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webrtc_m130/webrtc
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nisse e077ee472a Deleted unused method EstimateMTU, and the WinPing class. 
BUG=None

Review-Url: https://codereview.webrtc.org/2866183004
Cr-Commit-Position: refs/heads/master@{#18077}
2017-05-10 08:56:54 +00:00
..
api
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
audio
Drop deprecated AudioFrameOperations::Scale method signatures
2017-05-09 09:43:51 +00:00
base
Deleted unused method EstimateMTU, and the WinPing class.
2017-05-10 08:56:54 +00:00
call
Delete helper class MediaTypePacketReceiver.
2017-05-09 12:12:00 +00:00
common_audio
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
2017-04-28 12:24:50 +00:00
common_video
Actually move CoreVideoFrameBuffer from webrtc/common_video/ to webrtc/sdk/objc/
2017-05-08 12:32:05 +00:00
examples
Test for Gradle project generation.
2017-05-09 09:10:57 +00:00
logging
Create tool to print statistics about the file size usage of an RTC event log.
2017-04-28 14:48:17 +00:00
media
Update comments for removal of MediaController.
2017-05-08 10:21:43 +00:00
modules
iOS audio session isInterrupted flag does not get reset correctly:
2017-05-09 22:09:37 +00:00
ortc
Delete method MessageQueue::set_socketserver
2017-05-08 12:25:41 +00:00
p2p
Delete method MessageQueue::set_socketserver
2017-05-08 12:25:41 +00:00
pc
Delete method MessageQueue::set_socketserver
2017-05-08 12:25:41 +00:00
sdk
Actually move CoreVideoFrameBuffer from webrtc/common_video/ to webrtc/sdk/objc/
2017-05-08 12:32:05 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
2017-04-28 12:24:50 +00:00
test
Rename tools-webrtc -> tools_webrtc
2017-05-09 06:30:13 +00:00
tools
Moving scripts to download and build apprtc/collider.
2017-05-09 07:13:47 +00:00
video
Remove deprecated API
2017-05-08 09:59:38 +00:00
voice_engine
Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend()
2017-05-03 15:30:15 +00:00
.gitignore
…
BUILD.gn
Remove config.h from //webrtc:webrtc target.
2017-05-03 12:28:46 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
common_types.h
Delete RawVideoType enum, use the VideoType enum instead.
2017-04-28 14:18:05 +00:00
config.cc
Fix default extension id for video content type extension
2017-04-26 09:35:31 +00:00
config.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
DEPS
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +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_frame.h
Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854873003/ )
2017-05-04 09:11:12 +00:00
video_receive_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Add number of quality adapt changes to VideoSendStream stats.
2017-04-04 12:51:49 +00:00
webrtc.gni
Remove the expat and jsoncpp dependencies to reduce binary size.
2017-04-24 19:05:06 +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.