- ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 4 years, 11 months ago
- ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 4 years, 10 months ago
- fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 4 years, 11 months ago
- 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
- 149dc72 Add support for RTCTransportStats.selectedCandidatePairChanges by Jonas Oreland · 5 years ago
- 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
- 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 5 years ago
- bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 5 years ago
- 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 5 years ago
- a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
- 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 5 years ago
- fc02a79 Revert "Piping audio interruption metrics to API layer" by Henrik Andreassson · 5 years ago
- 299c4e6 Piping audio interruption metrics to API layer by Henrik Lundin · 5 years ago
- 6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 5 years ago
- efe4c92 Use RtpSender/RtpReceiver track ID for legacy GetStats by Steve Anton · 5 years ago
- c84f661 Stop using Googletest legacy APIs. by Mirko Bonadei · 6 years ago
- 64b626b Use Abseil container algorithms in pc/ by Steve Anton · 6 years ago
- d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 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/statscollector_unittest.cc]
- 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
- 5f2ffee Clean up deprecated APM stats by Sam Zackrisson · 6 years ago
- 2812763 Remove deprecated AudioProcessing::GetStatistics function by Sam Zackrisson · 6 years ago
- f25303e Reland: Modernize rtc::SSLCertificate by Steve Anton · 6 years ago
- 82c71af Revert "Modernize rtc::SSLCertificate" by Niklas Enbom · 6 years ago
- 55cd3ac Modernize rtc::SSLCertificate by Steve Anton · 6 years ago
- 2e00abc Reland "[cleanup] Remove useless includes." by Yves Gerey · 6 years ago
- 96a0f61 Revert "[cleanup] Remove useless includes." by Oleh Prypin · 6 years ago
- be8b534 [cleanup] Remove useless includes. by Yves Gerey · 6 years ago
- 6b1985d Reimplement rtc::ToString and rtc::FromString without streams. by Jonas Olsson · 6 years ago
- 8a3ab0e Revert "Add framesRendered to StatsReport" by Artem Titov · 6 years ago
- dcfa938 Add framesRendered to StatsReport by Joachim Reiersen · 6 years ago
- a76af0c Move base64.h to the proper location. by Artem Titov · 6 years ago
- 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
- 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
- 845e878 Name change from stream label to stream id for spec compliance. by Seth Hampson · 6 years ago
- c392866 Implement certificate chain stats. by Taylor Brandstetter · 6 years ago
- 5b38731 Use fake PeerConnection for RTCStatsCollector tests by Steve Anton · 7 years ago
- 3871f6f Rewrite StatsCollector tests to use a fake PeerConnection by Steve Anton · 7 years ago
- be5e208 Add FakePeerConnectionBase by Steve Anton · 7 years ago
- 75ceef2 Pivot old stats generation to iterate senders/receivers by Harald Alvestrand · 7 years ago
- 7411648 Remove SessionStats.proxy_to_transport by Steve Anton · 7 years ago
- 7c5597a Remove unused enum (kStatsValueNameEchoCancellationQualityMin). by Gustaf Ullberg · 7 years ago
- 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
- 36f8f3e Optional: Use nullopt and implicit construction in /pc by Oskar Sundbom · 7 years ago
- c61ce0d Fixing some clang-tidy findings. by Mirko Bonadei · 7 years ago
- ae02609 Add parallel stats interface with optional stats to APM. by Ivo Creusen · 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
- 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
- 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/statscollector_unittest.cc]
- 0d0b912 Add and modify a few ANA stats. by ivoc · 7 years ago
- e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
- 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
- 773be36 Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt by perkj · 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
- 42308f6 Fix uploading of available send bitrate statistics. by Alex Narest · 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
- 112b2e9 Switching some interfaces to use std::unique_ptr<>. by deadbeef · 8 years ago
- cc452e1 Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ ) by sakal · 8 years ago
- 69fb2cc Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ ) by skvlad · 8 years ago
- ff0e72f Add QP sum stats for received streams. by sakal · 8 years ago
- f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 8 years ago
- 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/statscollector_unittest.cc]
- c8ee882 Replace use of ASSERT in test code. by nisse · 8 years ago
- 84abeb1 RTC[In/Out]boundRTPStreamStats.mediaTrackId collected. by hbos · 8 years ago
- 4e477a1 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 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
- df6075a RTCStatsCollector: Utilize network thread to minimize thread hops. by hbos · 8 years ago
- 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
- 49f34fd Relanding: Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 8 years ago
- 57fd726 Revert of Refactoring that removes P2PTransport and DtlsTransport classes. (patchset #9 id:150001 of https://codereview.webrtc.org/2517883002/ ) by deadbeef · 8 years ago
- bd28681 Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 8 years ago
- 87da404 Implement qpSum stat for video send ssrc stats. by sakal · 8 years ago
- e5ba44e Implement framesDecoded stat in video receive ssrc stats. by sakal · 8 years ago
- 43536c3 Implement framesEncoded stat in video send ssrc stats. by sakal · 8 years ago
- 8c63a82 Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 8 years ago
- 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
- ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
- 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
- 6348978 Add new decoding statistics for muted output by henrik.lundin · 8 years ago
- b24b1ce Moving mock classes around so that they may be reused in other unittests by hbos · 8 years ago
- 29ff844 Add PeerConnection IsClosed check. by zhihuang · 8 years ago
- e9021a3 Propogate network-worker thread split to api by danilchap · 8 years ago
- 6ba3b19 Filter out some variables with initial -1 in the stats report. by zhihuang · 8 years ago
- 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 8 years ago