platform-external-webrtc

Files

elad.alon 4e7645118e Fix UT failure by temporarily uncommenting

BUG=webrtc:7322, webrtc:7405

Review-Url: https://codereview.webrtc.org/2780473002
Cr-Commit-Position: refs/heads/master@{#17393}

2017-03-27 15:53:11 +00:00

api

Make lint errors fatal in presubmit and fix files in whitelisted paths

2017-03-21 14:52:41 +00:00

audio

Fix UT failure by temporarily uncommenting

2017-03-27 15:53:11 +00:00

base

Delete unused Pathname methods.

2017-03-27 11:37:14 +00:00

call

Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType"

2017-03-27 14:15:49 +00:00

common_audio

iSAC fix entropy coder: Recently added DCHECK could in fact trigger

2017-03-13 12:28:47 +00:00

common_video

Fix cpplint errors in locations that are already being checked

2017-03-24 10:22:49 +00:00

examples

Roll chromium_revision 4f16f0c98f..a896ff44a3 (459568:459701)

2017-03-27 06:28:27 +00:00

logging

Delete support for receiving RTCP RPSI and SLI message.

2017-03-22 14:15:09 +00:00

media

Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType"

2017-03-27 14:15:49 +00:00

modules

remove more CriticalSectionWrappers.

2017-03-27 14:24:57 +00:00

ortc

Add the function ToRtpParameters.

2017-03-09 01:15:06 +00:00

p2p

Revert of Removing HTTPS and SOCKS proxy server code. (patchset #2 id:20001 of https://codereview.webrtc.org/2731673002/ )

2017-03-23 22:45:49 +00:00

Pass ownership of candidate to PeerConnection::OnIceCandidate

2017-03-25 15:31:12 +00:00

sdk

Add field trial to update quality scaler QP thresholds for Android HW encoder.

2017-03-23 21:40:08 +00:00

stats

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

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

system_wrappers

remove more CriticalSectionWrappers.

2017-03-27 14:24:57 +00:00

test

remove more CriticalSectionWrappers.

2017-03-27 14:24:57 +00:00

tools

Attach TransportFeedbackPacketLossTracker to ANA (PLR only)

2017-03-23 18:04:48 +00:00

video

Define RtpTransportControllerSendInterface.

2017-03-27 12:36:15 +00:00

voice_engine

Fix UT failure by temporarily uncommenting

2017-03-27 15:53:11 +00:00

.gitignore

…

BUILD.gn

Enable complete_static_lib on Mac and iOS

2017-03-23 09:49:15 +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

Reject XR TargetBitrate items with unsupported layer indices

2016-12-06 14:09:00 +00:00

common_types.h

Delete unused constant kConfigParameterSize.

2017-03-22 15:19:07 +00:00

config.cc

Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

2017-02-26 02:15:09 +00:00

config.h

Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

2017-02-26 02:15:09 +00:00

DEPS

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +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

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PRESUBMIT.py

…

README.chromium

…

typedefs.h

Use RTC_UNUSED instead of conditional compilation in BWE simulator tool.

2017-03-17 13:34:47 +00:00

video_decoder.h

Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )

2017-02-09 12:53:45 +00:00

video_encoder.h

Move usage of QualityScaler to ViEEncoder.

2016-11-29 09:44:22 +00:00

video_frame.h

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

video_receive_stream.h

Add |protected_by_flexfec| flag to VideoReceiveStream::Config.

2017-03-09 14:13:20 +00:00

video_send_stream.h

Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )

2017-03-21 20:26:06 +00:00

webrtc.gni

Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )

2017-03-04 03:41:59 +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.