Log - e8b962bd3c9775b2859f908004263dbc8840629a - platform/external/webrtc

ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 5 years ago
149dc72 Add support for RTCTransportStats.selectedCandidatePairChanges by Jonas Oreland · 5 years ago
0c141c5 Fix frames dropped statistics by Johannes Kron · 5 years 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
928e7a3 Make ID of datachannel stats not depend on dc.id by Harald Alvestrand · 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
8605fbf [getStats] Make remote-inbound-rtp.ssrc match outbound-rtp.ssrc. by Henrik Boström · 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 · 6 years ago
44125fa Reland "Piping audio interruption metrics to API layer" by Henrik Lundin · 6 years ago
cf96e0f Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. by Henrik Boström · 6 years ago
01738c6 Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. by Henrik Boström · 6 years ago
2e06926 Implement RTC[In/Out]boundRtpStreamStats.contentType. by Henrik Boström · 6 years ago
f71362f Wire up RTCOutboundRtpStreamStats.totalEncodeTime. by Henrik Boström · 6 years ago
232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 6 years ago
40b030e Reland "Fix getStats() freeze bug affecting Chromium but not WebRTC standalone." by Henrik Boström · 6 years ago
ca890ee Revert "Fix getStats() freeze bug affecting Chromium but not WebRTC standalone." by Mirko Bonadei · 6 years ago
05d43c6 Fix getStats() freeze bug affecting Chromium but not WebRTC standalone. by Henrik Boström · 6 years ago
0237106 Expose video freeze metrics in GetStats. by Sergey Silkin · 6 years ago
739baf0 [clang-tidy] Apply performance-for-range-copy fixes. by Mirko Bonadei · 6 years ago
0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from pc/rtcstatscollector.cc]
352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
dd9390c Prevent channels being set on stopped transceiver. by Amit Hilbuch · 6 years ago
6c6c9df Refactor: Renaming ssl_cert_chain to GetSSLCertificateChain() by Benjamin Wright · 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
6b1985d Reimplement rtc::ToString and rtc::FromString without streams. by Jonas Olsson · 6 years ago
918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
9f1de69 Add ADAPTER_TYPE_ANY in AdapterType. by Qingsi Wang · 6 years ago
43568dd Remove stringstreams from pc/ by Jonas Olsson · 6 years ago
7eca093 Ensure that data channel transport stats are included by Steve Anton · 7 years ago
5b3541f RTCStatsCollector::GetStatsReport() with optional selector argument. by Henrik Boström · 7 years ago
13b8bad Final name changing of MediaStreamInterface.label() to id(). by Seth Hampson · 7 years ago
25e022f Deliver cached stats reports asynchronously. by Taylor Brandstetter · 7 years ago
87d5a74 Fix crash that occurs if GetStats is called from within OnStatsDelivered by Taylor Brandstetter · 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
c392866 Implement certificate chain stats. by Taylor Brandstetter · 7 years ago
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
5dfde18 Change PeerConnection stats interface to be more flexible 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
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
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
89e7126 Optional: Use nullopt and implicit construction in /pc/rtcstatscollector.cc by Oskar Sundbom · 7 years ago
36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
d5585ca Move almost all references from WebRtcSession to PeerConnection by Steve Anton · 7 years ago
e2d6a06 Reland "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
1af3d82 Revert "Reland "Clean up libjingle API dependencies."" by Henrik Kjellander · 7 years ago
9185aca Reland "Clean up libjingle API dependencies." by Patrik Höglund · 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
581df61 Revert "Reland "Clean up libjingle API dependencies."" by Patrik Höglund · 7 years ago
5117b04 Reland "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
7bcfc3b Revert "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
bf66794 Revert "Move clients of WebRtcSession to use PeerConnection" by Alex Loiko · 7 years ago
57fb315 Clean up libjingle API dependencies. by Patrik Höglund · 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
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.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
80c829f Enable tracing on rtcstats_integrationtest.cc by ehmaldonado · 7 years ago
b0b721a Increase the size of the buffer for type.name.id. by ehmaldonado · 7 years ago
a26196b Trace stats in RTCStatsCollector. 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
7b0c6fa RTCStatsCollector: Get track IDs from senders/receivers instead of streams. by hbos · 7 years ago
c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago