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webrtc / src / 6c94d58e34a1f511c18863b01b342342348079eb
  1. 6c94d58 Roll chromium_revision bbca8ebcc5..6f6904aacc (887470:887571) by chromium-webrtc-autoroll · 3 years, 11 months ago
  2. cbb4421 Remove DeliverPacketAsync. by Tommi · 3 years, 11 months ago
  3. 7857251 Update WebRTC code version (2021-05-29T04:03:30). by webrtc-version-updater · 3 years, 11 months ago
  4. d5b0199 Roll chromium_revision 4a5a62a362..bbca8ebcc5 (887362:887470) by chromium-webrtc-autoroll · 3 years, 11 months ago
  5. 3d46d0b Proxy: solve event tracing with compile time strings. by Markus Handell · 3 years, 11 months ago
  6. d325f32 Update WebRTC code version (2021-05-28T04:03:27). by webrtc-version-updater · 3 years, 11 months ago
  7. 319bac6 Roll chromium_revision f713d4fb04..4a5a62a362 (887229:887362) by chromium-webrtc-autoroll · 3 years, 11 months ago
  8. d595f6c Roll chromium_revision a0132a2044..f713d4fb04 (887053:887229) by chromium-webrtc-autoroll · 3 years, 11 months ago
  9. c39080c Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions by Artem Titov · 3 years, 11 months ago
  10. 236ac50 dcsctp: Add public API for BufferedAmountLow by Victor Boivie · 3 years, 11 months ago
  11. bd9031b dcsctp: Add OnTotalBufferedAmountLow in Send Queue by Victor Boivie · 3 years, 11 months ago
  12. 791adaf dcsctp: Add OnBufferedAmountLow in Send Queue by Victor Boivie · 3 years, 11 months ago
  13. a1b8201 Move proxies into pc/. by Markus Handell · 3 years, 11 months ago
  14. 7d2e669 dcsctp: Allocate TSN for end of abandoned message by Victor Boivie · 3 years, 11 months ago
  15. 9700d88 dcsctp: Avoid recalculation of outstanding bytes by Victor Boivie · 3 years, 11 months ago
  16. 36ad606 Update WebRTC code version (2021-05-27T04:02:45). by webrtc-version-updater · 3 years, 11 months ago
  17. e52cfab PipeWire capturer: request mouse cursor to be part of the stream by Jan Grulich · 3 years, 11 months ago
  18. 2182096 RtpFrameReferenceFinder return frames directly instead of via callback. by philipel · 3 years, 11 months ago
  19. 7f11067 Clean up RtpSenderTest and remove RtpSenderEgress dependencies. by Erik Språng · 3 years, 11 months ago
  20. b4f3204 Remove Win UWP mb config by Christoffer Jansson · 3 years, 11 months ago
  21. 940108b Apply autoformat to the docs by Artem Titov · 3 years, 11 months ago
  22. 8f8bf25 Remove usage of InjectPacket and transport_ in rtp_sender_unittest by Erik Språng · 3 years, 11 months ago
  23. b412efd payload type mapping: restrict lower range to <= 63 by Philipp Hancke · 3 years, 11 months ago
  24. a9af50f Introduce CreateDataChannelOrError by Harald Alvestrand · 3 years, 11 months ago
  25. 0d0ed76 Fix RTP header extension encryption by Lennart Grahl · 3 years, 11 months ago
  26. 4a54be7 doc: update dtls_transport.md to use new link style by Philipp Hancke · 3 years, 11 months ago
  27. fec79b7 add srtp docs by Philipp Hancke · 3 years, 11 months ago
  28. 770acab Refactor mid/rid rtp tests to avoid using egress/transport logic. by Erik Språng · 3 years, 11 months ago
  29. a39d966 Remove unused property isLocked from RTCAudioSession by Byoungchan Lee · 3 years, 11 months ago
  30. 8d9d575 PipeWire capturer: fix stream width in PW 0.2 code by Jan Grulich · 3 years, 11 months ago
  31. 27df007 Update WebRTC code version (2021-05-26T04:05:14). by webrtc-version-updater · 3 years, 11 months ago
  32. 048bf18 Roll chromium_revision 8a4c5eb899..19159a8788 (886374:886529) by chromium-webrtc-autoroll · 3 years, 11 months ago
  33. 0a52ede Support for map of string keys to uint64_t / double values in RTCStats by Byoungchan Lee · 3 years, 11 months ago
  34. cbeff55 Roll chromium_revision 1b27d646a6..8a4c5eb899 (886225:886374) by chromium-webrtc-autoroll · 3 years, 11 months ago
  35. 1573716 Enforce thread invoke policy for invokes to itself by Artem Titov · 3 years, 11 months ago
  36. 4fbc3fc Move SendPacketUpdates* tests to rtp_sender_egress_unittest. by Erik Språng · 3 years, 11 months ago
  37. fade919 Partial revert: "Use unordered map in RtpDemuxer" by Victor Boivie · 3 years, 11 months ago
  38. 238da9a Remove obsolete SendPacketMatches* tests from rtp_sender_egress_unittest. by Erik Språng · 3 years, 11 months ago
  39. 552169c Refactor RtpPacketCounter tests and move to rtp_sender_egress_unittest. by Erik Språng · 3 years, 11 months ago
  40. af0dff0 dcsctp: start SCTP_DUMP on a new line by Philipp Hancke · 3 years, 11 months ago
  41. 2ab4764 Clean-up for calculation of upper bandwidth limit. by Christoffer Rodbro · 3 years, 11 months ago
  42. 81e13d3 Roll chromium_revision 2826799ea1..1b27d646a6 (885837:886225) by chromium-webrtc-autoroll · 3 years, 11 months ago
  43. 36005af Refactor and improve RtpSender packet history test. by Erik Språng · 3 years, 11 months ago
  44. 02c0295 Remove obsolete DCHECK in RtpPacket::CopyHeaderFrom by Danil Chapovalov · 3 years, 11 months ago
  45. 6396b48 Avoid modifying BWE internal state on reception of REMB feedback. by Christoffer Rodbro · 3 years, 11 months ago
  46. c09c581 dcsctp: Limit the size of generated SACK chunks by Victor Boivie · 3 years, 11 months ago
  47. e4adedc Update WebRTC code version (2021-05-25T04:03:57). by webrtc-version-updater · 3 years, 11 months ago
  48. 41a111d Switch to av_packet_alloc() by Ted Meyer · 3 years, 11 months ago
  49. 0f50678 Remove usage of TOOLKIT_GTK define. by Byoungchan Lee · 3 years, 11 months ago
  50. 816134a Reland "Fix race between enabled() and set_enabled() in VideoTrack." by Tommi · 3 years, 11 months ago
  51. 2f3c5e6 Skip WindowCapturerTest.Capture on macOS. by Mirko Bonadei · 3 years, 11 months ago
  52. ae0d117 Implement the mixer-to-client per CSRC audio level extension (RFC 6465). by Doudou Kisabaka · 3 years, 11 months ago
  53. 096ad02 Revert "Fix race between enabled() and set_enabled() in VideoTrack." by Evan Shrubsole · 3 years, 11 months ago
  54. 2fa4774 Revert "Deprecate microsecond timestamps in RTC event log." by Björn Terelius · 3 years, 11 months ago
  55. cf49789 Refactor some retransmission tests. by Erik Språng · 3 years, 11 months ago
  56. 006815e Add temp peer_connection_interface include rtc_base/event.h by Evan Shrubsole · 3 years, 11 months ago
  57. e6ee8fa Deprecate microsecond timestamps in RTC event log. by Björn Terelius · 3 years, 11 months ago
  58. 5ffefe9 Fix race between enabled() and set_enabled() in VideoTrack. by Tommi · 3 years, 11 months ago
  59. 13e5851 Update WebRTC code version (2021-05-24T04:02:02). by webrtc-version-updater · 3 years, 11 months ago
  60. f3d71c2 Roll chromium_revision 70eb2d0977..2826799ea1 (885736:885837) by chromium-webrtc-autoroll · 3 years, 11 months ago
  61. 93faab1 dcsctp: Implement Round Robin scheduler by Victor Boivie · 3 years, 11 months ago
  62. 70cd086 SEA: Only spawn multi-layered encoders if active layers > 1. by Henrik Boström · 3 years, 11 months ago
  63. bcadacd Update WebRTC code version (2021-05-23T04:03:13). by webrtc-version-updater · 3 years, 11 months ago
  64. 2440d34 dcsctp: Rename FCFSSendQueue to RRSendQueue by Victor Boivie · 3 years, 11 months ago
  65. 913c3af Update WebRTC code version (2021-05-22T04:04:00). by webrtc-version-updater · 3 years, 11 months ago
  66. 7ee9b6b Roll chromium_revision 9431bab2be..70eb2d0977 (885625:885736) by chromium-webrtc-autoroll · 3 years, 11 months ago
  67. 9f6808b Remove cricket::DtlsTransportState. by Mirko Bonadei · 3 years, 11 months ago
  68. e976f75 Roll chromium_revision e5dd2eb61e..9431bab2be (884954:885625) by chromium-webrtc-autoroll · 3 years, 11 months ago
  69. 20f9401 Remove GTEST_ALLOW_UNINSTANTIATED in datachannel tests by Harald Alvestrand · 3 years, 11 months ago
  70. 32ee3b8 dcsctp: Ensure RTO is always greater than RTT by Victor Boivie · 3 years, 11 months ago
  71. cab90db Delete `NtpOffsetMs` and `TimeMicrosToNtp` methods. by Paul Hallak · 3 years, 11 months ago
  72. 46fbefa Convert to NTP time using the real clock. by Paul Hallak · 3 years, 11 months ago
  73. a6b0d53 Delete the old flavor of RtcpTransceiverImpl::ReceivePacket by Paul Hallak · 3 years, 11 months ago
  74. fe3dd51 Use the injected clock in rtcp_transciever. by Paul Hallak · 3 years, 11 months ago
  75. 61a287a Add accessor for UTC start time in event log by Björn Terelius · 3 years, 11 months ago
  76. 00f6e75 Use webrtc::Clock to query for the NTP time and to convert timestamps to NTP. by Paul Hallak · 3 years, 11 months ago
  77. 1cb796f Add performance tracing for API calls (inside api proxies). by Tommi · 3 years, 11 months ago
  78. 47ed998 Use the clock to convert absolute capture timestamps to NTP times. by Paul Hallak · 3 years, 11 months ago
  79. edc347c Introduce (Un)SubscribeDtlsTransportState methods. by Mirko Bonadei · 3 years, 11 months ago
  80. 95f1e51 Do not attempt setting the absolute capture time extension if we don't by Paul Hallak · 3 years, 11 months ago
  81. 2491dbd Make Clock::ConvertTimestampToNtpTime pure virtual by Paul Hallak · 3 years, 11 months ago
  82. e93fe6c Enable Chromium to stop including api/proxy.h indirectly. by Markus Handell · 3 years, 11 months ago
  83. b59e904 Add the ability to convert a timestamp to NTP time. by Paul Hallak · 3 years, 11 months ago
  84. 0cff391 Start with a BeginLog event in event log encoder unittest by Björn Terelius · 3 years, 11 months ago
  85. b8dc7fa Make AgcManagerDirect clipping parameters configurable by Hanna Silen · 3 years, 11 months ago
  86. e2b9fc6 Move FecOverheadRate, BitrateCallbacks to rtp_sender_egress_unittest. by Erik Språng · 3 years, 11 months ago
  87. 63b3095 Make local to capturer clock offset a separate entry in PacketInfo. by Minyue Li · 3 years, 11 months ago
  88. cbde0cf Roll chromium_revision 6291fe6f0e..e5dd2eb61e (884821:884954) by chromium-webrtc-autoroll · 3 years, 11 months ago
  89. 0de1ed0 Have only two pure virtual methods for webrtc::Clock, by Paul Hallak · 3 years, 11 months ago
  90. 8ed1e93 Switch from check_targets to no_check_targets in .gn by Byoungchan Lee · 3 years, 11 months ago
  91. 193f4bf Replace legacy getStats with standard getStats in the iOS example by Jaehyun Ko · 3 years, 11 months ago
  92. bd346d7 Update WebRTC code version (2021-05-20T04:01:58). by webrtc-version-updater · 3 years, 11 months ago
  93. 069ed35 Roll chromium_revision 0321a6153c..6291fe6f0e (884706:884821) by chromium-webrtc-autoroll · 3 years, 11 months ago
  94. 4fb5684 Roll chromium_revision b8d2317c2e..0321a6153c (884575:884706) by chromium-webrtc-autoroll · 3 years, 11 months ago
  95. 6c96611 Roll chromium_revision 3bdbd47d23..b8d2317c2e (884422:884575) by chromium-webrtc-autoroll · 3 years, 11 months ago
  96. 86bd92f Add test for many non-dropped packets in unreliable mode. by Harald Alvestrand · 3 years, 11 months ago
  97. e7481a4 Add an UlpFec test to RtpRtcp unit tests. by Erik Språng · 3 years, 11 months ago
  98. 398def6 Improvements to AEC3 logging to simplify debugging by Gustaf Ullberg · 3 years, 11 months ago
  99. aeb8ce8 AEC3: Change adaptation speed of the matched filter after a delay is found by Gustaf Ullberg · 3 years, 11 months ago
  100. 402ceff sctp: Reduce logging level for common calls by Victor Boivie · 3 years, 11 months ago