admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
eladalon 0860bee402 Remove deprecated DirectTransport constructors 
Remove deprecated DirectTransport constructors, now that they're no longer used in internal projects.

BUG=webrtc:8125
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/3006513002
Cr-Commit-Position: refs/heads/master@{#19496}
2017-08-24 13:56:15 +00:00
..
api
Report max interframe delay over window insdead of interframe delay sum
2017-08-23 12:24:10 +00:00
audio
Replacing NetEq discard rate with secondary discarded rate.
2017-08-24 13:46:52 +00:00
call
New accessor function VideoReceiveStream::Config::Rtp::AddRtxBinding
2017-08-24 12:47:16 +00:00
common_audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
common_video
Change logging method from WEBRTC_TRACE macro to LOG macro.
2017-08-22 07:29:13 +00:00
examples
Rename all objc targets to be suffixed for consistency
2017-08-24 12:43:45 +00:00
logging
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +00:00
media
Increase logging severity level for SW fallback.
2017-08-24 12:19:57 +00:00
modules
Replacing NetEq discard rate with secondary discarded rate.
2017-08-24 13:46:52 +00:00
ortc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
p2p
Renamed dtlstransportchannel.h/.cc/_unittest.cc.
2017-08-24 01:05:50 +00:00
pc
Delete unneeded include of videocapturer.h
2017-08-24 07:29:04 +00:00
rtc_base
Fix rtc::CurrentThreadId() on Android
2017-08-24 13:41:05 +00:00
rtc_tools
Move PacedSender ownership to RtpTransportControllerSend.
2017-08-22 14:16:49 +00:00
sdk
Fix bugs in HardwareVideoDecoder reinitialization.
2017-08-24 13:03:59 +00:00
stats
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
system_wrappers
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
test
Remove deprecated DirectTransport constructors
2017-08-24 13:56:15 +00:00
video
Make CodecType conversion functions non-optional.
2017-08-24 10:52:48 +00:00
voice_engine
Delete unneeded includes of atomic32.h.
2017-08-24 10:30:05 +00:00
.gitignore
…
BUILD.gn
Rename all objc targets to be suffixed for consistency
2017-08-24 12:43:45 +00:00
codereview.settings
…
common_types.cc
Make CodecType conversion functions non-optional.
2017-08-24 10:52:48 +00:00
common_types.h
Replacing NetEq discard rate with secondary discarded rate.
2017-08-24 13:46:52 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
config.h
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
DEPS
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +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
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +00:00
video_send_stream.h
Move video send/receive stream headers to webrtc/call.
2017-08-22 12:43:23 +00:00
webrtc.gni
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +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.