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jbauch 7d0a77eef4 Handle case where UDP packet contains multiple DTLS records. 
Our DTLS implementation doesn't do this, but other implementations may; see https://tools.ietf.org/html/rfc6347#section-4.1.1.

BUG=chromium:537189

Review-Url: https://codereview.webrtc.org/2970883005
Cr-Commit-Position: refs/heads/master@{#18934}
2017-07-07 20:44:07 +00:00
..
api
Report interframe delay sum in old GetStats
2017-07-07 08:26:24 +00:00
audio
External APM usage downstream dependency support cleanup
2017-07-07 11:25:11 +00:00
base
Delete SignalThread class.
2017-06-29 13:21:20 +00:00
call
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_audio
base->rtc_base: Update .c, .mm and .java files.
2017-07-07 10:09:51 +00:00
common_video
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
examples
base->rtc_base: Update .c, .mm and .java files.
2017-07-07 10:09:51 +00:00
logging
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
media
External APM usage downstream dependency support cleanup
2017-07-07 11:25:11 +00:00
modules
Decreased the adaptation rate for the adaptive filter in the echo canceller 3
2017-07-07 14:59:24 +00:00
ortc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
p2p
Handle case where UDP packet contains multiple DTLS records.
2017-07-07 20:44:07 +00:00
pc
Report interframe delay sum in old GetStats
2017-07-07 08:26:24 +00:00
rtc_base
Trace loggging: Check for g_event_logger is not null before calling it.
2017-07-07 18:02:15 +00:00
rtc_tools
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
sdk
Trace loggging: Check for g_event_logger is not null before calling it.
2017-07-07 18:02:15 +00:00
stats
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
system_wrappers
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
test
base->rtc_base: Update .c, .mm and .java files.
2017-07-07 10:09:51 +00:00
video
Report interframe delay sum in old GetStats
2017-07-07 08:26:24 +00:00
voice_engine
Fix a variable naming typo
2017-07-07 12:29:47 +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
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_types.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
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
video_receive_stream.h
Report interframe delay sum in old GetStats
2017-07-07 08:26:24 +00:00
video_send_stream.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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.