platform-external-webrtc

Files

brandtr dbb1be59ff Use field_trial::IsEnabled for FlexFEC.

Also clarify some comments.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2839883002
Cr-Commit-Position: refs/heads/master@{#17875}

2017-04-26 07:02:34 +00:00

api

GN: Tighten up test target visibility + refactorings

2017-04-25 11:04:50 +00:00

audio

GN: Tighten up test target visibility + refactorings

2017-04-25 11:04:50 +00:00

base

Add SafeMin() and SafeMax(), which accept args of different types

2017-04-25 19:35:07 +00:00

call

Creating webrtc:video_stream_api

2017-04-26 06:42:15 +00:00

common_audio

Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ )

2017-04-24 19:26:27 +00:00

common_video

Add SafeMin() and SafeMax(), which accept args of different types

2017-04-25 19:35:07 +00:00

examples

Remove the expat and jsoncpp dependencies to reduce binary size.

2017-04-24 19:05:06 +00:00

logging

Creating webrtc:video_stream_api

2017-04-26 06:42:15 +00:00

media

Use field_trial::IsEnabled for FlexFEC.

2017-04-26 07:02:34 +00:00

modules

Creating webrtc:video_stream_api

2017-04-26 06:42:15 +00:00

ortc

Enabling 'gn check' on webrtc/video.

2017-04-25 07:35:35 +00:00

p2p

GN: Tighten up test target visibility + refactorings

2017-04-25 11:04:50 +00:00

Fix comment about remote restart being requested in createOffer

2017-04-25 19:57:14 +00:00

sdk

Android: Move EglBase10/EglBase14 from to API to src

2017-04-25 12:32:07 +00:00

stats

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

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

system_wrappers

Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ )

2017-04-24 19:26:27 +00:00

test

GN: Tighten up test target visibility + refactorings

2017-04-25 11:04:50 +00:00

tools

Creating webrtc:video_stream_api

2017-04-26 06:42:15 +00:00

video

Creating webrtc:video_stream_api

2017-04-26 06:42:15 +00:00

voice_engine

Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ )

2017-04-24 19:26:27 +00:00

.gitignore

…

BUILD.gn

Creating webrtc:video_stream_api

2017-04-26 06:42:15 +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

Add read support of RtpStreamId/RepairedRtpStreamId header extensions.

2017-04-19 09:59:48 +00:00

config.cc

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

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

Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it

2016-11-29 10:52:22 +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

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

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.