platform-external-webrtc

Files

stefan 27925de951 Enable trendline experiment and bayesian bitrate estimator experiment by default.

BUG=webrtc:6566, webrtc:7415

Review-Url: https://codereview.webrtc.org/2777333003
Cr-Commit-Position: refs/heads/master@{#17491}

2017-03-31 14:38:06 +00:00

api

Rewrite PeerConnection integration tests using better testing practices.

2017-03-30 04:08:16 +00:00

audio

Let PacketRouter separate send and receive modules.

2017-03-31 12:44:52 +00:00

base

Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )

2017-03-31 10:08:07 +00:00

call

Enable trendline experiment and bayesian bitrate estimator experiment by default.

2017-03-31 14:38:06 +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

Delete unneeded includes of deprecated system_wrappers include files.

2017-03-30 07:31:15 +00:00

examples

Enable trendline experiment and bayesian bitrate estimator experiment by default.

2017-03-31 14:38:06 +00:00

logging

Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )

2017-03-31 10:08:07 +00:00

media

Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )

2017-03-30 17:44:38 +00:00

modules

Enable trendline experiment and bayesian bitrate estimator experiment by default.

2017-03-31 14:38:06 +00:00

ortc

Disable the ORTC integration tests on TSan.

2017-03-28 20:45:18 +00:00

p2p

Disable possibly flaky AudioCodingModuleTest.TestPacketLossStereo for Linux.

2017-03-28 10:00:07 +00:00

Rewrite PeerConnection integration tests using better testing practices.

2017-03-30 04:08:16 +00:00

sdk

Enable trendline experiment and bayesian bitrate estimator experiment by default.

2017-03-31 14:38:06 +00:00

stats

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

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

system_wrappers

Remove ALL usage of CriticalSectionWrapper.

2017-03-31 09:03:55 +00:00

test

Let PacketRouter separate send and receive modules.

2017-03-31 12:44:52 +00:00

tools

Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )

2017-03-31 10:08:07 +00:00

video

Let PacketRouter separate send and receive modules.

2017-03-31 12:44:52 +00:00

voice_engine

Let PacketRouter separate send and receive modules.

2017-03-31 12:44:52 +00:00

.gitignore

…

BUILD.gn

Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )

2017-03-31 10:08:07 +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

…

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 #8 id:410001 of https://codereview.webrtc.org/2781433002/ )

2017-03-30 17:44:38 +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.