kwiberg 327af33151 iSAC fixed-point implementation of the Audio{En,De}coderFactoryTemplate APIs ...

BUG=webrtc:7835, webrtc:7841

Review-Url: https://codereview.webrtc.org/2996693002
Cr-Commit-Position: refs/heads/master@{#19409}

2017-08-18 11:09:40 +00:00

..

audio

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

audio_codecs

iSAC fixed-point implementation of the Audio{En,De}coderFactoryTemplate APIs

2017-08-18 11:09:40 +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

Wire up RTP keep-alive in ortc api.

2017-08-09 13:42:32 +00:00

stats

Audit of kConstants missing the const qualifier

2017-08-08 17:48:15 +00:00

test

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

video

Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )

2017-08-18 09:51:12 +00:00

video_codecs

Move kMinPixelsPerFrame constant in VideoStreamEncoder to VideoEncoder::ScalingSettings.

2017-08-17 15:58:54 +00:00

BUILD.gn

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

datachannel.h

Create //webrtc/api:libjingle_peerconnection_api + refactorings.

2017-01-23 12:56:25 +00:00

datachannelinterface.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

DEPS

Opus implementation of the AudioEncoderFactoryTemplate API

2017-06-30 11:23:22 +00:00

dtmfsenderinterface.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

fakemetricsobserver.cc

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

fakemetricsobserver.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

jsep.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

jsepicecandidate.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

jsepsessiondescription.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

mediaconstraintsinterface.cc

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

mediaconstraintsinterface.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +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 Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt

2017-08-01 06:22:01 +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

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

mediatypes.h

Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

2017-02-26 02:15:09 +00:00

notifier.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +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

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

peerconnectioninterface.h

Add "max_ipv6_networks" field to RTCConfiguration.

2017-07-26 23:50:11 +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

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

rtcerror_unittest.cc

Move rtc_api_unittests into rtc_unittests.

2017-02-24 09:17:43 +00:00

rtcerror.cc

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

rtcerror.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

rtpparameters.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

rtpreceiverinterface.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

rtpsender.h

Create //webrtc/api:libjingle_peerconnection_api + refactorings.

2017-01-23 12:56:25 +00:00

rtpsenderinterface.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +00:00

statstypes.cc

Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.

2017-07-14 19:17:49 +00:00

statstypes.h

Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.

2017-07-14 19:17:49 +00:00

streamcollection.h

Create //webrtc/api:libjingle_peerconnection_api + refactorings.

2017-01-23 12:56:25 +00:00

umametrics.h

Rename webrtc/base -> webrtc/rtc_base

2017-07-06 19:11:40 +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