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Magnus Jedvert 9932e255ea ObjC: Marshal all VideoTrackSource methods to the signaling thread 
This CL makes sure the real VideoTrackSourceInterface implementation is
destroyed on the signaling thread and marshals all method calls to the
signaling thread. This is done using VideoTrackSourceProxy.

Bug: webrtc:7767
Change-Id: Iba3b67bb32a684ba289bc8b9981585ea58084359
Reviewed-on: https://chromium-review.googlesource.com/526634
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18476}
2017-06-07 14:53:41 +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
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
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
Change all numerical string inputs to int and remove unused stderr
2017-06-02 09:16:27 +00:00
video
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
voice_engine
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.
2017-06-01 11:02:35 +00:00
.gitignore
…
BUILD.gn
enabling gn check on the whole WebRTC repo
2017-06-01 20:01:48 +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
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
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +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.