webrtc_m130

History

Zeke Chin c8f9230667 Remove OWNERS check for webrtc/sdk/BUILD.gn

Bug: 0
Change-Id: I3ae2fe48367bf3cac0ce1fb4b4cc2f096456a68a
Reviewed-on: https://chromium-review.googlesource.com/570822
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19015}

2017-07-14 09:12:42 +00:00

api

API for periodically regathering ICE candidates

2017-07-11 21:49:38 +00:00

audio

Use relative paths in GN files.

2017-07-11 13:20:45 +00:00

base

Revert of Delete SignalThread class. (patchset #20 id:380001 of https://codereview.webrtc.org/2915253002/ )

2017-07-11 23:56:05 +00:00

call

Use relative paths in GN files.

2017-07-11 13:20:45 +00:00

common_audio

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

common_video

Use relative paths in GN files.

2017-07-11 13:20:45 +00:00

examples

Revert of Delete SignalThread class. (patchset #20 id:380001 of https://codereview.webrtc.org/2915253002/ )

2017-07-11 23:56:05 +00:00

logging

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

media

Use relative paths in GN files.

2017-07-11 13:20:45 +00:00

modules

Output DeviceName from various windows ScreenCapturer related implementations

2017-07-13 23:05:24 +00:00

ortc

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

p2p

Add additional check when setting RTCConfiguration

2017-07-13 21:05:28 +00:00

Add additional check when setting RTCConfiguration

2017-07-13 21:05:28 +00:00

rtc_base

Revert of Make the default ctor of rtc::Thread, protected (patchset #3 id:40001 of https://codereview.webrtc.org/2981623002/ )

2017-07-13 14:06:39 +00:00

rtc_tools

Use relative paths in GN files.

2017-07-11 13:20:45 +00:00

sdk

Remove OWNERS check for webrtc/sdk/BUILD.gn

2017-07-14 09:12:42 +00:00

stats

Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )

2017-07-10 12:58:42 +00:00

system_wrappers

Fix DetectNumberOfCores for 32-bit processes

2017-07-13 17:06:12 +00:00

test

Use relative paths in GN files.

2017-07-11 13:20:45 +00:00

video

Add rtpdump and rtc log functionality to screenshare_loopback and video_loopback

2017-07-13 07:47:03 +00:00

voice_engine

TransmitMixer: Check GetSendCodec return value.

2017-07-11 15:19:31 +00:00

.gitignore

…

BUILD.gn

Use relative paths in GN files.

2017-07-11 13:20:45 +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

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +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.