platform-external-webrtc

Files

deadbeef 0687829794 Fixing SignalSentPacket for TCP connections.

The signal was only being hooked up for incoming connections, not
outgoing connections.

As a result, the bandwidth estimator didn't know when packets were sent
and couldn't calculate delays.

BUG=webrtc:7509

Review-Url: https://codereview.webrtc.org/2834083002
Cr-Commit-Position: refs/heads/master@{#17817}

2017-04-21 21:22:23 +00:00

api

Enable GN check for webrtc/{p2p,system_wrappers}

2017-04-21 12:17:08 +00:00

audio

Fix lint errors to enable stricter PyLint rules

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

base

Negotiate the same SRTP crypto suites for every DTLS association formed.

2017-04-21 10:23:33 +00:00

call

Replace first_packet_sent_ms_ in Call.

2017-04-20 06:28:53 +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

Removed duplicated annotation in webrtc/examples/androidtests/AndroidManifest.xml

2017-04-20 12:31:32 +00:00

logging

Delay based logging.

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

media

Enable GN check for webrtc/{p2p,system_wrappers}

2017-04-21 12:17:08 +00:00

modules

Don't call unconfigureWebRTCSession if configureWebRTCSession fails.

2017-04-21 20:56:39 +00:00

ortc

Relanding: Remove rtc_p2p_unittests from ortc_unittests and rtc_media_unittests

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

p2p

Fixing SignalSentPacket for TCP connections.

2017-04-21 21:22:23 +00:00

Negotiate the same SRTP crypto suites for every DTLS association formed.

2017-04-21 10:23:33 +00:00

sdk

Android: Prepare moving EglBase10/EglBase14 from to API to src

2017-04-21 08:34:12 +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/{p2p,system_wrappers}

2017-04-21 12:17:08 +00:00

test

Revert of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #3 id:40001 of https://codereview.webrtc.org/2781713004/ )

2017-04-20 10:54:22 +00:00

tools

Add network tester client [android] to be able to test mobile networks in terms of packet size and sending rates.

2017-04-20 13:56:27 +00:00

video

Only increment scale counter and adaptation stats (and store last_adaptation_request_) if sink_wants_ is updated.

2017-04-21 08:47:31 +00:00

voice_engine

Fix invalid output buffer usage

2017-04-20 22:19:10 +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

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PRESUBMIT.py

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README.chromium

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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

Revert of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #3 id:40001 of https://codereview.webrtc.org/2781713004/ )

2017-04-20 10:54:22 +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.