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webrtc_m130/webrtc/api
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ilnik 2e1b40bdf6 Implement googContentType GetStats metric reported on receive side. 
Reported per video stream as a string.

BUG=webrtc:8174

Review-Url: https://codereview.webrtc.org/3009793002
Cr-Commit-Position: refs/heads/master@{#19667}
2017-09-04 14:57:17 +00:00
..
audio
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
audio_codecs
Move array_view.h to webrtc/api/
2017-09-04 12:43:17 +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
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 kMinPixelsPerFrame constant in VideoStreamEncoder to VideoEncoder::ScalingSettings.
2017-08-17 15:58:54 +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 array_view.h to webrtc/api/
2017-09-04 12:43:17 +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
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
Support a user-provided string for the TLS ALPN extension.
2017-08-29 20:11:16 +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 RtpExtension to api/ directory and config.h/.cc to call/.
2017-09-01 13:29:30 +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
Implement googContentType GetStats metric reported on receive side.
2017-09-04 14:57:17 +00:00
statstypes.h
Implement googContentType GetStats metric reported on receive side.
2017-09-04 14:57:17 +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