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gerrit-public.fairphone.software
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platform
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external
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webrtc
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2ae140ae7e34bd1e102ca832a14b7b0adbb72d15
/
pc
/
rtcstatscollector_unittest.cc
57858b3
Reland "Update RTCStatsCollector to work with RtpTransceivers"
by Steve Anton
· 7 years ago
ee2388f
Revert "Update RTCStatsCollector to work with RtpTransceivers"
by Guido Urdaneta
· 7 years ago
56bae8d
Update RTCStatsCollector to work with RtpTransceivers
by Steve Anton
· 7 years ago
5b38731
Use fake PeerConnection for RTCStatsCollector tests
by Steve Anton
· 7 years ago
76d2952
Don't crash when sender info has been discarded by lower layers.
by Harald Alvestrand
· 7 years ago
be5e208
Add FakePeerConnectionBase
by Steve Anton
· 7 years ago
2d8609c
Move internal PeerConnection methods to PeerConnectionInternal
by Steve Anton
· 7 years ago
b8e1201
Generate track stats when SSRC=0
by Harald Alvestrand
· 7 years ago
a3dab84
Refactor stream stats generation
by Harald Alvestrand
· 7 years ago
c72af93
Reland "Move stats ID generation from SSRC to local ID"
by Harald Alvestrand
· 7 years ago
c0092c3
Revert "Move stats ID generation from SSRC to local ID"
by Erik Språng
· 7 years ago
e357a4d
Move stats ID generation from SSRC to local ID
by Harald Alvestrand
· 7 years ago
8906187
Pivot generation of stats to iterate senders/receivers
by Harald Alvestrand
· 7 years ago
7411648
Remove SessionStats.proxy_to_transport
by Steve Anton
· 7 years ago
593e325
Change RTCStatsCollector to only access channels from signaling thread
by Steve Anton
· 7 years ago
719487e
Generate signed packets_lost in WebRTC-stats
by Harald Alvestrand
· 7 years ago
56d4609
Use the new AudioProcessing statistics everywhere.
by Ivo Creusen
· 7 years ago
37e489c
Add network_type to local RTCIceCandidateStats
by Gary Liu
· 7 years ago
c61ce0d
Fixing some clang-tidy findings.
by Mirko Bonadei
· 7 years ago
cbc71b2
Optional: Use nullopt and implicit construction in /pc/rtcstatscollector_unittest.cc
by Oskar Sundbom
· 7 years ago
8699a32
Have BaseChannel take MediaChannel as unique_ptr
by Steve Anton
· 7 years ago
75737c0
Merge WebRtcSession into PeerConnection
by Steve Anton
· 7 years ago
ba81867
Prepare WebRtcSession to be merged into PeerConnection
by Steve Anton
· 7 years ago
36b29d1
Enable cpplint in pc/
by Steve Anton
· 7 years ago
978b876
Move clients of WebRtcSession to use PeerConnection
by Steve Anton
· 7 years ago
b0a0207
Added RTCMediaStreamTrackStats.jitterBufferDelay for audio
by Gustaf Ullberg
· 7 years ago
bf66794
Revert "Move clients of WebRtcSession to use PeerConnection"
by Alex Loiko
· 7 years ago
3dc4d4a
Move clients of WebRtcSession to use PeerConnection
by Steve Anton
· 7 years ago
9a2e906
Added RTCMediaStreamTrackStats.concealmentEvents
by Gustaf Ullberg
· 7 years ago
563934e
Clean up dependencies of peerconnection_unittest.
by Patrik Höglund
· 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/rtcstatscollector_unittest.cc]
2dbc69f
Add stats totalSamplesReceived and concealedSamples
by Steve Anton
· 7 years ago
1cc5fc3
Fix places that trigger no-unused-lambda-capture
by eladalon
· 7 years ago
773be36
Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt
by perkj
· 7 years ago
35a872c
Make RTCStatsReport::ToString() return JSON-parseable string.
by ehmaldonado
· 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
539d104
Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ )
by mbonadei
· 7 years ago
f1377f7
Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread.
by perkj
· 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
f79ade1
Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )"
by stefan
· 7 years ago
d72098a
Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )
by charujain
· 7 years ago
e80f4c9
Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls.
by Stefan Holmer
· 7 years ago
eaabdf6
Delete MediaController class, move Call ownership to PeerConnection.
by nisse
· 7 years ago
5bf9def
RTCStatsCollector: Remove closed channels from opened set.
by hbos
· 7 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
92eaec6
RTCIceCandidatePairStats.nominated collected.
by hbos
· 8 years ago
a51d4f3
Re-land of RTCInboundRTPStreamStats.qpSum collected.
by hbos
· 8 years ago
112b2e9
Switching some interfaces to use std::unique_ptr<>.
by deadbeef
· 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
3443bb7
RTCRTPStreamStats.ssrc changed type to uint32_t.
by hbos
· 8 years ago
e702b30
Adding C++ versions of currently spec'd "RtpParameters" structs.
by deadbeef
· 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/rtcstatscollector_unittest.cc]
f64941f
RTCMediaStreamTrackStats.framesDecoded collected.
by hbos
· 8 years ago
fefe076
RTCMediaStreamTrackStats.framesSent collected by RTCStatsCollector.
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
c8ee882
Replace use of ASSERT in test code.
by nisse
· 8 years ago
bad5dad
More minor improvements to BaseChannel/transport code.
by deadbeef
· 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
ac22f70
Refactoring of RTCP options in BaseChannel.
by deadbeef
· 8 years ago
f5b251b
Remove BaseChannel's dependency on TransportController.
by zhihuang
· 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
c3a2b7f
RTCIceCandidateStats.isRemote added and collected.
by hbos
· 8 years ago
f415f8a
Removed RTCStatsCollector::ProducePartialResultsOnWorkerThread.
by hbos
· 8 years ago
23368e1
RTCStatsCollectorTest: ExpectReportContainsCertificateInfo /w EXPECT_EQ
by hbos
· 8 years ago
c42ba32
RTCStatsCollectorTest: Remove ExpectReportContainsCandidate.
by hbos
· 8 years ago
dbb64d8
RTCStatsCollectorTest: Remove ExpectReportContainsDataChannel.
by hbos
· 8 years ago
02d2a92
RTCStatsReport::AddStats DCHECKs that the ID is unique.
by hbos
· 8 years ago
df6075a
RTCStatsCollector: Utilize network thread to minimize thread hops.
by hbos
· 8 years ago
3168c7a
Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time.
by hbos
· 8 years ago
e10e6d1
RTCOutboundRTPStreamStats.roundTripTime: Only report non-negative values.
by hbos
· 8 years ago
9a394f0
Skip RTCMediaStreamTrackStats.echoReturnLoss[Enhancement] default value.
by hbos
· 8 years ago
7af91dd
Removing "crypto_required" from MediaContentDescription.
by deadbeef
· 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
0adb828
RTCCodecStats[1] added.
by hbos
· 8 years ago
820f578
RTCInboundRTPStreamStats's [fir/pli/nack]_count are collected for video.
by hbos
· 8 years ago
82ebe02
Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed].
by hbos
· 8 years ago
acd935b
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
by nisse
· 8 years ago
09bc128
RTCMediaStream[Track]Stats added.
by hbos
· 8 years ago
eeafe94
RTCInboundRTPStreamStats[1] added.
by hbos
· 8 years ago
6ded190
RTCOutboundRTPStreamStats[1] added.
by hbos
· 8 years ago
7eaa836
Revert of RTCOutboundRTPStreamStats added. (patchset #3 id:80001 of https://codereview.webrtc.org/2456463002/ )
by perkj
· 8 years ago
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