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platform-external-webrtc/webrtc
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pbos c7a6569713 Revert of Disable failing modules_unittests for UBSan. (patchset #1 id:40001 of https://codereview.webrtc.org/1915813002/ ) 
Reason for revert:
Fix upstream should've landed in our repository.

Original issue's description:
> Disable failing modules_unittests for UBSan.
>
> BUG=webrtc:5820
> TBR=pbos@webrtc.org
>
> Committed: https://crrev.com/c23bf2e54d922486254cdd7657aafceaa958ce25
> Cr-Commit-Position: refs/heads/master@{#12482}

TBR=kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5820

Review-Url: https://codereview.webrtc.org/1937153002
Cr-Commit-Position: refs/heads/master@{#12647}
2016-05-06 19:50:08 +00:00
..
api
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
audio
Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )"
2016-05-04 15:13:06 +00:00
base
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
build
Fix BoringSSL license path.
2016-05-06 18:19:54 +00:00
call
Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )"
2016-05-04 15:13:06 +00:00
common_audio
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
common_video
This cl deletes the class webrtc::VideoRendererCallback.
2016-05-03 08:09:17 +00:00
examples
Update CPU monitor to use moving averages.
2016-05-04 18:04:18 +00:00
libjingle
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
media
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
modules
Revert of Disable failing modules_unittests for UBSan. (patchset #1 id:40001 of https://codereview.webrtc.org/1915813002/ )
2016-05-06 19:50:08 +00:00
p2p
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
pc
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
sdk
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
system_wrappers
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
test
Corrected bug in checking the third number and added extra checks
2016-05-06 13:12:27 +00:00
tools
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
video
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
voice_engine
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
Partial revert of Enable -Winconsistent-missing-override flag. (patchset #5 id:80001 of https://codereview.webrtc.org/1921653002/ )
2016-05-04 09:37:50 +00:00
call.h
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
…
common_types.h
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
2016-05-03 06:44:11 +00:00
common.gyp
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Change transport sequence number extension strings to specify what revision is implemented.
2016-03-01 15:27:27 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
DEPS
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Make pbos owner of additional video files.
2016-02-26 12:29:17 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Remove use_openssl from webrtc
2016-04-18 18:13:17 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Update attributes to match gclibc's ansidecl.h
2016-01-08 12:51:47 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
2016-04-29 09:39:33 +00:00
video_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Build dynamic iOS SDK.
2016-04-27 08:54:27 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Revert of Remove the rtc_relative_path GYP variable and similar defines (patchset #1 id:1 of https://codereview.webrtc.org/1925733002/ )
2016-05-04 08:54:39 +00:00
webrtc.gyp
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +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.