Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / c71d85bc4e719fcf28f8f89b2975292391f1c021 / call
  1. 04671b0 Delete unused method PacedSender::QueueSizePackets by Niels Möller · 5 years ago
  2. 7ea9b80 Set StreamDataCountersCallback on construction of RTP modules by Erik Språng · 5 years ago
  3. 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 5 years ago
  4. 82ed2e8 Cleanup: Propagating BitrateAllocationUpdate to RtpVideoSender by Sebastian Jansson · 5 years ago
  5. 6841d25 Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const" by Erik Språng · 5 years ago
  6. f39c815 Cleanup: Replacing set extension status bool with CHECK. by Sebastian Jansson · 5 years ago
  7. e8a6bc3 Revert "Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const"" by Erik Språng · 5 years ago
  8. c934821 Reland "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const" by Erik Språng · 5 years ago
  9. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
  10. 4ed0b52 Revert "RtpRtcp modules and below: Make media, RTX and FEC SSRCs const" by Erik Språng · 5 years ago
  11. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
  12. 17608dc RtpRtcp modules and below: Make media, RTX and FEC SSRCs const by Erik Språng · 5 years ago
  13. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
  14. 51bf200 Reduce number of RTPVideoSender::SendVideo parameters by Danil Chapovalov · 5 years ago
  15. 28214cd Fix handling of large packets in RtxReceiveStream by Niels Möller · 5 years ago
  16. dc34a25 Adds RTPSenderVideo::Config struct with red/ulpfec config by Erik Språng · 5 years ago
  17. b9bfe65 Delete VCMEncodedFrame::VerifyAndAllocate by Niels Möller · 5 years ago
  18. 6cf554e Reduces locking in RtpSenderVideo. by Erik Språng · 5 years ago
  19. 44db436 Propagate task queue to create test::DirectTransport by TaskQueueBase interface by Danil Chapovalov · 5 years ago
  20. 538ca57 Converts const methods in BitrateAllocator to non-member functions. by Sebastian Jansson · 5 years ago
  21. f34116e Replacing bandwidth adaptation trial with stable target in Opus encoder. by Sebastian Jansson · 5 years ago
  22. 40de3cc Propagating TargetRate struct to BitrateAllocator. by Sebastian Jansson · 5 years ago
  23. 0a5ed89 Adds remote estimates to rtc event log. by Sebastian Jansson · 5 years ago
  24. 93b1ea2 Using struct for bitrate allocation limits. by Sebastian Jansson · 5 years ago
  25. 53227cc Remove webrtc::MinPositive from api/. by Mirko Bonadei · 5 years ago
  26. 738bfa7 Remove api/bitrate_constraints.h. by Mirko Bonadei · 5 years ago
  27. 4d461ba Reusing MediaStreamAllocationConfig struct in ObserverConfig. by Sebastian Jansson · 5 years ago
  28. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  29. 7bf7a42 Delete flag VideoReceiveStream::Config::Rtp::remb by Niels Möller · 5 years ago
  30. eaaaf41 Introduce api/crypto/BUILD.gn. by Mirko Bonadei · 5 years ago
  31. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 5 years ago
  32. fcfeefe Move rtc_error.{h,cc} to its own build target. by Mirko Bonadei · 5 years ago
  33. 0987273 Add option to enable retransmission for all temporal layers in the constructor for rtp_sender_video. by Andrei Dumitru · 5 years ago
  34. cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 5 years ago
  35. 6516f76 Deprecate SingleThreadedTaskQueueForTesting class. by Yves Gerey · 5 years ago
  36. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 5 years ago
  37. f13df86 Delete audio methods SignalNetworkState by Niels Möller · 5 years ago
  38. 55dd72c Remove lock for process thread pointer from PacedSender. by Tommi · 5 years ago
  39. 25eb47c Make the RtpHeaderParserImpl available to tests and tools only. by Tommi · 5 years ago
  40. b4a6128 Delete unneeded dependencies on libjingle_peerconnection_api by Niels Möller · 5 years ago
  41. 6dcd4dc New target for api/rtp_parameters.h and api/media_types.h. by Niels Möller · 5 years ago
  42. fac7e31 Removes TransportSequenceNumberAllocator by Erik Språng · 5 years ago
  43. 0c141c5 Fix frames dropped statistics by Johannes Kron · 5 years ago
  44. d77cc24 New const method StreamStatistician::GetStats by Niels Möller · 5 years ago
  45. 4e615d5 Wire the stable target bitrate from GoogCC to the BitrateAllocator by Florent Castelli · 5 years ago
  46. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  47. 54d5d2c Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc by Erik Språng · 5 years ago
  48. 6b117a5 Make the callbacks to PollStats for RampUp* tests more regular. by Tommi · 5 years ago
  49. 93f5189 Remove some usage of RtpRtcp::SetSSRC() by Erik Språng · 5 years ago
  50. 6950b30 Fix thread naming in Call constructor by Erik Språng · 5 years ago
  51. c14b233 Disable the most flaky tests on iOS. by Sami Kalliomäki · 5 years ago
  52. eea605d Make fake network degradation work also for sent audio by Erik Språng · 5 years ago
  53. 48b48e5 Enable thread check in Call::GetStats(). by Tommi · 5 years ago
  54. 78a7138 Remove MediaTransport from Call. by Tommi · 5 years ago
  55. 891d393 Call Call::GetStats() from the correct thread in ProbingEndToEndTest. by Tommi · 5 years ago
  56. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 5 years ago
  57. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 5 years ago
  58. c24a5b1 Fix CallPerfTests to call Call::GetStats() from the right thread. by Tommi · 5 years ago
  59. c648819 DegradedCall: fake network using TaskQueue instead of ProcessThread by Erik Språng · 5 years ago
  60. 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
  61. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 5 years ago
  62. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 5 years ago
  63. 425d6aa Add RtpPacketPacer interface for pacer control by Erik Språng · 5 years ago
  64. 463d44a Don't crash when simulcast layer count is different from RID count by Florent Castelli · 5 years ago
  65. aa59eca Move RtpPacketSender and merge it with RtpPacketPacer. by Erik Språng · 5 years ago
  66. e1795f4 Adds remote estimate RTCP packet. by Sebastian Jansson · 5 years ago
  67. 2ab97f6 Migrate WebRTC test infra to ABSL_FLAG. by Mirko Bonadei · 5 years ago
  68. 0182a03 Reland "Remove the injectable bitrate allocation strategy API." by Jonas Olsson · 5 years ago
  69. e95b57c Revert "Remove the injectable bitrate allocation strategy API." by Mirko Bonadei · 5 years ago
  70. 80cb3f6 Remove the injectable bitrate allocation strategy API. by Jonas Olsson · 5 years ago
  71. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  72. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
  73. 53d45ba Make TaskQueueFactory required construction parameter for Call by Danil Chapovalov · 5 years ago
  74. bfd343b Add totalDecodeTime to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
  75. 67daf71 Implement RtpVideoSender::SetFecAllowed() by Elad Alon · 5 years ago
  76. 8f01c4e Define FecControllerOverride and plumb it down to VideoEncoder by Elad Alon · 5 years ago
  77. 4ba04b7 Delete RtcEventLogFactory factory as now unused by Danil Chapovalov · 5 years ago
  78. 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 5 years ago
  79. 59b8654 Switch from RtpPacketSender to RtpPacketPacer interface usage. by Erik Språng · 5 years ago
  80. 4d504c7 New interface EncodedImageBufferInterface, replacing use of CopyOnWriteBuffer by Niels Möller · 5 years ago
  81. 3d61ab1 Adds send time to ReceivedPacket struct. by Sebastian Jansson · 5 years ago
  82. 8aba8fe Reland "Populate the GFD-00 for H264 and generic codecs." by philipel · 5 years ago
  83. 607a6f1 Moves conversion to ReceivedPacket from RtpPacketReceived to Call. by Sebastian Jansson · 5 years ago
  84. 04e129a Revert "Populate the GFD-00 for H264 and generic codecs." by Philip Eliasson · 5 years ago
  85. d3c6f9c Populate the GFD-00 for H264 and generic codecs. by philipel · 5 years ago
  86. 08fa953 Reland "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" by Danil Chapovalov · 5 years ago
  87. 370f93a Reland "Inform VideoEncoder of negotiated capabilities" by Elad Alon · 5 years ago
  88. fd5166c Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" by Philip Eliasson · 5 years ago
  89. fc96135 Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory by Danil Chapovalov · 5 years ago
  90. 49d661a Revert "Inform VideoEncoder of negotiated capabilities" by Philip Eliasson · 5 years ago
  91. 11dfff0 Inform VideoEncoder of negotiated capabilities by Elad Alon · 5 years ago
  92. 4ad51d8 Removes SendSideCongestionController. by Sebastian Jansson · 5 years ago
  93. cf41eb1 Reland "Cleanup of video packet overhead calculation." by Sebastian Jansson · 5 years ago
  94. 5d24b16 Prepare for splitting the api/video:video_frames build rule. by Chen Xing · 5 years ago
  95. 27a7e9f Remove myself from OWNERS in a few places. by Fredrik Solenberg · 5 years ago
  96. f6c914a Revert "Reland "Cleanup of video packet overhead calculation."" by Sebastian Jansson · 5 years ago
  97. 35d4e43 Reland "Cleanup of video packet overhead calculation." by Sebastian Jansson · 5 years ago
  98. b64af4b Add retransmission_allowed flag to encoder output by Elad Alon · 5 years ago
  99. 845c6aa Add support for early loss detection using transport feedback. by Erik Språng · 5 years ago
  100. 6737841 Add jitterBufferDelay and jitterBufferEmittedCount stats for video by Guido Urdaneta · 5 years ago