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deadbeef 71cfe690b7 For TestResolverShutdown, use address that can't be resolved. 
This test only currently works because stun.l.google.com has an IPv4
address and the TURN port is created with an IPv6 address. But the test
would start failing if/when it starts providing an IPv6 address. Which
may already be happening, as indicated by a recent test failure.

Review URL: https://codereview.webrtc.org/1290233003

Cr-Commit-Position: refs/heads/master@{#9841}
2015-09-02 02:01:40 +00:00
..
base
Fixes for PNaCl build of remoting client plugin with GN.
2015-09-01 18:36:50 +00:00
build
Re-enable LLVM LTO on Neon targets.
2015-08-18 04:05:59 +00:00
common_audio
Convert some more things to size_t.
2015-08-29 00:31:15 +00:00
common_video
VideoFrameBuffer: Make non-const data access explicit
2015-08-26 14:06:30 +00:00
examples
Handle all RTCICEConnectionState values in ARDVideoCallViewController
2015-09-01 22:35:42 +00:00
libjingle
Remove webrtc/libjingle/{examples,session}.
2015-06-12 11:13:46 +00:00
modules
Fixes for PNaCl build of remoting client plugin with GN.
2015-09-01 18:36:50 +00:00
overrides
Attempt at fixing error on the Chrome Windows FYI bots.
2015-05-25 19:22:23 +00:00
p2p
For TestResolverShutdown, use address that can't be resolved.
2015-09-02 02:01:40 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Update a ton of audio code to use size_t more correctly and in general reduce
2015-08-24 21:52:45 +00:00
test
Add send transports to individual webrtc::Call streams.
2015-08-28 11:07:15 +00:00
tools
Update a ton of audio code to use size_t more correctly and in general reduce
2015-08-24 21:52:45 +00:00
video
Added send stream test case for VP9 header.
2015-09-01 14:01:54 +00:00
video_engine
Wire up currently-received video codec to stats.
2015-08-28 14:35:40 +00:00
voice_engine
Stylizing AudioConferenceMixer.
2015-08-31 14:04:44 +00:00
.gitignore
…
audio_receive_stream.h
Control combined_audio_video_bwe with config bool.
2015-07-24 14:10:25 +00:00
audio_send_stream.h
Add send transports to individual webrtc::Call streams.
2015-08-28 11:07:15 +00:00
BUILD.gn
Includes webrtc/build/protoc.gypi instead of build/protoc.gypi
2015-07-30 10:45:24 +00:00
call.h
Add send transports to individual webrtc::Call streams.
2015-08-28 11:07:15 +00:00
codereview.settings
Update deeper codereview.settings files to match the root.
2015-06-24 19:40:39 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Update a ton of audio code to use size_t more correctly and in general reduce
2015-08-24 21:52:45 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Make Config::default_value leak instead of having an exit-time destructor.
2015-05-23 00:50:33 +00:00
config.cc
Add support for transport wide sequence numbers
2015-08-03 11:38:48 +00:00
config.h
Add support for transport wide sequence numbers
2015-08-03 11:38:48 +00:00
engine_configurations.h
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
frame_callback.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
libjingle_examples.gyp
Add stats overlay to iOS AppRTCDemo.
2015-08-14 18:00:11 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
Define Stream base classes
2015-07-16 07:30:20 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
…
typedefs.h
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
video_decoder.h
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
video_encoder.h
Allow for framerate reduction for HW encoder.
2015-07-13 23:26:40 +00:00
video_engine_tests.isolate
…
video_frame.h
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
video_receive_stream.h
Wire up currently-received video codec to stats.
2015-08-28 14:35:40 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Add send transports to individual webrtc::Call streams.
2015-08-28 11:07:15 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Includes webrtc/build/protoc.gypi instead of build/protoc.gypi
2015-07-30 10:45:24 +00:00
webrtc.gyp
Includes webrtc/build/protoc.gypi instead of build/protoc.gypi
2015-07-30 10:45:24 +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.