platform-external-webrtc

Files

sprang 02569adfd4 Update screen simulcast config

Lower then bitrate so that the delta between the highest layer of the
lower stream and lowest layer of the higher stream is not too large.

BUG=webrtc:4172

This is a reland of the following CL:

Review-Url: https://codereview.webrtc.org/2791273002
Cr-Commit-Position: refs/heads/master@{#18232}
Committed: dceb42da3e

https: //codereview.webrtc.org/2883963002
Review-Url: https://codereview.webrtc.org/2966833002
Cr-Commit-Position: refs/heads/master@{#18913}

2017-07-06 12:05:50 +00:00

api

Report timing frames info in GetStats.

2017-07-06 10:06:50 +00:00

audio

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

base

Delete SignalThread class.

2017-06-29 13:21:20 +00:00

call

Reland of Add received audio/video call duration metrics based on packets.

2017-07-04 11:05:06 +00:00

common_audio

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

common_video

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

examples

Fixes build issue based on usage of Android O specific API

2017-07-04 14:02:35 +00:00

logging

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

media

Update screen simulcast config

2017-07-06 12:05:50 +00:00

modules

Implement RTP keepalive in native stack.

2017-07-06 11:38:06 +00:00

ortc

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

p2p

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

Report timing frames info in GetStats.

2017-07-06 10:06:50 +00:00

rtc_base

Reland "Update includes for webrtc/{base => rtc_base} rename (3/3)"

2017-06-30 17:45:21 +00:00

rtc_tools

fix comment length

2017-07-03 07:47:44 +00:00

sdk

Fix null ref in NetworkMonitorAutoDetect if Connectivity Manager service is unavailable

2017-07-05 16:55:09 +00:00

stats

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

system_wrappers

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

test

Report timing frames info in GetStats.

2017-07-06 10:06:50 +00:00

video

Implement RTP keepalive in native stack.

2017-07-06 11:38:06 +00:00

voice_engine

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

.gitignore

…

BUILD.gn

Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ )

2017-06-30 15:11:44 +00:00

codereview.settings

…

common_types.cc

Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"

2017-07-01 14:48:18 +00:00

common_types.h

Implement RTP keepalive in native stack.

2017-07-06 11:38:06 +00:00

config_unittest.cc

Support encrypted RTP extensions (RFC 6904)

2017-06-29 19:31:36 +00:00

config.cc

Support encrypted RTP extensions (RFC 6904)

2017-06-29 19:31:36 +00:00

config.h

Support encrypted RTP extensions (RFC 6904)

2017-06-29 19:31:36 +00:00

DEPS

Move webrtc/{tools => rtc_tools}

2017-06-30 10:04:59 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

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OWNERS

Don't add or rename files in webrtc/ and webrtc/api/ without a proper review

2017-05-08 13:01:54 +00:00

PATENTS

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PRESUBMIT.py

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README.chromium

…

typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

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

video_receive_stream.h

Report timing frames info in GetStats.

2017-07-06 10:06:50 +00:00

video_send_stream.h

Implement RTP keepalive in native stack.

2017-07-06 11:38:06 +00:00

webrtc.gni

Adds detection of audio glitches for playout on iOS (reland)

2017-06-13 16:00:18 +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.