platform-external-webrtc

Files

mbonadei 8a24f47bad Revert of Enabling 'gn check' on //webrtc/test. (patchset #10 id:180001 of https://codereview.webrtc.org/2796363003/ )

Reason for revert:
It is breaking a downstream bot.

Original issue's description:
> Enabling 'gn check' on //webrtc/test.
>
> BUG=webrtc:6828
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2796363003
> Cr-Commit-Position: refs/heads/master@{#17689}
> Committed: b41445858e

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828

Review-Url: https://codereview.webrtc.org/2817003002
Cr-Commit-Position: refs/heads/master@{#17690}

2017-04-13 09:13:09 +00:00

api

Modified the rtp_receiver_unittests.

2017-04-11 18:28:10 +00:00

audio

Fix lint errors to enable stricter PyLint rules

2017-04-12 19:06:13 +00:00

base

Revert of Android Logging.java: Load native library only when needed (patchset #2 id:2 of https://codereview.webrtc.org/2817593003/ )

2017-04-12 17:13:21 +00:00

call

Making FakeNetworkPipe demux audio and video packets.

2017-04-10 23:57:57 +00:00

common_audio

Rename COMPILE_ASSERT macro to RTC_COMPILE_ASSERT

2017-04-11 06:21:43 +00:00

common_video

Delete unneeded includes of deprecated system_wrappers include files.

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

examples

Address denicija's comments for AppRTCMobile video codec setting.

2017-04-11 12:36:43 +00:00

logging

Delay based logging.

2017-04-11 08:50:23 +00:00

media

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

modules

RTC_DCHECK() is used in place of assert().

2017-04-13 06:07:19 +00:00

ortc

Enabling 'gn check' on //webrtc/ortc.

2017-04-11 09:52:49 +00:00

p2p

Fix compilation issues of std::unique_ptr

2017-04-09 16:09:06 +00:00

Make RtpTransport actually implement RtpTransportInterface

2017-04-10 23:17:57 +00:00

sdk

Added support for changing the volume of AudioTrack as discussed in BUG=webrtc:6533

2017-04-12 23:58:48 +00:00

stats

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

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

system_wrappers

Rename COMPILE_ASSERT macro to RTC_COMPILE_ASSERT

2017-04-11 06:21:43 +00:00

test

Revert of Enabling 'gn check' on //webrtc/test. (patchset #10 id:180001 of https://codereview.webrtc.org/2796363003/ )

2017-04-13 09:13:09 +00:00

tools

Reland of land "Add first part of the network_tester functionality" (patchset #1 id:1 of https://codereview.webrtc.org/2813193002/ )

2017-04-13 08:24:03 +00:00

video

Fix lint errors to enable stricter PyLint rules

2017-04-12 19:06:13 +00:00

voice_engine

Removing unnecessary parameters from initializeAndroidGlobals.

2017-04-10 22:08:02 +00:00

.gitignore

…

BUILD.gn

Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ )

2017-04-07 07:59:12 +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

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

common_types.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

config.cc

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

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

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

Make WARN_UNUSED_RESULT a no-op on gcc

2017-04-06 11:32:27 +00:00

video_decoder.h

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

video_encoder.h

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

video_frame.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

video_receive_stream.h

Add |protected_by_flexfec| flag to VideoReceiveStream::Config.

2017-03-09 14:13:20 +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

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.