Log - 68586e80fcb7d34860452995578e2be4afc485c0 - platform/external/webrtc

68586e8 Replace starts_with and ends_with with Abseil by Steve Anton · 6 years ago
3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
59cfd35 Address vptr race condition while PeerConnection is destructed. by Yves Gerey · 6 years ago
8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
b5bb513 Disable RTCStatsIntegrationTest.GetsStatsWhileDestroyingPeerConnection by Yves Gerey · 6 years ago
9551375 getStats: add relayProtocol by Philipp Hancke · 6 years ago
3bc0166 getStats: add kind alias for mediaType by Philipp Hancke · 6 years ago
f06f923 Delete almost all use of MediaConstraintsInterface in the PeerConnection API by Niels Möller · 6 years ago
e12c1fe Removing warning suppression flags from pc/. by Mirko Bonadei · 6 years ago
665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
2d02e08 Delete deprecated CreateAudioSource method, with constraints. by Niels Möller · 6 years ago
25cfeb9 Fix possible race between the stats collector and transport controller by Steve Anton · 6 years ago
1df1bf8 PeerConnectionInterface::GetStats() with selector argument added. by Henrik Boström · 7 years ago
b619936 Stats traversal algorithm added. by Henrik Boström · 7 years ago
70473fc Reland "Add hugeFramesSent GetStats metric" by Ilya Nikolaevskiy · 7 years ago
8ddc2e6 Revert "Add hugeFramesSent GetStats metric" by Max Morin · 7 years ago
f9f71b9 Add hugeFramesSent GetStats metric by Ilya Nikolaevskiy · 7 years ago
719487e Generate signed packets_lost in WebRTC-stats by Harald Alvestrand · 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
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