webrtc_m130

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

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

Implement operator<< for AudioCodecInfo and AudioCodecSpec

2017-06-14 10:27:40 +00:00

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

Injectable audio encoders: BuiltinAudioEncoderFactory

2017-04-06 17:03:21 +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

Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ )

2017-05-10 12:25:59 +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

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

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