platform-external-webrtc

Files

magjed eaf4a1e103 Add separate base classes for I420 and I444 buffers

Previously, the base class PlanarYuvBuffer was used directly. Having
separate base classes will allow us to improve type safety in some
places.

BUG=webrtc:7632
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2914463002
Cr-Commit-Position: refs/heads/master@{#18317}

2017-05-30 08:21:59 +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

Store/restore RTP state for audio streams with same SSRC within a call

2017-05-23 13:07:11 +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

Revert of Split iOS sdk in to separate targets (patchset #3 id:320001 of https://codereview.webrtc.org/2893843003/ )

2017-05-29 16:57:37 +00:00

logging

Remove final from RtcEventLogNullImpl

2017-05-29 09:46:05 +00:00

media

Avoid toggling default receive streams in WebRtcVideoChannel2.

2017-05-30 06:33:31 +00:00

modules

Remove RTPSenderVideo::FecPacketOverhead() since it is unused.

2017-05-30 06:34:37 +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

Revert of Split iOS sdk in to separate targets (patchset #3 id:320001 of https://codereview.webrtc.org/2893843003/ )

2017-05-29 16:57:37 +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

Revert of Split iOS sdk in to separate targets (patchset #3 id:320001 of https://codereview.webrtc.org/2893843003/ )

2017-05-29 16:57:37 +00:00

tools

Revert of Remove unecessary non fatal error statement that very often is printed in the PSNR or SSIM metric n… (patchset #1 id:1 of https://codereview.webrtc.org/2901793002/ )

2017-05-23 15:31:11 +00:00

video

Delete RtpData::OnRecoveredPacket, use RecoveredPacketReceiver instead.

2017-05-29 15:16:37 +00:00

voice_engine

Delete RtpData::OnRecoveredPacket, use RecoveredPacketReceiver instead.

2017-05-29 15:16:37 +00:00

.gitignore

…

BUILD.gn

Revert of Split iOS sdk in to separate targets (patchset #3 id:320001 of https://codereview.webrtc.org/2893843003/ )

2017-05-29 16:57:37 +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

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

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PRESUBMIT.py

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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.