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webrtc_m130/webrtc
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Alex Loiko 04ca637be3 Make 'aleloi@' OWNER of webrtc/modules/audio_processing 
This reflects currently active developers of the module.

NOTRY=True

Bug: None
Change-Id: Ibc0810b08db753404fcb94038a4bd857d5585ef9
Reviewed-on: https://chromium-review.googlesource.com/528075
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18492}
2017-06-08 09:36:10 +00:00
..
api
Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
2017-06-02 21:37:37 +00:00
audio
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.
2017-06-01 11:02:35 +00:00
base
Deleting unused build target.
2017-06-07 11:50:13 +00:00
call
Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
2017-06-02 21:37:37 +00:00
common_audio
Moving compile_assert_c.h to webrtc/base
2017-05-15 15:06:47 +00:00
common_video
Update I420Buffer to new VideoFrameBuffer interface
2017-06-01 17:02:26 +00:00
examples
Remove webrtc deps from AppRTCMobile.
2017-06-07 11:13:48 +00:00
logging
Delete rtc_event_log/ringbuffer.h
2017-06-07 08:53:30 +00:00
media
Revert of Periodically update codec bit/frame rate settings. (patchset #8 id:140001 of https://codereview.webrtc.org/2883963002/ )
2017-06-07 13:17:49 +00:00
modules
Make 'aleloi@' OWNER of webrtc/modules/audio_processing
2017-06-08 09:36:10 +00:00
ortc
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
p2p
Fix issue with send-side bandwidth estimation over TURN TCP connections.
2017-05-27 01:40:05 +00:00
pc
Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
2017-06-02 21:37:37 +00:00
sdk
ObjC: Marshal all VideoTrackSource methods to the signaling thread
2017-06-07 14:53:41 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Prevent data race in GetStaticInstance
2017-05-29 07:01:14 +00:00
test
Revert of Periodically update codec bit/frame rate settings. (patchset #8 id:140001 of https://codereview.webrtc.org/2883963002/ )
2017-06-07 13:17:49 +00:00
tools
Revert of Refactored incoming bitrate estimator. (patchset #8 id:140001 of https://codereview.webrtc.org/2917873002/ )
2017-06-08 07:10:31 +00:00
video
Add new screenshare full stack test with limited queue.
2017-06-08 08:29:15 +00:00
voice_engine
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.
2017-06-01 11:02:35 +00:00
.gitignore
…
BUILD.gn
Delete webrtc/call.h (replaced with webrtc/call/call.h).
2017-06-08 09:21:17 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Prevent memory corruption by StreamId::Set
2017-06-02 10:37:48 +00:00
common_types.h
Delete RawVideoType enum, use the VideoType enum instead.
2017-04-28 14:18:05 +00:00
config.cc
Fix default extension id for video content type extension
2017-04-26 09:35:31 +00:00
config.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
DEPS
Delete webrtc/call.h (replaced with webrtc/call/call.h).
2017-06-08 09:21:17 +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_frame.h
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +00:00
video_receive_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Update adaptation stats to support degradations in both resolution and framerate.
2017-05-16 06:40:18 +00:00
webrtc.gni
Reland of Removes usage of native base::android::GetApplicationContext()
2017-05-26 08:51:53 +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.