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gerrit-public.fairphone.software / platform / external / webrtc / 01611ed3ac4102dc209e50bd618dd7933e11f314 / pc
  1. 7a9a092 Delete media transport integration. by Bjorn A Mellem · 4 years, 9 months ago
  2. 44d7ec0 Add Opus-only audio codec factories by Karl Wiberg · 4 years, 9 months ago
  3. d5e2f21 VideoRtpTrackSource: implement encoded source methods. by Markus Handell · 4 years, 9 months ago
  4. 353a718 Address failing wpt test cases for the rollback feature by Eldar Rello · 4 years, 9 months ago
  5. 15be528 Add support for RtpEncodingParameters::max_framerate by Florent Castelli · 4 years, 9 months ago
  6. 00376e1 Add totalInterFrameDelay to RTCInboundRTPStreamStats by Johannes Kron · 4 years, 9 months ago
  7. 15f2ff4 VideoRtpReceiver: factor out VideoRtpTrackSource. by Markus Handell · 4 years, 9 months ago
  8. 662678d Adds injectable trials from peerconnection down to transport controller. by Erik Språng · 4 years, 9 months ago
  9. 408cb4b Make SCTPtransport enter "closed" state when DTLStransport does. by Harald Alvestrand · 4 years, 9 months ago
  10. 25ec888 Make ICE transports injectable. by Qingsi Wang · 4 years, 9 months ago
  11. 7b46e17 In rtc::ByteBuffer drop support for ORDER_HOST as unused by Danil Chapovalov · 4 years, 9 months ago
  12. 01294f0 Don't configure video codec switching if no video stream has been created. by philipel · 4 years, 9 months ago
  13. ee6f4f6 [PeerConnection] Implement asynchronous version of AddIceCandidate(). by Henrik Boström · 4 years, 10 months ago
  14. 3ac6375 Add 3 missing RTC_EXPORT. by Mirko Bonadei · 4 years, 10 months ago
  15. c3d1f9b Enable injection of a custom NetEqFactory into PeerConnectionFactory. by Ivo Creusen · 4 years, 10 months ago
  16. 2ebbff8 do not offer gcm as the preferred cipher suite by Philipp Hancke · 4 years, 10 months ago
  17. 32913c1 Removes the flakiness in PeerConnectionUsageHistogramTest. by Qingsi Wang · 4 years, 10 months ago
  18. 4e19670 [PeerConnection] Implement parameterless SetLocalDescription(). by Henrik Boström · 4 years, 10 months ago
  19. c6c3f86 Expose TLS version and SRTP cipher to API by Harald Alvestrand · 4 years, 10 months ago
  20. a3728d3 Reland "[PeerConnection] Use an OperationsChain in PeerConnection for async ops." by Henrik Boström · 4 years, 10 months ago
  21. 5cb7807 Implement crypto stats on DTLS transport by Harald Alvestrand · 4 years, 10 months ago
  22. 16cec3b Added allow_codec_switching parameter to RTCConfig. by philipel · 4 years, 10 months ago
  23. 49c0880 Revert "[PeerConnection] Use an OperationsChain in PeerConnection for async ops." by Henrik Boström · 4 years, 10 months ago
  24. 1dddaa1 [PeerConnection] Use an OperationsChain in PeerConnection for async ops. by Henrik Boström · 4 years, 10 months ago
  25. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 4 years, 10 months ago
  26. ead0ec9 Add firing of OnRemoveTrack and OnRenegotationNeeded during rollback by Eldar Rello · 4 years, 10 months ago
  27. 86d053c Use source_sets in component builds and static_library in release builds. by Mirko Bonadei · 4 years, 10 months ago
  28. 8038541 Update the header extensions capabilities with mid, rid and rrid by Florent Castelli · 4 years, 10 months ago
  29. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 4 years, 11 months ago
  30. 35214fc Add missing RTC_EXPORT for the component build. by Mirko Bonadei · 5 years ago
  31. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 4 years, 10 months ago
  32. f8998cf Add a turn port prune policy to keep the first ready turn port. by Honghai Zhang · 4 years, 10 months ago
  33. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 4 years, 11 months ago
  34. 5ab79e6 Reland "Implement rollback for setRemoteDescription" by Eldar Rello · 4 years, 11 months ago
  35. 5740f3e Clarify expectation on GlobalLock by Danil Chapovalov · 4 years, 11 months ago
  36. d0704ce Remove RTCP tests from channel_unittest. by Bjorn A Mellem · 4 years, 11 months ago
  37. 907f154 Revert "Implement rollback for setRemoteDescription" by Alex Loiko · 4 years, 11 months ago
  38. 16d4c4d Implement rollback for setRemoteDescription by Eldar Rello · 4 years, 11 months ago
  39. f3f03e2 Removing outdated tests. by Alex Loiko · 4 years, 11 months ago
  40. ff27da5 Add/remove receive streams with SSRC 0 from media channels by Saurav Das · 5 years ago
  41. bfcec4c Delete old placeholders for moved api/ header files by Niels Möller · 5 years ago
  42. 8e1343a Add an alt-protocol to SDP to indicate which m= sections use a plugin transport. by Bjorn A Mellem · 5 years ago
  43. 7da4e56 Allow receive-only use of datagram transport for data channels. by Bjorn A Mellem · 5 years ago
  44. fc604aa Unset sinks when deleting CompositeDataChannelTransport. by Bjorn A Mellem · 5 years ago
  45. bc3eebc Reland "Reland "Refactor SCTP data channels to use DataChannelTransportInterface."" by Bjorn A Mellem · 5 years ago
  46. 1b57541 Always pass arguments to INSTANTIATE_TEST_SUITE_P. by Mirko Bonadei · 5 years ago
  47. ef14f07 Delete AudioDecoder method IncomingPacket by Niels Möller · 5 years ago
  48. 988e63e Proxy OnRtcpPacketReceived to the worker thread in channel tests. by Bjorn A Mellem · 5 years ago
  49. d702231 Cleanup deprecated monitoring of MediaTransport state. by Bjorn A Mellem · 5 years ago
  50. e942b14 New build target api:media_interface by Niels Möller · 5 years ago
  51. 1b83a9e Only handle each RTCP once. by Sebastian Jansson · 5 years ago
  52. 7b04a91 Delete almost all default methods on PeerConnectionInterface by Niels Möller · 5 years ago
  53. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  54. 7262fc2 Refactor Rtp Receivers to accept SSRC 0. by Saurav Das · 5 years ago
  55. eaaaf41 Introduce api/crypto/BUILD.gn. by Mirko Bonadei · 5 years ago
  56. e78fd80 New class DummyPeerConnection by Niels Möller · 5 years ago
  57. 01be33b Using lambdas instead of rtc::Bind in BaseChannel. by Sebastian Jansson · 5 years ago
  58. 262bbae Fix rare audioLevel flake in RTCStatsIntegrationTest. by Henrik Boström · 5 years ago
  59. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 5 years ago
  60. cc46b10 Add a usage pattern bit for host-host connections. by Qingsi Wang · 5 years ago
  61. fcfeefe Move rtc_error.{h,cc} to its own build target. by Mirko Bonadei · 5 years ago
  62. 437077d Revert "Reland "Refactor SCTP data channels to use DataChannelTransportInterface."" by Qingsi Wang · 5 years ago
  63. 487f9a1 Reland "Refactor SCTP data channels to use DataChannelTransportInterface." by Bjorn A Mellem · 5 years ago
  64. 116ffe7 Switch to compiling WebRTC -std=c++14 by default by Danil Chapovalov · 5 years ago
  65. cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 5 years ago
  66. be2e5f7 Make payload type demux conditional on media direction by Steve Anton · 5 years ago
  67. 20232a9 Use obfuscated IPs in logging in p2p/ and pc/. by Qingsi Wang · 5 years ago
  68. 662e31f Prepare to move packet_socket_factory to api/. by Patrik Höglund · 5 years ago
  69. 340e0c5 Delete old version of PeerConnection::SetConfiguration by Niels Möller · 5 years ago
  70. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 5 years ago
  71. 8b14b0d Revert "Refactor SCTP data channels to use DataChannelTransportInterface." by Henrik Boström · 5 years ago
  72. 4c85828 Refactor SCTP data channels to use DataChannelTransportInterface. by Bjorn A Mellem · 5 years ago
  73. 25eb47c Make the RtpHeaderParserImpl available to tests and tools only. by Tommi · 5 years ago
  74. 6dcd4dc New target for api/rtp_parameters.h and api/media_types.h. by Niels Möller · 5 years ago
  75. 703ea95 Only create a datagram RTP transport if the datagram transport should be used for RTP. by Bjorn A Mellem · 5 years ago
  76. fa046b3 Remove unused using statements in webrtc_sdp.cc by Elad Alon · 5 years ago
  77. 149dc72 Add support for RTCTransportStats.selectedCandidatePairChanges by Jonas Oreland · 5 years ago
  78. 0c141c5 Fix frames dropped statistics by Johannes Kron · 5 years ago
  79. 3c02842 Add TURN_LOGGING_ID by Jonas Oreland · 5 years ago
  80. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  81. b689af4 Changes to enable use of DatagramTransport as a data channel transport. by Bjorn A Mellem · 5 years ago
  82. 21e99da Add implemented-but-missing members to RTCMediaStreamTrackStats::Members by Henrik Boström · 5 years ago
  83. 364b267 Replace DatagramDtlsAdaptor with DatagramRtpTransport. by Bjorn A Mellem · 5 years ago
  84. 2579f0c RTCError as return type for PeerConnectionInterface::SetConfiguration by Niels Möller · 5 years ago
  85. a0b52b5 Remove zhihuang@webrtc.org from OWNERS by Steve Anton · 5 years ago
  86. 1ba5dec Reland "Set the usage pattern bits for adding remote ICE candidates from SDP." by Qingsi Wang · 5 years ago
  87. 6b43086 Reland "[GetStats] Expose video codec implementation in standardized metrics." by Henrik Boström · 5 years ago
  88. fa8f4ee Only combine media transport and ICE states if used for media. by Bjorn A Mellem · 5 years ago
  89. df625f4 Revert "[GetStats] Expose video codec implementation in standardized metrics." by Henrik Andreassson · 5 years ago
  90. 2b9fa09 [GetStats] Expose video codec implementation in standardized metrics. by Henrik Boström · 5 years ago
  91. bbeb109 Reporting audio device underrun counter by Alex Narest · 5 years ago
  92. d419808 Revert "Set the usage pattern bits for adding remote ICE candidates from SDP." by Qingsi Wang · 5 years ago
  93. 7c6f74a Set the usage pattern bits for adding remote ICE candidates from SDP. by Qingsi Wang · 5 years ago
  94. 9cfdb20 Control PeerConnectionFactory's default min/starting/max bitrates from experiment by Elad Alon · 5 years ago
  95. 43faee0 Implement JNI and objc implementation for Ice Candidate Pair Change event surfacing by Alex Drake · 5 years ago
  96. 78a7138 Remove MediaTransport from Call. by Tommi · 5 years ago
  97. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 5 years ago
  98. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 5 years ago
  99. 00c7ecf Surface CandidatePairChange event by Alex Drake · 5 years ago
  100. 6563934 Revert "Sanitize the codec list before sending it to the media engine" by Artem Titov · 5 years ago