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Honghai Zhang b73d269707 Replace RelayPort with TurnPort in p2ptransportchannel tests. 
Also remove the relay servers in the tests.
Most of the code and the downstream apps are using TurnPort, not RelayPort. Most of the tests in this file are not using RelayPort anyway.

BUG=None
R=deadbeef@webrtc.org

Review URL: https://codereview.webrtc.org/2380923002 .

Committed: https://crrev.com/c8d21712dde64c7d613d1ea56c840438505a909f
Cr-Original-Commit-Position: refs/heads/master@{#14441}
Cr-Commit-Position: refs/heads/master@{#14446}
2016-09-30 05:46:16 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Let ViEEncoder handle resolution changes.
2016-09-30 05:39:15 +00:00
audio
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
base
Do not spam "Connect failed with 101/65" in logs.
2016-09-29 19:59:44 +00:00
build
Roll chromium_revision f86fb54ec3..dd442d4812 (420104:421425)
2016-09-28 07:45:12 +00:00
call
Let ViEEncoder handle resolution changes.
2016-09-30 05:39:15 +00:00
common_audio
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
common_video
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
examples
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
libjingle
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
media
Let ViEEncoder handle resolution changes.
2016-09-30 05:39:15 +00:00
modules
Move RTCPHelp::RTCPReceiveInformation inside RTCPReceiver
2016-09-29 22:28:12 +00:00
p2p
Replace RelayPort with TurnPort in p2ptransportchannel tests.
2016-09-30 05:46:16 +00:00
pc
Update WebRTC to build against libsrtp 2.0
2016-09-28 21:08:53 +00:00
sdk
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
stats
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
system_wrappers
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
test
Let ViEEncoder handle resolution changes.
2016-09-30 05:39:15 +00:00
tools
Revert of Fix race / crash in OnNetworkRouteChanged(). (patchset #5 id:80001 of https://codereview.webrtc.org/2366333003/ )
2016-09-29 11:19:42 +00:00
video
Let ViEEncoder handle resolution changes.
2016-09-30 05:39:15 +00:00
voice_engine
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
.gitignore
…
BUILD.gn
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
call.h
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
2016-09-01 08:17:43 +00:00
codereview.settings
…
common_types.cc
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
common_types.h
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
common.gyp
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
config.cc
Let ViEEncoder handle resolution changes.
2016-09-30 05:39:15 +00:00
config.h
Let ViEEncoder handle resolution changes.
2016-09-30 05:39:15 +00:00
DEPS
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
engine_configurations.h
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
…
video_encoder.h
Add codec name to CodecSpecificInfo and get the codec name stats from there instead.
2016-09-01 07:21:18 +00:00
video_frame.h
Let ViEEncoder handle resolution changes.
2016-09-30 05:39:15 +00:00
video_receive_stream.h
Expose Ivf logging through the native API
2016-09-28 13:19:53 +00:00
video_send_stream.h
Add VideoSendStream::Stats::prefered_media_bitrate_bps
2016-09-29 09:49:01 +00:00
webrtc_examples.gyp
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +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.