platform-external-webrtc

Files

deadbeef d0ffa8615d Run clang-format on sigslot.h.

It's diverged enough from the original version that we no longer
benefit from keeping the old formatting.

BUG=NONE

Review-Url: https://codereview.webrtc.org/2854533002
Cr-Commit-Position: refs/heads/master@{#18129}

2017-05-13 05:52:51 +00:00

api

Reland of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2877013002/ )

2017-05-12 05:07:37 +00:00

audio

Fixing video loopback test with encoder factory.

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

base

Run clang-format on sigslot.h.

2017-05-13 05:52:51 +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

Support running AppRTC without a TURN ICE server

2017-05-12 07:40:58 +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

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

WebRtcVideoEncoderFactory cleanup

2017-05-12 07:24:40 +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

…

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

…

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.