webrtc_m130

History

nisse c4a3173db0 Delete unused features of AsyncInvoke.

Also eliminates its dependency on callback.h.

BUG=None

Review-Url: https://codereview.webrtc.org/2871403003
Cr-Commit-Position: refs/heads/master@{#18163}

2017-05-16 12:51:29 +00:00

api

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +00:00

audio

New class RtpDemuxer and RtpPacketSinkInterface, use in Call.

2017-05-16 11:47:04 +00:00

base

Delete unused features of AsyncInvoke.

2017-05-16 12:51:29 +00:00

call

New class RtpDemuxer and RtpPacketSinkInterface, use in Call.

2017-05-16 11:47:04 +00:00

common_audio

Moving compile_assert_c.h to webrtc/base

2017-05-15 15:06:47 +00:00

common_video

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +00:00

examples

Update testing tools (AppRTC, Go) to new versions

2017-05-16 12:36:15 +00:00

logging

DCHECK that we don't insert nullptr into event log.

2017-05-11 13:05:27 +00:00

media

WebRtcVideoEncoderFactory cleanup

2017-05-12 07:24:40 +00:00

modules

New class RtpDemuxer and RtpPacketSinkInterface, use in Call.

2017-05-16 11:47:04 +00:00

ortc

Delete method MessageQueue::set_socketserver

2017-05-08 12:25:41 +00:00

p2p

Get tests working on systems that only support IPv6.

2017-05-16 02:43:33 +00:00

Get tests working on systems that only support IPv6.

2017-05-16 02:43:33 +00:00

sdk

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +00:00

stats

Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

2017-03-01 09:02:45 +00:00

system_wrappers

Adding target to track asm_defines.h

2017-05-15 21:59:17 +00:00

test

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +00:00

tools

Update testing tools (AppRTC, Go) to new versions

2017-05-16 12:36:15 +00:00

video

New class RtpDemuxer and RtpPacketSinkInterface, use in Call.

2017-05-16 11:47:04 +00:00

voice_engine

Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend()

2017-05-03 15:30:15 +00:00

.gitignore

…

BUILD.gn

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +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

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +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

…

README.chromium

…

typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

2017-04-06 11:32:27 +00:00

video_frame.h

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +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

Update adaptation stats to support degradations in both resolution and framerate.

2017-05-16 06:40:18 +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.