ivoc 0d0b912128 Add and modify a few ANA stats. ...

This CL adds seperate counters for ANA frame length increases and decreases, which gives more insight into what actions are taken. In addition, a new stat is added to track the sum of the uplink packet loss fraction that is set by the ANA FEC controller.

BUG=webrtc:8127

Review-Url: https://codereview.webrtc.org/3007243002
Cr-Commit-Position: refs/heads/master@{#19756}

2017-09-08 20:24:21 +00:00

..

audio

Rename webrtc/base -> webrtc/rtc_base

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

audio_codecs

Add and modify a few ANA stats.

2017-09-08 20:24:21 +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

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

stats

Add stats totalSamplesReceived and concealedSamples

2017-08-25 00:50:42 +00:00

test

Rename webrtc/base -> webrtc/rtc_base

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

video

Implement googContentType GetStats metric reported on receive side.

2017-09-04 14:57:17 +00:00

video_codecs

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

array_view_unittest.cc

Move array_view.h to webrtc/api/

2017-09-04 12:43:17 +00:00

array_view.h

Move array_view.h to webrtc/api/

2017-09-04 12:43:17 +00:00

BUILD.gn

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +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

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +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

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +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

Change RtpSender to have multiple stream_ids

2017-09-07 17:03:24 +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

optional_unittest.cc

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

optional.cc

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

optional.h

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +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 support for changing the TLS elliptic curve set.

2017-09-08 20:02:03 +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_unittest.cc

Move RtpExtension to api/ directory and config.h/.cc to call/.

2017-09-01 13:29:30 +00:00

rtpparameters.cc

Move RtpExtension to api/ directory and config.h/.cc to call/.

2017-09-01 13:29:30 +00:00

rtpparameters.h

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

rtpreceiverinterface.h

Add audio_level member to RtpSource and set it from RtpReceiverImpl::IncomingRtpPacket.

2017-08-24 21:52:17 +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

Add and modify a few ANA stats.

2017-09-08 20:24:21 +00:00

statstypes.h

Add and modify a few ANA stats.

2017-09-08 20:24:21 +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