webrtc_m130

History

minyue 678d2ee651 Cosmetic change in ANA: renaming characteristic points for consistency.

BUG=none

Review-Url: https://codereview.webrtc.org/2907153004
Cr-Commit-Position: refs/heads/master@{#18329}

2017-05-30 14:36:20 +00:00

api

Add separate base classes for I420 and I444 buffers

2017-05-30 08:21:59 +00:00

audio

Rename elad.alon to eladalon, to avoid confusion between repositories.

2017-05-25 07:15:35 +00:00

base

Disable SIGPIPE for sockets created on iOS.

2017-05-26 23:27:09 +00:00

call

This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport.

2017-05-30 10:52:10 +00:00

common_audio

Moving compile_assert_c.h to webrtc/base

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

common_video

Add separate base classes for I420 and I444 buffers

2017-05-30 08:21:59 +00:00

examples

Reland of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2911053002/ )

2017-05-30 08:48:47 +00:00

logging

This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport.

2017-05-30 10:52:10 +00:00

media

Recreate FlexfecReceiveStream separately from VideoReceiveStream.

2017-05-30 08:31:37 +00:00

modules

Cosmetic change in ANA: renaming characteristic points for consistency.

2017-05-30 14:36:20 +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

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

2017-05-26 08:51:53 +00:00

sdk

Fixing absolute path that was breaking chromium

2017-05-30 13:23:41 +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

Fixing absolute path that was breaking chromium

2017-05-30 13:23:41 +00:00

tools

This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport.

2017-05-30 10:52:10 +00:00

video

Persist RTP state for FlexFEC.

2017-05-30 09:32:12 +00:00

voice_engine

This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport.

2017-05-30 10:52:10 +00:00

.gitignore

…

BUILD.gn

Reland of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2911053002/ )

2017-05-30 08:48:47 +00:00

call.h

…

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

…

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.