webrtc_m130

History

henrika 3296256f0e Fixing lint issue

NOTRY=TRUE
BUG=NONE

Review-Url: https://codereview.webrtc.org/2988533002
Cr-Commit-Position: refs/heads/master@{#19110}

2017-07-21 14:28:41 +00:00

api

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

audio

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

base

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

call

Remove remains of webrtc/base

2017-07-19 17:40:47 +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

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

logging

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

media

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

modules

Fixing lint issue

2017-07-21 14:28:41 +00:00

ortc

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

p2p

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

Introduce RtpTransportInternal and SrtpTransport.

2017-07-19 20:38:02 +00:00

rtc_base

Add common TLS extensions

2017-07-21 00:10:31 +00:00

rtc_tools

Print general usage information for event_log_analyzer

2017-07-20 15:05:09 +00:00

sdk

Set target API to 18 for MediaCodecUtils.

2017-07-21 10:30:02 +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

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

video

Remove remains of webrtc/base

2017-07-19 17:40:47 +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

…

BUILD.gn

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

codereview.settings

…

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

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PRESUBMIT.py

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README.chromium

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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.