admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
zstein e8ab543589 Make BaseChannel::rtp_transport_ a unique_ptr. 
This will eventually be a unique_ptr<RtpTransportInternal> so that we can choose to use an RtpTransport or SrtpTransport.

BUG=None

Review-Url: https://codereview.webrtc.org/2974903003
Cr-Commit-Position: refs/heads/master@{#18987}
2017-07-12 18:48:11 +00:00
..
api
API for periodically regathering ICE candidates
2017-07-11 21:49:38 +00:00
audio
Use relative paths in GN files.
2017-07-11 13:20:45 +00:00
base
Revert of Delete SignalThread class. (patchset #20 id:380001 of https://codereview.webrtc.org/2915253002/ )
2017-07-11 23:56:05 +00:00
call
Use relative paths in GN files.
2017-07-11 13:20:45 +00:00
common_audio
Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
2017-07-10 12:58:42 +00:00
common_video
Use relative paths in GN files.
2017-07-11 13:20:45 +00:00
examples
Revert of Delete SignalThread class. (patchset #20 id:380001 of https://codereview.webrtc.org/2915253002/ )
2017-07-11 23:56:05 +00:00
logging
Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
2017-07-10 12:58:42 +00:00
media
Use relative paths in GN files.
2017-07-11 13:20:45 +00:00
modules
Reland "Adding ANA config event to debug dump."
2017-07-12 12:51:37 +00:00
ortc
Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
2017-07-10 12:58:42 +00:00
p2p
API for periodically regathering ICE candidates
2017-07-11 21:49:38 +00:00
pc
Make BaseChannel::rtp_transport_ a unique_ptr.
2017-07-12 18:48:11 +00:00
rtc_base
Reland of Make "set_ignore_non_default_routes" actually use its argument. (patchset #1 id:1 of https://codereview.webrtc.org/2974193002/ )
2017-07-12 03:07:37 +00:00
rtc_tools
Use relative paths in GN files.
2017-07-11 13:20:45 +00:00
sdk
Revert of Injectable Obj-C video codecs (patchset #8 id:140001 of https://codereview.webrtc.org/2966023002/ )
2017-07-11 22:46:31 +00:00
stats
Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
2017-07-10 12:58:42 +00:00
system_wrappers
Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
2017-07-10 12:58:42 +00:00
test
Use relative paths in GN files.
2017-07-11 13:20:45 +00:00
video
Let alr detector use a budged to detect underuse.
2017-07-11 13:56:04 +00:00
voice_engine
TransmitMixer: Check GetSendCodec return value.
2017-07-11 15:19:31 +00:00
.gitignore
…
BUILD.gn
Use relative paths in GN files.
2017-07-11 13:20:45 +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
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
video_receive_stream.h
Report interframe delay sum in old GetStats
2017-07-07 08:26:24 +00:00
video_send_stream.h
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
2017-07-11 10:44:17 +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.