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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
8a47db590d18a1c723ca89bd3e2fd8600b2fd303
/
pc
/
rtcstats_integrationtest.cc
36b29d1
Enable cpplint in pc/
by Steve Anton
· 7 years ago
b0a0207
Added RTCMediaStreamTrackStats.jitterBufferDelay for audio
by Gustaf Ullberg
· 7 years ago
9a2e906
Added RTCMediaStreamTrackStats.concealmentEvents
by Gustaf Ullberg
· 7 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago
[Renamed from webrtc/pc/rtcstats_integrationtest.cc]
8ab0fd8
Reland of Trace the stats report as JSON instead of each stat separately. (patchset #1 id:1 of https://codereview.webrtc.org/3001683002/ )
by ehmaldonado
· 7 years ago
2dbc69f
Add stats totalSamplesReceived and concealedSamples
by Steve Anton
· 7 years ago
3439c89
Revert of Trace the stats report as JSON instead of each stat separately. (patchset #3 id:100001 of https://codereview.webrtc.org/2986453002/ )
by mbonadei
· 7 years ago
80c6578
Trace the stats report as JSON instead of each stat separately.
by ehmaldonado
· 7 years ago
35a872c
Make RTCStatsReport::ToString() return JSON-parseable string.
by ehmaldonado
· 7 years ago
80c829f
Enable tracing on rtcstats_integrationtest.cc
by ehmaldonado
· 7 years ago
e725159
Reland of Make the default ctor of rtc::Thread, protected
by tommi
· 7 years ago
e76bd3a
Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy.
by zstein
· 7 years ago
a117b04
Revert of Make the default ctor of rtc::Thread, protected (patchset #3 id:40001 of https://codereview.webrtc.org/2981623002/ )
by charujain
· 7 years ago
a8a3515
Make the default ctor of rtc::Thread, protected.
by tommi
· 7 years ago
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 7 years ago
a80c16a
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 7 years ago
c3771cc
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 7 years ago
86c40a1
Fixing RTCIceCandidatePairStats.nominated for ICE controlling agent.
by deadbeef
· 7 years ago
6488ea4
Remove temporary include of builtin_audio_encoder_factory.h.
by ossu
· 7 years ago
98e186c
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
by deadbeef
· 7 years ago
9e5b11e
Test CreatePeerConnectionFactory() with a forwarding mock AudioDecoderFactory
by kwiberg
· 8 years ago
a7a9be1
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
by hbos
· 8 years ago
13f54b2
Rename RTCCodecStats.codec -> mimeType, parameters -> sdpFmtpLine.
by hbos
· 8 years ago
bf8d3e5
RTCIceCandidatePairStats.[total/current]RoundTripTime collected.
by hbos
· 8 years ago
3fd31fe
Fix TSAN race in webrtc::voe::Channel.
by hbos
· 8 years ago
92eaec6
RTCIceCandidatePairStats.nominated collected.
by hbos
· 8 years ago
1498def
Disable RTCStatsIntegrationTest on TSAN bots.
by philipel
· 8 years ago
5fec128
Add QP for libvpx VP8 decoder.
by sakal
· 8 years ago
a51d4f3
Re-land of RTCInboundRTPStreamStats.qpSum collected.
by hbos
· 8 years ago
ed02c6d
Revert of RTCInboundRTPStreamStats.qpSum collected. (patchset #4 id:80001 of https://codereview.webrtc.org/2675943002/ )
by skvlad
· 8 years ago
cd195be
RTCInboundRTPStreamStats.qpSum collected.
by hbos
· 8 years ago
338f78a
RTCIceCandidatePairStats.available[Outgoing/Incoming]Bitrate collected.
by hbos
· 8 years ago
b0ae920
RTCRTPStreamStats.mediaTrackId renamed to trackId.
by hbos
· 8 years ago
50cfe1f
RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector.
by hbos
· 8 years ago
7bb87ee
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
by ossu
· 8 years ago
[Renamed (99%) from webrtc/api/rtcstats_integrationtest.cc]
f64941f
RTCMediaStreamTrackStats.framesDecoded collected.
by hbos
· 8 years ago
fefe076
RTCMediaStreamTrackStats.framesSent collected by RTCStatsCollector.
by hbos
· 8 years ago
2d4d653
Fix msan flake in rtcstats_integrationtest.cc.
by hbos
· 8 years ago
42f6d2f
RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector.
by hbos
· 8 years ago
9e30274
RTCMediaStreamTrackStats collected on a per-attachment basis.
by hbos
· 8 years ago
160e4a7
RTCMediaStreamTrackStats.kind added and collected.
by hbos
· 8 years ago
7064d59
RTCTransportStats.dtlsState replaces .activeConnection
by hbos
· 8 years ago
84abeb1
RTC[In/Out]boundRTPStreamStats.mediaTrackId collected.
by hbos
· 8 years ago
bb37751
RTCMediaStreamTrackStats.ssrcIds removed.
by hbos
· 8 years ago
b4e426e
RTCIceCandidateStats.transportId added and collected.
by hbos
· 8 years ago
6769c49
RTC[In/Out]boundRTPStreamStats: qpSum,framesDecoded,framesEncoded added.
by hbos
· 8 years ago
d17a5a7
RTCIceCandidateStats.deleted = false added.
by hbos
· 8 years ago
06495bc
RTCIceCandidatePairStats.[state/priority] added, ConnectionInfo updated.
by hbos
· 8 years ago
2335119
Improve rtcstats_integrationtest.cc by sanity checking values.
by hbos
· 8 years ago
c3a2b7f
RTCIceCandidateStats.isRemote added and collected.
by hbos
· 8 years ago
40610e2
Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define.
by deadbeef
· 8 years ago
b78306a
Fix segfault when PeerConnection is destroyed during stats collection.
by hbos
· 8 years ago
3168c7a
Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time.
by hbos
· 8 years ago
7bf5369
RTCStatsIntegrationTest: TestMemberIsIDReference on all defined IDs.
by hbos
· 8 years ago
e10e6d1
RTCOutboundRTPStreamStats.roundTripTime: Only report non-negative values.
by hbos
· 8 years ago
e448dd5
RTCIceCandidatePairStats.consentRequestsSent set by RTCStatsCollector
by hbos
· 8 years ago
02cd4d6
RTCInboundRTPStreamStats.packetsLost set by RTCStatsCollector.
by hbos
· 8 years ago
d82f512
RTCIceCandidatePairStats.requestsReceived defined by RTCStatsCollector.
by hbos
· 8 years ago
0583b28
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
by hbos
· 8 years ago
db346a7
RTCStatsIntegrationTest added.
by hbos
· 8 years ago