admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
solenberg fe7dd6d9ff Remove VoEAudioProcessing interface. 
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2738543002
Cr-Commit-Position: refs/heads/master@{#17185}
2017-03-11 16:10:43 +00:00
..
api
Add skeleton webrtc::SessionDescription and webrtc::MediaDescription classes.
2017-03-11 02:33:45 +00:00
audio
Move delay_based_bwe_ into CongestionController
2017-03-09 14:40:08 +00:00
base
Refactor Windows TaskQueue code to only need a single high res timer.
2017-03-10 17:33:53 +00:00
call
Revert of rewrite frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #2 id:70001 of https://codereview.webrtc.org/2745583006/ )
2017-03-10 17:49:42 +00:00
common_audio
Reland of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #1 id:1 of https://codereview.webrtc.org/2739143002/ )
2017-03-09 14:25:06 +00:00
common_video
Add FullStack test for simulcast screenshare mode.
2017-03-09 17:23:30 +00:00
examples
Revert of PyLint fixes for tools-webrtc and webrtc/tools (patchset #3 id:40001 of https://codereview.webrtc.org/2736233003/ )
2017-03-09 14:09:33 +00:00
logging
Enable GN check for webrtc/logging
2017-03-08 10:12:11 +00:00
media
Fixed invalid filtering of SCTP datachannel packets on high ports.
2017-03-11 00:20:04 +00:00
modules
Include the header <cmath>
2017-03-10 21:30:04 +00:00
ortc
Add the function ToRtpParameters.
2017-03-09 01:15:06 +00:00
p2p
Making candidate pool size behave as decided in JSEP.
2017-03-10 23:18:00 +00:00
pc
Making candidate pool size behave as decided in JSEP.
2017-03-10 23:18:00 +00:00
sdk
Reland of Add unit tests for RTCMTLVideoView. (patchset #1 id:1 of https://codereview.webrtc.org/2733953006/ )
2017-03-09 11:36:58 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Replace Clock::CurrentNtp with Clock::CurrentNtpTime
2017-03-07 10:51:09 +00:00
test
Remove VoEAudioProcessing interface.
2017-03-11 16:10:43 +00:00
tools
Move delay_based_bwe_ into CongestionController
2017-03-09 14:40:08 +00:00
video
Don't set the priority of the decoder to 'high' on Android.
2017-03-11 12:54:06 +00:00
voice_engine
Remove VoEAudioProcessing interface.
2017-03-11 16:10:43 +00:00
.gitignore
…
BUILD.gn
Remove voe_cmd_test.
2017-03-07 09:43:21 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Speculative fix for division by zero in Vp8EncoderImpl.
2017-03-10 14:08:26 +00:00
config.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
config.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +00:00
video_decoder.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
2017-03-04 03:41:59 +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.