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sprang a8ae6f2aca Add flag enabling more packets to be retransmittable. 
If not indicated otherwise, allow adding a packet to the retransmission
history at least every 1/15s in order to reduce frame dropping.

BUG=webrtc:7694

Review-Url: https://codereview.webrtc.org/2999063002
Cr-Commit-Position: refs/heads/master@{#19665}
2017-09-04 14:23:56 +00:00
..
api
Move array_view.h to webrtc/api/
2017-09-04 12:43:17 +00:00
audio
Recently we moved webrtc/base to webrtc/rtc_base, so these
2017-08-30 11:45:58 +00:00
call
Move array_view.h to webrtc/api/
2017-09-04 12:43:17 +00:00
common_audio
[MIPS]: Fix compiler error for mips32r2 dspr2
2017-08-31 12:48:24 +00:00
common_video
Delete static method TaskQueue::IsCurrent.
2017-09-04 12:18:21 +00:00
examples
Make PeerConnectionClient non-singleton.
2017-08-31 15:03:46 +00:00
logging
Move RtpExtension to api/ directory and config.h/.cc to call/.
2017-09-01 13:29:30 +00:00
media
Refactor some loops and remove double iteration.
2017-09-04 13:29:23 +00:00
modules
Add flag enabling more packets to be retransmittable.
2017-09-04 14:23:56 +00:00
ortc
Recently we moved webrtc/base to webrtc/rtc_base, so these
2017-08-30 11:45:58 +00:00
p2p
Completed the functionalities of SrtpTransport.
2017-08-31 23:00:07 +00:00
pc
Move array_view.h to webrtc/api/
2017-09-04 12:43:17 +00:00
rtc_base
Revert of Decoupling rtc_base from Obj-C code (patchset #8 id:140001 of https://codereview.webrtc.org/3001623003/ )
2017-09-04 14:19:15 +00:00
rtc_tools
Add a readme for the network tester.
2017-08-31 13:24:56 +00:00
sdk
Tightening visibility of video_toolbox_cc.
2017-09-04 12:51:34 +00:00
stats
Recently we moved webrtc/base to webrtc/rtc_base, so these
2017-08-30 11:45:58 +00:00
system_wrappers
Recently we moved webrtc/base to webrtc/rtc_base, so these
2017-08-30 11:45:58 +00:00
test
Move array_view.h to webrtc/api/
2017-09-04 12:43:17 +00:00
video
Unwrap picture ids in the RtpFrameReferencerFinder.
2017-09-04 14:03:46 +00:00
voice_engine
Move array_view.h to webrtc/api/
2017-09-04 12:43:17 +00:00
.gitignore
…
BUILD.gn
Move RtpExtension to api/ directory and config.h/.cc to call/.
2017-09-01 13:29:30 +00:00
codereview.settings
…
common_types.cc
Revert of Remove deprecated CodecType methods. (patchset #1 id:1 of https://codereview.webrtc.org/3009583002/ )
2017-08-28 14:16:43 +00:00
common_types.h
Move array_view.h to webrtc/api/
2017-09-04 12:43:17 +00:00
config.h
Move RtpExtension to api/ directory and config.h/.cc to call/.
2017-09-01 13:29:30 +00:00
DEPS
Move RtpExtension to api/ directory and config.h/.cc to call/.
2017-09-01 13:29:30 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
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
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
webrtc.gni
Removing EXPAT_RELATIVE_PATH and rtc_relative_path.
2017-08-25 15:14:04 +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.