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philipel 77415f561d Revert of Disable SeqNumUnwrapper death tests to avoid breaking downstream builds. (patchset #1 id:1 of https://codereview.chromium.org/2985083002/ ) 
Reason for revert:
Creating revert to fix these tests.

Original issue's description:
> Disable SeqNumUnwrapper death tests to avoid breaking downstream builds.
>
> BUG=None
> TBR=stefan@webrtc.org
> NOTRY=true
>
> Review-Url: https://codereview.webrtc.org/2985083002
> Cr-Commit-Position: refs/heads/master@{#19155}
> Committed: 8e245561f2

TBR=stefan@webrtc.org
BUG=None

Review-Url: https://codereview.webrtc.org/2992643002
Cr-Commit-Position: refs/heads/master@{#19166}
2017-07-27 11:37:18 +00:00
..
api
Add "max_ipv6_networks" field to RTCConfiguration.
2017-07-26 23:50:11 +00:00
audio
Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"
2017-07-26 14:28:51 +00:00
base
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
call
Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.
2017-07-26 09:09:44 +00:00
common_audio
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
common_video
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
examples
Update native plugin dll for turn servers and video.
2017-07-25 23:04:31 +00:00
logging
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
media
Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.
2017-07-26 09:09:44 +00:00
modules
Revert of Disable SeqNumUnwrapper death tests to avoid breaking downstream builds. (patchset #1 id:1 of https://codereview.chromium.org/2985083002/ )
2017-07-27 11:37:18 +00:00
ortc
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
p2p
Relanding: Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting.
2017-07-26 23:09:33 +00:00
pc
Add "max_ipv6_networks" field to RTCConfiguration.
2017-07-26 23:50:11 +00:00
rtc_base
Relanding: Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting.
2017-07-26 23:09:33 +00:00
rtc_tools
Print general usage information for event_log_analyzer
2017-07-20 15:05:09 +00:00
sdk
Adding Obj-C binding for RTCConfiguration::max_ipv6_networks.
2017-07-27 01:25:43 +00:00
stats
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
system_wrappers
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
test
Rename isolated_output to test_output and add a method to get the test_output directory.
2017-07-25 14:31:18 +00:00
video
Report minimum PSNR in VideoQualityTest and save corresponding frame to file
2017-07-25 12:45:03 +00:00
voice_engine
Move total audio energy and duration tracking to AudioLevel and protect with existing critial section.
2017-07-20 16:57:42 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_types.h
Add parsing/serializing for MID RTP header extension.
2017-07-21 17:33:25 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_receive_stream.h
Report interframe delay sum in old GetStats
2017-07-07 08:26:24 +00:00
video_send_stream.h
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
2017-07-11 10:44:17 +00:00
webrtc.gni
Adds detection of audio glitches for playout on iOS (reland)
2017-06-13 16:00:18 +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.