webrtc_m130

History

Tarun Chawla 8e857d10fd Adding capture device selection capability for video_loopback. It will help to select any capture device to test the utility. In future we can add screen share as capture device.

BUG=webrtc:7719

Change-Id: Iddc66188341c0c90e96766dff671ac3863bf3f5d
Reviewed-on: https://chromium-review.googlesource.com/517523
Commit-Queue: Peter Boström <pbos@webrtc.org>
Reviewed-by: Peter Boström <pbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18392}

2017-06-01 21:10:29 +00:00

api

Update I420Buffer to new VideoFrameBuffer interface

2017-06-01 17:02:26 +00:00

audio

New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.

2017-06-01 11:02:35 +00:00

base

Support epoll in PhysicalSocketServer.

2017-05-31 20:09:18 +00:00

call

New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.

2017-06-01 11:02:35 +00:00

common_audio

Moving compile_assert_c.h to webrtc/base

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

common_video

Update I420Buffer to new VideoFrameBuffer interface

2017-06-01 17:02:26 +00:00

examples

Update I420Buffer to new VideoFrameBuffer interface

2017-06-01 17:02:26 +00:00

logging

Print configured header extensions and codecs in rtc_event_log2text.

2017-06-01 12:23:03 +00:00

media

Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.

2017-06-01 20:22:42 +00:00

modules

Update I420Buffer to new VideoFrameBuffer interface

2017-06-01 17:02:26 +00:00

ortc

Remove VirtualSocketServer's dependency on PhysicalSocketServer.

2017-05-17 01:00:06 +00:00

p2p

Fix issue with send-side bandwidth estimation over TURN TCP connections.

2017-05-27 01:40:05 +00:00

Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.

2017-06-01 20:22:42 +00:00

sdk

Update I420Buffer to new VideoFrameBuffer interface

2017-06-01 17:02:26 +00:00

stats

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

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

system_wrappers

Prevent data race in GetStaticInstance

2017-05-29 07:01:14 +00:00

test

Adding capture device selection capability for video_loopback. It will help to select any capture device to test the utility. In future we can add screen share as capture device.

2017-06-01 21:10:29 +00:00

tools

Fix indexing error in event log analyzer.

2017-06-01 10:50:31 +00:00

video

Adding capture device selection capability for video_loopback. It will help to select any capture device to test the utility. In future we can add screen share as capture device.

2017-06-01 21:10:29 +00:00

voice_engine

New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.

2017-06-01 11:02:35 +00:00

.gitignore

Flesh out webrtc/.gitignore

2015-11-16 19:02:02 +00:00

BUILD.gn

enabling gn check on the whole WebRTC repo

2017-06-01 20:01:48 +00:00

call.h

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

codereview.settings

Give a more specific URL for creating WebRTC checkout

2016-04-01 07:36:53 +00:00

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

…

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

Reland of Removes usage of native base::android::GetApplicationContext()

2017-05-26 08:51:53 +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.