platform-external-webrtc

Files

brandtr 11fb472ae4 Recreate FlexfecReceiveStream separately from VideoReceiveStream.

After this CL, reconfiguring the FlexFEC payload type at the
WebRtcVideoChannel2 level will no longer lead to the recreation of
the VideoReceiveStream. This means that the jitter buffer will
not be destroyed and a smoother video playback is achieved during
SDP renegotiation.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2911913002
Cr-Commit-Position: refs/heads/master@{#18318}

2017-05-30 08:31:37 +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

Recreate FlexfecReceiveStream separately from VideoReceiveStream.

2017-05-30 08:31:37 +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

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

…

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.