ilnik 04f4d126f8 Implement timing frames. ...

Timing information is gathered in EncodedImage,
starting at encoders. Then it's sent using RTP header extension. In the
end, it's gathered at the GenericDecoder. Actual reporting and tests
will be in the next CLs.

BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/2911193002
Cr-Commit-Position: refs/heads/master@{#18659}

2017-06-19 14:18:55 +00:00

..

audio

Changed the interface AudioMixer::RemoveSource to have a void return type.

2016-11-18 10:03:08 +00:00

audio_codecs

Revert of Opus implementation of the AudioEncoderFactoryTemplate API (patchset #4 id:80001 of https://codereview.webrtc.org/2930243003/ )

2017-06-18 09:38:58 +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

Implement timing frames.

2017-06-19 14:18:55 +00:00

video_codecs

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +00:00

BUILD.gn

Implement timing frames.

2017-06-19 14:18:55 +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