admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
mbonadei 63f968e0ce Removing WEBRTC_BUILD_LIBEVENT macro definition 
This seems to be unused in WebRTC and a search in cs.chromium.orm shows
that it is never used in chromium too.

Interesting CL about this macro: https://codereview.webrtc.org/2153563003

BUG=None

Review-Url: https://codereview.webrtc.org/3006243002
Cr-Commit-Position: refs/heads/master@{#19747}
2017-09-08 13:20:53 +00:00
..
api
Change RtpSender to have multiple stream_ids
2017-09-07 17:03:24 +00:00
audio
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
call
Make CreateRtcLogStreamConfig() return a std::unique_ptr
2017-09-08 13:15:52 +00:00
common_audio
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
common_video
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
examples
Make PeerConnectionClient non-singleton.
2017-08-31 15:03:46 +00:00
logging
Remove RtcEventLogImpl forward declaration from rtc_event_log.h
2017-09-07 15:02:35 +00:00
media
Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ )
2017-09-08 12:51:54 +00:00
modules
Revert of Delete Rtx-related methods from RTPPayloadRegistry. (patchset #3 id:40001 of https://codereview.webrtc.org/3006993002/ )
2017-09-08 12:00:54 +00:00
ortc
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
p2p
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
pc
In PC integration tests, create tracks/streams with random IDs.
2017-09-07 21:12:05 +00:00
rtc_base
Removing WEBRTC_BUILD_LIBEVENT macro definition
2017-09-08 13:20:53 +00:00
rtc_tools
Add visualization of late feedback packets, and avoid plotting incorrect delays computed based on those.
2017-09-06 09:06:32 +00:00
sdk
Remove static/default interface method usage.
2017-09-08 11:46:33 +00:00
stats
Recently we moved webrtc/base to webrtc/rtc_base, so these
2017-08-30 11:45:58 +00:00
system_wrappers
Revert of Rename thread annotation macros to have RTC prefix for syncrhonization primitives. (patchset #1 id:1 of https://codereview.webrtc.org/3004393002/ )
2017-09-06 08:38:35 +00:00
test
Keep track of the capacity delay error in the FakeNetworkPipe.
2017-09-07 16:08:50 +00:00
video
Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ )
2017-09-08 12:51:54 +00:00
voice_engine
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
.gitignore
…
BUILD.gn
Add VideoProcessorIntegrationTest for MediaCodec implementations.
2017-09-07 14:50:06 +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 optional.h to webrtc/api/
2017-09-05 15:43:13 +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
Revert of Add //build/android/pylib/results/ to Android isolate packaging (patchset #2 id:40001 of https://codereview.webrtc.org/3011023002/ )
2017-09-06 19:11:27 +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.