving no_op_function.cc out of webrtc/build 2017-01-24 13:25:12 +00:00
OWNERS Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it 2016-11-29 10:52:22 +00:00
PATENTS
PRESUBMIT.py
README.chromium
typedefs.h Only define NO_RETURN if undefined 2017-01-30 08:54:19 +00:00
video_decoder.h Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ ) 2017-02-09 12:53:45 +00:00
video_encoder.h Move usage of QualityScaler to ViEEncoder. 2016-11-29 09:44:22 +00:00
video_frame.h Move VideoFrame and related declarations to webrtc/api/video. 2017-01-10 15:44:26 +00:00
video_receive_stream.h Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ ) 2017-02-09 12:53:45 +00:00
video_send_stream.h Remove FlexfecConfig and replace with specific struct in VideoSendStream. 2017-01-16 14:59:19 +00:00
webrtc.gni Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ ) 2017-03-04 03:41:59 +00:00
whitespace.txt Moving whitespace file up by one folder 2017-01-24 13:19:42 +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.