admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
deadbeef 7a07f133c4 Fix TimeCallback used by BoringSSL. 
Wasn't setting microseconds properly in timeval structure.

TBR=pthatcher@webrtc.org
BUG=webrtc:6737

Review-Url: https://codereview.webrtc.org/2523533003
Cr-Commit-Position: refs/heads/master@{#15174}
2016-11-21 22:34:02 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
RTCStats operator== bugfix
2016-11-21 17:17:51 +00:00
audio
Expose RtpCodecParameters to VoiceMediaInfo stats.
2016-11-18 07:43:39 +00:00
base
Fix TimeCallback used by BoringSSL.
2016-11-21 22:34:02 +00:00
build
iOS: Cleanup buildbot JSON files + bump iOS version to 10.0
2016-11-21 08:41:35 +00:00
call
Passed AudioMixer to AudioState::Config.
2016-11-17 14:48:56 +00:00
common_audio
Revert of Move smoothing filter to common audio. (patchset #3 id:60001 of https://codereview.webrtc.org/2484153002/ )
2016-11-18 09:31:19 +00:00
common_video
Remove all references to GYP
2016-11-16 19:11:38 +00:00
examples
Revert of Allow custom metrics implementations on Android. (patchset #11 id:260001 of https://codereview.webrtc.org/2403463002/ )
2016-11-21 16:30:05 +00:00
libjingle
Remove all references to GYP
2016-11-16 19:11:38 +00:00
logging
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
media
In VoiceEngine, the settings for APM are applied in such a way that
2016-11-21 14:28:23 +00:00
modules
Remove deprecated CroppingWindowCapturer::Create
2016-11-21 21:54:30 +00:00
p2p
Remove all references to GYP
2016-11-16 19:11:38 +00:00
pc
Remove all references to GYP
2016-11-16 19:11:38 +00:00
sdk
iOS: Add AudioSendSideBwe field trial.
2016-11-17 18:29:50 +00:00
stats
RTCStats operator== bugfix
2016-11-21 17:17:51 +00:00
system_wrappers
Use NtpTime in RtcpMeasurement instead of uint sec/uint frac.
2016-11-17 10:27:20 +00:00
test
Changed the way we find the ProjectRootPath.
2016-11-18 15:06:47 +00:00
tools
Remove all references to GYP
2016-11-16 19:11:38 +00:00
video
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ )
2016-11-18 09:34:14 +00:00
voice_engine
Support multiple timestamp rates for sending DTMF.
2016-11-17 13:25:45 +00:00
.gitignore
…
BUILD.gn
Revert of Allow custom metrics implementations on Android. (patchset #11 id:260001 of https://codereview.webrtc.org/2403463002/ )
2016-11-21 16:30:05 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
…
common_types.cc
Reland of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2491613005/ )
2016-11-17 07:23:15 +00:00
common_types.h
Add overhead per packet observer to the rtp_sender.
2016-11-17 09:38:48 +00:00
config.cc
Remove Absolute Send Time from list of supported header extensions for audio streams.
2016-11-17 14:26:59 +00:00
config.h
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
Remove all references to GYP
2016-11-16 19:11:38 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_encoder.h
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ )
2016-11-18 09:34:14 +00:00
video_frame.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
video_receive_stream.h
Remove RED/RTX workaround from sender/receiver and VideoEngine2.
2016-11-11 11:28:38 +00:00
video_send_stream.h
Wire up FlexfecSender in RTP module and VideoSendStream.
2016-11-15 13:25:44 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +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.