admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc/api
History
Emad Omara dab1d2d34e Enable SNI in ssl adapter. 
Bug: webrtc:6973
Change-Id: I13d28cf41c586880bd7fea523005233921794cdf
Reviewed-on: https://chromium-review.googlesource.com/523024
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Justin Uberti <juberti@chromium.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Emad Omara <emadomara@google.com>
Cr-Commit-Position: refs/heads/master@{#18640}
2017-06-16 23:30:48 +00:00
..
audio
Changed the interface AudioMixer::RemoveSource to have a void return type.
2016-11-18 10:03:08 +00:00
audio_codecs
Templated AudioEncoderFactory
2017-06-16 17:42:05 +00:00
call
Reland of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2877013002/ )
2017-05-12 05:07:37 +00:00
ortc
Make PacketTransportInternal inherit from PacketTransportInterface.
2017-04-07 04:47:33 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
test
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
2017-04-10 14:39:05 +00:00
video
Update I420Buffer to new VideoFrameBuffer interface
2017-06-01 17:02:26 +00:00
video_codecs
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +00:00
BUILD.gn
Add support for I444 in VideoFrameBuffer
2017-05-11 12:11:57 +00:00
datachannel.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
datachannelinterface.h
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
DEPS
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
dtmfsenderinterface.h
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
fakemetricsobserver.cc
Add UMA metrics for ICE regathering reasons.
2016-10-05 18:47:39 +00:00
fakemetricsobserver.h
Enable cpplint and fix cpplint errors in webrtc/api
2017-02-01 09:55:59 +00:00
jsep.h
Replace NULL with nullptr or null in webrtc/api/.
2017-02-27 22:47:33 +00:00
jsepicecandidate.h
Replace NULL with nullptr or null in webrtc/api/.
2017-02-27 22:47:33 +00:00
jsepsessiondescription.h
Replace NULL with nullptr or null in webrtc/api/.
2017-02-27 22:47:33 +00:00
mediaconstraintsinterface.cc
Enable cpplint and fix cpplint errors in webrtc/api
2017-02-01 09:55:59 +00:00
mediaconstraintsinterface.h
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
mediastream.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreaminterface.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreaminterface.h
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
mediastreamproxy.h
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
mediastreamtrack.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreamtrackproxy.h
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
mediatypes.cc
Fixing "control reaches end of non-void function" compile warning.
2017-03-01 02:30:35 +00:00
mediatypes.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
notifier.h
Replace NULL with nullptr or null in webrtc/api/.
2017-02-27 22:47:33 +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
peerconnectionfactoryproxy.h
Switching some interfaces to use std::unique_ptr<>.
2017-02-11 04:13:37 +00:00
peerconnectioninterface.h
Enable SNI in ssl adapter.
2017-06-16 23:30:48 +00:00
peerconnectionproxy.h
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
proxy.h
Replace NULL with nullptr or null in webrtc/api/.
2017-02-27 22:47:33 +00:00
rtcerror_unittest.cc
Move rtc_api_unittests into rtc_unittests.
2017-02-24 09:17:43 +00:00
rtcerror.cc
Adding RTCErrorOr class to be used by ORTC APIs.
2017-02-17 07:31:33 +00:00
rtcerror.h
Make lint errors fatal in presubmit and fix files in whitelisted paths
2017-03-21 14:52:41 +00:00
rtpparameters.h
Create the SrtpTransportInterface.
2017-03-03 22:39:06 +00:00
rtpreceiverinterface.h
Modified the rtp_receiver_unittests.
2017-04-11 18:28:10 +00:00
rtpsender.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
rtpsenderinterface.h
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
statstypes.cc
Replace NULL with nullptr or null in webrtc/api/.
2017-02-27 22:47:33 +00:00
statstypes.h
Replace NULL with nullptr or null in webrtc/api/.
2017-02-27 22:47:33 +00:00
streamcollection.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
umametrics.h
Enable cpplint and fix cpplint errors in webrtc/api
2017-02-01 09:55:59 +00:00
videosourceproxy.h
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
videotracksource.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
webrtcsdp.h
Added api/webrtcsdp.h forwarding header to work around upstream projects.
2017-01-24 15:13:59 +00:00