platform-external-webrtc

Files

gyzhou ad7cad8aba An example of Unity native plugin of webrtc for Windows OS

Unity native plugin has to use Pinvoke technology in its APIs
This plugin dll can also be used by Windows C# applications other than
Unity.

BUG=webrtc:7389

Review-Url: https://codereview.webrtc.org/2823783002
Cr-Commit-Position: refs/heads/master@{#18108}

2017-05-11 23:10:03 +00:00

api

Make AudioSinkInterface::Data hold a const pointer to the audio buffer.

2017-05-11 20:35:28 +00:00

audio

Fixing video loopback test with encoder factory.

2017-05-10 11:06:11 +00:00

base

Delete left-over declaration of AdjustCurrentProcessPrivilege.

2017-05-10 13:04:45 +00:00

call

Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket.

2017-05-11 15:00:58 +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

Add support for I444 in VideoFrameBuffer

2017-05-11 12:11:57 +00:00

examples

An example of Unity native plugin of webrtc for Windows OS

2017-05-11 23:10:03 +00:00

logging

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

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

media

Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ )

2017-05-10 12:25:59 +00:00

modules

Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket.

2017-05-11 15:00:58 +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

Delete method MessageQueue::set_socketserver

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

sdk

Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ )

2017-05-10 12:25:59 +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

Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket.

2017-05-11 15:00:58 +00:00

tools

Moving scripts to download and build apprtc/collider.

2017-05-09 07:13:47 +00:00

video

Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket.

2017-05-11 15:00:58 +00:00

voice_engine

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

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

.gitignore

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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

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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

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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

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README.chromium

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typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

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

video_frame.h

Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ )

2017-05-10 12:25:59 +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.