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solenberg 07d9e545ff Revert of Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ (patchset #7 id:120001 of https://codereview.webrtc.org/2319583005/ ) 
Reason for revert:
Breaks downstream code

Original issue's description:
> Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/
>
> Only used in VoE tests so I'm moving it in under voice_engine/ until we've removed its usage and can deprecate.
>
> Note: submitting this with PRESUBMIT=false because the files do not adhere to style guide conventions and I'd rather change that in a separate CL.
>
> BUG=
> NOPRESUBMIT=true
>
> Committed: https://crrev.com/ade2a038a9290ee0c85d8c682eba5447aca943cd
> Cr-Commit-Position: refs/heads/master@{#14191}

TBR=kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review-Url: https://codereview.webrtc.org/2336123002
Cr-Commit-Position: refs/heads/master@{#14193}
2016-09-13 08:24:10 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Add printStackTrace method to CameraCapturer.
2016-09-12 14:35:21 +00:00
audio
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
base
webrtc/base: Use RTC_DCHECK() instead of assert()
2016-09-13 08:17:19 +00:00
build
MB: Move iOS GYP bots to use limited support config
2016-09-12 12:34:01 +00:00
call
Remove RTC_LOGGED_* macro.
2016-09-10 05:40:34 +00:00
common_audio
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
common_video
Revert "Optimize Android NV12 capture"
2016-09-09 23:15:15 +00:00
examples
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
libjingle
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
media
The current scheme for setting parameters and specifying the
2016-09-12 23:47:32 +00:00
modules
The current scheme for setting parameters and specifying the
2016-09-12 23:47:32 +00:00
p2p
Removing assert error when we fail to create a connection for a ping from an unknown address.
2016-09-10 03:48:08 +00:00
pc
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
sdk
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
stats
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
system_wrappers
Remove RTC_LOGGED_* macro.
2016-09-10 05:40:34 +00:00
test
Revert of Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ (patchset #7 id:120001 of https://codereview.webrtc.org/2319583005/ )
2016-09-13 08:24:10 +00:00
tools
Probing: Add support for exponential startup probing
2016-09-12 19:29:05 +00:00
video
Add periodic logging of number of captured and dropped frames in VideoCaptureInput. Logged every minute.
2016-09-12 07:10:53 +00:00
voice_engine
Revert of Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ (patchset #7 id:120001 of https://codereview.webrtc.org/2319583005/ )
2016-09-13 08:24:10 +00:00
.gitignore
…
BUILD.gn
Revert of Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ (patchset #7 id:120001 of https://codereview.webrtc.org/2319583005/ )
2016-09-13 08:24:10 +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
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.gyp
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
DEPS
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
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
…
rtc_unittests_apk.isolate
…
rtc_unittests.isolate
…
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_engine_tests_apk.isolate
…
video_engine_tests.isolate
…
video_frame.h
Introduce webrtc::VideoFrame::timestamp_us, and corresponding constructor.
2016-09-05 07:51:25 +00:00
video_receive_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
video_send_stream.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
webrtc_examples.gyp
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
webrtc_nonparallel_tests_apk.isolate
…
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
…
webrtc_perf_tests.isolate
Add an HD resolution perf test.
2016-08-11 14:01:03 +00:00
webrtc_tests.gypi
Revert of Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ (patchset #7 id:120001 of https://codereview.webrtc.org/2319583005/ )
2016-09-13 08:24:10 +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.