platform-external-webrtc

Files

magjed d5c77abbaa Revert of Android: Move CameraStatistics from API to src (patchset #1 id:1 of https://codereview.webrtc.org/2821373003/ )

Reason for revert:
Breaks external clients.

Original issue's description:
> Android: Move CameraStatistics from API to src
>
> BUG=webrtc:7172
>
> Review-Url: https://codereview.webrtc.org/2821373003
> Cr-Commit-Position: refs/heads/master@{#17765}
> Committed: 800daef50a

TBR=sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7172

Review-Url: https://codereview.webrtc.org/2825383002
Cr-Commit-Position: refs/heads/master@{#17767}

2017-04-19 15:45:45 +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

Enable GN check for webrtc/base

2017-04-19 15:37:36 +00:00

call

Delete VieRemb class, move functionality to PacketRouter.

2017-04-19 06:38:35 +00:00

common_audio

Enable GN check for webrtc/base

2017-04-19 15:37:36 +00:00

common_video

Delete unneeded includes of deprecated system_wrappers include files.

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

examples

Add error codes for Android audio recording start errors.

2017-04-18 17:30:29 +00:00

logging

Delay based logging.

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

media

Relanding: Remove rtc_p2p_unittests from ortc_unittests and rtc_media_unittests

2017-04-18 22:49:09 +00:00

modules

Enable GN check for webrtc/base

2017-04-19 15:37:36 +00:00

ortc

Relanding: Remove rtc_p2p_unittests from ortc_unittests and rtc_media_unittests

2017-04-18 22:49:09 +00:00

p2p

Relanding: Remove rtc_p2p_unittests from ortc_unittests and rtc_media_unittests

2017-04-18 22:49:09 +00:00

Test CreatePeerConnectionFactory() with a forwarding mock AudioDecoderFactory

2017-04-19 10:47:57 +00:00

sdk

Revert of Android: Move CameraStatistics from API to src (patchset #1 id:1 of https://codereview.webrtc.org/2821373003/ )

2017-04-19 15:45:45 +00:00

stats

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

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

system_wrappers

Enable GN check for webrtc/base

2017-04-19 15:37:36 +00:00

test

Enable GN check for webrtc/base

2017-04-19 15:37:36 +00:00

tools

Event log visualizer TimeSeries is now created on the stack and then moved into the vector of series.

2017-04-19 12:58:51 +00:00

video

Do not report cpu limited resolution stats when degradation preference is disabled and no scaling is done.

2017-04-19 09:01:06 +00:00

voice_engine

Enable GN check for webrtc/base

2017-04-19 15:37:36 +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

Add read support of RtpStreamId/RepairedRtpStreamId header extensions.

2017-04-19 09:59:48 +00:00

common_types.h

Add read support of RtpStreamId/RepairedRtpStreamId header extensions.

2017-04-19 09:59:48 +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

Delete VieRemb class, move functionality to PacketRouter.

2017-04-19 06:38:35 +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.