- fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 4 years, 9 months ago
- ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 4 years, 9 months ago
- ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 4 years, 9 months ago
- fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 4 years, 9 months ago
- 262bbae Fix rare audioLevel flake in RTCStatsIntegrationTest. by Henrik Boström · 4 years, 10 months ago
- cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 4 years, 10 months ago
- 149dc72 Add support for RTCTransportStats.selectedCandidatePairChanges by Jonas Oreland · 4 years, 11 months ago
- 21e99da Add implemented-but-missing members to RTCMediaStreamTrackStats::Members by Henrik Boström · 4 years, 11 months ago
- 6b43086 Reland "[GetStats] Expose video codec implementation in standardized metrics." by Henrik Boström · 5 years ago
- df625f4 Revert "[GetStats] Expose video codec implementation in standardized metrics." by Henrik Andreassson · 5 years ago
- 2b9fa09 [GetStats] Expose video codec implementation in standardized metrics. by Henrik Boström · 5 years ago
- a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
- d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
- bfd343b Add totalDecodeTime to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
- 2efae77 Add RTCStats for keyFramesEncoded, keyFramesDecoded. by Rasmus Brandt · 5 years ago
- 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 5 years ago
- 6737841 Add jitterBufferDelay and jitterBufferEmittedCount stats for video by Guido Urdaneta · 5 years ago
- ce33b6a Implement QualityLimitationReasonTracker and expose "reason". by Henrik Boström · 5 years ago
- 883eefc Implement RTCRemoteInboundRtpStreamStats for both audio and video. by Henrik Boström · 5 years ago
- 646fda0 Implement RTCMediaSourceStats and friends in standard getStats(). by Henrik Boström · 5 years ago
- 23aff9b Implement RTCOutboundRtpStreamStats.totalEncodedBytesTarget. by Henrik Boström · 5 years ago
- 9fe1834 Implement RTCOutboundRtpStreamStats.totalPacketSendDelay for video. by Henrik Boström · 5 years ago
- 8d8ffdb Expose new audio stats on the API by Ivo Creusen · 5 years ago
- 44125fa Reland "Piping audio interruption metrics to API layer" by Henrik Lundin · 5 years ago
- cf96e0f Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. by Henrik Boström · 5 years ago
- 01738c6 Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. by Henrik Boström · 5 years ago
- 6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 5 years ago
- 2e06926 Implement RTC[In/Out]boundRtpStreamStats.contentType. by Henrik Boström · 5 years ago
- f71362f Wire up RTCOutboundRtpStreamStats.totalEncodeTime. by Henrik Boström · 5 years ago
- 232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 5 years ago
- 5c4ddad Delete obsolete usage of FakeConstraints by Niels Möller · 5 years ago
- 0237106 Expose video freeze metrics in GetStats. by Sergey Silkin · 5 years ago
- 64b626b Use Abseil container algorithms in pc/ by Steve Anton · 5 years ago
- d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 5 years ago
- 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 5 years ago
- 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 5 years ago
- 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 5 years ago[Renamed from pc/rtcstats_integrationtest.cc]
- 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 · 6 years ago
- b619936 Stats traversal algorithm added. by Henrik Boström · 6 years ago
- 70473fc Reland "Add hugeFramesSent GetStats metric" by Ilya Nikolaevskiy · 6 years ago
- 8ddc2e6 Revert "Add hugeFramesSent GetStats metric" by Max Morin · 6 years ago
- f9f71b9 Add hugeFramesSent GetStats metric by Ilya Nikolaevskiy · 6 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 · 7 years ago
- a7a9be1 Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect. by hbos · 7 years ago
- 13f54b2 Rename RTCCodecStats.codec -> mimeType, parameters -> sdpFmtpLine. by hbos · 7 years ago
- bf8d3e5 RTCIceCandidatePairStats.[total/current]RoundTripTime collected. by hbos · 7 years ago
- 3fd31fe Fix TSAN race in webrtc::voe::Channel. by hbos · 7 years ago
- 92eaec6 RTCIceCandidatePairStats.nominated collected. by hbos · 7 years ago
- 1498def Disable RTCStatsIntegrationTest on TSAN bots. by philipel · 7 years ago
- 5fec128 Add QP for libvpx VP8 decoder. by sakal · 7 years ago
- a51d4f3 Re-land of RTCInboundRTPStreamStats.qpSum collected. by hbos · 7 years ago
- ed02c6d Revert of RTCInboundRTPStreamStats.qpSum collected. (patchset #4 id:80001 of https://codereview.webrtc.org/2675943002/ ) by skvlad · 7 years ago
- cd195be RTCInboundRTPStreamStats.qpSum collected. by hbos · 7 years ago
- 338f78a RTCIceCandidatePairStats.available[Outgoing/Incoming]Bitrate collected. by hbos · 7 years ago
- b0ae920 RTCRTPStreamStats.mediaTrackId renamed to trackId. by hbos · 7 years ago
- 50cfe1f RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector. by hbos · 7 years ago
- 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 7 years ago[Renamed (99%) from webrtc/api/rtcstats_integrationtest.cc]
- f64941f RTCMediaStreamTrackStats.framesDecoded collected. by hbos · 7 years ago
- fefe076 RTCMediaStreamTrackStats.framesSent collected by RTCStatsCollector. by hbos · 7 years ago
- 2d4d653 Fix msan flake in rtcstats_integrationtest.cc. by hbos · 7 years ago
- 42f6d2f RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector. by hbos · 7 years ago
- 9e30274 RTCMediaStreamTrackStats collected on a per-attachment basis. by hbos · 7 years ago
- 160e4a7 RTCMediaStreamTrackStats.kind added and collected. by hbos · 7 years ago