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gerrit-public.fairphone.software / platform / external / webrtc / 0e96535be97916d8fcaa9873ffab3c636539f9d8 / call
  1. cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 4 years, 11 months ago
  2. 6516f76 Deprecate SingleThreadedTaskQueueForTesting class. by Yves Gerey · 5 years ago
  3. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 5 years ago
  4. f13df86 Delete audio methods SignalNetworkState by Niels Möller · 5 years ago
  5. 55dd72c Remove lock for process thread pointer from PacedSender. by Tommi · 5 years ago
  6. 25eb47c Make the RtpHeaderParserImpl available to tests and tools only. by Tommi · 5 years ago
  7. b4a6128 Delete unneeded dependencies on libjingle_peerconnection_api by Niels Möller · 5 years ago
  8. 6dcd4dc New target for api/rtp_parameters.h and api/media_types.h. by Niels Möller · 5 years ago
  9. fac7e31 Removes TransportSequenceNumberAllocator by Erik Språng · 5 years ago
  10. 0c141c5 Fix frames dropped statistics by Johannes Kron · 5 years ago
  11. d77cc24 New const method StreamStatistician::GetStats by Niels Möller · 5 years ago
  12. 4e615d5 Wire the stable target bitrate from GoogCC to the BitrateAllocator by Florent Castelli · 5 years ago
  13. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  14. 54d5d2c Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc by Erik Språng · 5 years ago
  15. 6b117a5 Make the callbacks to PollStats for RampUp* tests more regular. by Tommi · 5 years ago
  16. 93f5189 Remove some usage of RtpRtcp::SetSSRC() by Erik Språng · 5 years ago
  17. 6950b30 Fix thread naming in Call constructor by Erik Språng · 5 years ago
  18. c14b233 Disable the most flaky tests on iOS. by Sami Kalliomäki · 5 years ago
  19. eea605d Make fake network degradation work also for sent audio by Erik Språng · 5 years ago
  20. 48b48e5 Enable thread check in Call::GetStats(). by Tommi · 5 years ago
  21. 78a7138 Remove MediaTransport from Call. by Tommi · 5 years ago
  22. 891d393 Call Call::GetStats() from the correct thread in ProbingEndToEndTest. by Tommi · 5 years ago
  23. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 5 years ago
  24. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 5 years ago
  25. c24a5b1 Fix CallPerfTests to call Call::GetStats() from the right thread. by Tommi · 5 years ago
  26. c648819 DegradedCall: fake network using TaskQueue instead of ProcessThread by Erik Språng · 5 years ago
  27. 5e005f4 Fix RampUp tests to call Call::GetStats() from the right thread - and remove the need for a dedicated polling thread. by Tommi · 5 years ago
  28. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 5 years ago
  29. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 5 years ago
  30. 425d6aa Add RtpPacketPacer interface for pacer control by Erik Språng · 5 years ago
  31. 463d44a Don't crash when simulcast layer count is different from RID count by Florent Castelli · 5 years ago
  32. aa59eca Move RtpPacketSender and merge it with RtpPacketPacer. by Erik Språng · 5 years ago
  33. e1795f4 Adds remote estimate RTCP packet. by Sebastian Jansson · 5 years ago
  34. 2ab97f6 Migrate WebRTC test infra to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  35. 0182a03 Reland "Remove the injectable bitrate allocation strategy API." by Jonas Olsson · 5 years ago
  36. e95b57c Revert "Remove the injectable bitrate allocation strategy API." by Mirko Bonadei · 5 years ago
  37. 80cb3f6 Remove the injectable bitrate allocation strategy API. by Jonas Olsson · 5 years ago
  38. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  39. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
  40. 53d45ba Make TaskQueueFactory required construction parameter for Call by Danil Chapovalov · 5 years ago
  41. bfd343b Add totalDecodeTime to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
  42. 67daf71 Implement RtpVideoSender::SetFecAllowed() by Elad Alon · 5 years ago
  43. 8f01c4e Define FecControllerOverride and plumb it down to VideoEncoder by Elad Alon · 5 years ago
  44. 4ba04b7 Delete RtcEventLogFactory factory as now unused by Danil Chapovalov · 5 years ago
  45. 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 5 years ago
  46. 59b8654 Switch from RtpPacketSender to RtpPacketPacer interface usage. by Erik Språng · 5 years ago
  47. 4d504c7 New interface EncodedImageBufferInterface, replacing use of CopyOnWriteBuffer by Niels Möller · 5 years ago
  48. 3d61ab1 Adds send time to ReceivedPacket struct. by Sebastian Jansson · 5 years ago
  49. 8aba8fe Reland "Populate the GFD-00 for H264 and generic codecs." by philipel · 5 years ago
  50. 607a6f1 Moves conversion to ReceivedPacket from RtpPacketReceived to Call. by Sebastian Jansson · 5 years ago
  51. 04e129a Revert "Populate the GFD-00 for H264 and generic codecs." by Philip Eliasson · 5 years ago
  52. d3c6f9c Populate the GFD-00 for H264 and generic codecs. by philipel · 5 years ago
  53. 08fa953 Reland "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" by Danil Chapovalov · 5 years ago
  54. 370f93a Reland "Inform VideoEncoder of negotiated capabilities" by Elad Alon · 5 years ago
  55. fd5166c Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" by Philip Eliasson · 5 years ago
  56. fc96135 Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory by Danil Chapovalov · 5 years ago
  57. 49d661a Revert "Inform VideoEncoder of negotiated capabilities" by Philip Eliasson · 5 years ago
  58. 11dfff0 Inform VideoEncoder of negotiated capabilities by Elad Alon · 5 years ago
  59. 4ad51d8 Removes SendSideCongestionController. by Sebastian Jansson · 5 years ago
  60. cf41eb1 Reland "Cleanup of video packet overhead calculation." by Sebastian Jansson · 5 years ago
  61. 5d24b16 Prepare for splitting the api/video:video_frames build rule. by Chen Xing · 5 years ago
  62. 27a7e9f Remove myself from OWNERS in a few places. by Fredrik Solenberg · 5 years ago
  63. f6c914a Revert "Reland "Cleanup of video packet overhead calculation."" by Sebastian Jansson · 5 years ago
  64. 35d4e43 Reland "Cleanup of video packet overhead calculation." by Sebastian Jansson · 5 years ago
  65. b64af4b Add retransmission_allowed flag to encoder output by Elad Alon · 5 years ago
  66. 845c6aa Add support for early loss detection using transport feedback. by Erik Språng · 5 years ago
  67. 6737841 Add jitterBufferDelay and jitterBufferEmittedCount stats for video by Guido Urdaneta · 5 years ago
  68. ce33b6a Implement QualityLimitationReasonTracker and expose "reason". by Henrik Boström · 5 years ago
  69. 8b27910 Include downlink delay into congestion window size. by Ying Wang · 5 years ago
  70. 6e436d1 [audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
  71. 87e3f9d [video] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
  72. a0e9943 Negotiation of LNTF controls instantiation of RTPSenderVideo::rtp_sequence_number_map_ by Elad Alon · 5 years ago
  73. fadb181 Negotiate use of RTCP loss notification feedback (LNTF) by Elad Alon · 5 years ago
  74. f3db34d Revert "Cleanup of video packet overhead calculation." by Sebastian Jansson · 5 years ago
  75. 62ce035 RtpVideoSender nits by Elad Alon · 5 years ago
  76. 890bc30 Cleanup of video packet overhead calculation. by Sebastian Jansson · 5 years ago
  77. fe68daa Add option to configure raw RTP packetization per payload type. by Mirta Dvornicic · 5 years ago
  78. 23aff9b Implement RTCOutboundRtpStreamStats.totalEncodedBytesTarget. by Henrik Boström · 5 years ago
  79. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
  80. 9fe1834 Implement RTCOutboundRtpStreamStats.totalPacketSendDelay for video. by Henrik Boström · 5 years ago
  81. fd26ef7 Delete unused RTPFragmentationHeader members by Niels Möller · 5 years ago
  82. 166b45d Adds route changes in event logs. by Sebastian Jansson · 5 years ago
  83. bd7046c Remove redundant feedback_packet_seq_num_set_ in RtpVideoSender by Erik Språng · 5 years ago
  84. 8f119ca Enable experiments with audio bitrate priority. by Jonas Olsson · 5 years ago
  85. d8b9ed7 Promote RtcEventLogOutputFile to api/ by Niels Möller · 5 years ago
  86. 490d76c Remove packets from RtpPacketHistory if acked via TransportFeedback by Erik Språng · 5 years ago
  87. 8d8ffdb Expose new audio stats on the API by Ivo Creusen · 5 years ago
  88. d42a449 Reland "Add more adaptation text logs for VideoSendStream." by Rasmus Brandt · 5 years ago
  89. 44125fa Reland "Piping audio interruption metrics to API layer" by Henrik Lundin · 5 years ago
  90. fc02a79 Revert "Piping audio interruption metrics to API layer" by Henrik Andreassson · 5 years ago
  91. 299c4e6 Piping audio interruption metrics to API layer by Henrik Lundin · 5 years ago
  92. f8c1ed5 Revert "Remove packets from RtpPacketHistory if acked via TransportFeedback" by Erik Språng · 5 years ago
  93. ce92817 Split test:test_common source set by Danil Chapovalov · 5 years ago
  94. 3890e99 Remove packets from RtpPacketHistory if acked via TransportFeedback by Erik Språng · 5 years ago
  95. 30a276b Add RTP sequence number to TransportFeedbackObserver::AddPacket() by Erik Språng · 5 years ago
  96. 81687b3 Use explicit TaskQueueFactory for FrameGeneratorCapturer in BitrateEstimatorTest. by Danil Chapovalov · 5 years ago
  97. cf96e0f Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. by Henrik Boström · 5 years ago
  98. 034f191 Revert "Add more adaptation text logs for VideoSendStream." by Rasmus Brandt · 5 years ago
  99. d0298f4 Add more adaptation text logs for VideoSendStream. by Rasmus Brandt · 5 years ago
  100. 884adca Log details when RtpDemuxer fails to deliver a packet by Steve Anton · 5 years ago