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gerrit-public.fairphone.software / platform / external / webrtc / 18f26d143411710fb64544246ecf0d30bc1d4e82 / video / rtp_video_stream_receiver.h
  1. a194e58 Move sequence_number_utils.h to rtc_base/ by Bjorn Terelius · 7 years ago
  2. 22ec952 Delete in_order argument to RtpReceiver::IncomingRtpPacket by Niels Möller · 7 years ago
  3. c62f6c7 RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs by Karl Wiberg · 7 years ago
  4. b0573bc Reorganize config of RTP header extensions for video receive streams. by Niels Möller · 7 years ago
  5. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  6. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  7. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/video/rtp_video_stream_receiver.h]
  8. ca5706d Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ ) by nisse · 7 years ago
  9. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
  10. 8e7eee0 Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ ) by nisse · 7 years ago
  11. 35713ea Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ ) by nisse · 7 years ago
  12. d4fac69 Unwrap picture ids in the RtpFrameReferencerFinder. by philipel · 7 years ago
  13. 3c39c01 Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ ) by nisse · 7 years ago
  14. 5c0f6c6 Use RtxReceiveStream. by nisse · 7 years ago
  15. 8b07305 Eliminate RtpVideoStreamReceiver::receive_cs_ in favor of using a SequencedTaskChecker by eladalon · 7 years ago
  16. 440b6d9 Move video send/receive stream headers to webrtc/call. by aleloi · 7 years ago
  17. c0d481a Protected streams report RTP messages directly to the FlexFec streams by eladalon · 7 years ago
  18. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  19. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  20. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  21. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 8 years ago
  22. b4ab381 Use the configured remote ssrc instead of relying on the first received packet RtpStreamReceiver. by stefan · 8 years ago
  23. b1f2ff9 Rename class RtpStreamReceiver --> RtpVideoStreamReceiver. by nisse · 8 years ago[Renamed (91%) from webrtc/video/rtp_stream_receiver.h]
  24. 30e8931 Delete RtpData::OnRecoveredPacket, use RecoveredPacketReceiver instead. by nisse · 8 years ago
  25. 3184f8e Dont request keyframes if the stream is inactive or if we are currently receiving a keyframe. by philipel · 8 years ago
  26. 2c53b13 Request keyframe if the first received frame is not a keyframe. by philipel · 8 years ago
  27. 0584331 Delete VieRemb class, move functionality to PacketRouter. by nisse · 8 years ago
  28. cd386eb Delete support for sending RTCP RPSI and SLI messages. by nisse · 8 years ago
  29. a45102f Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ ) by philipel · 8 years ago
  30. 38cc1d6 Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  31. e525d6a Revert Make the new jitter buffer the default jitter buffer. by stefan · 8 years ago
  32. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  33. e5bd702 Reland of Make the new jitter buffer the default jitter buffer. (patchset #2 id:260001 of https://codereview.chromium.org/2656983002/ ) by philipel · 8 years ago
  34. 27378f3 Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:290001 of https://codereview.chromium.org/2652043005/ ) by philipel · 8 years ago
  35. 09d6ef0 Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2638423003/ ) by philipel · 8 years ago
  36. bfb11b2 Call RtpStreamReceiver.AddReceiveCodec() with codec_params. by johan · 8 years ago
  37. 15389c0 Drop pacer and retransmission_rate_limiter from RtpStreamReceiver constructor. by nisse · 8 years ago
  38. 04926b8 Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ ) by kjellander · 8 years ago
  39. f20dd00 Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ ) by philipel · 8 years ago
  40. c08c191 Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ ) by philipel · 8 years ago
  41. 0f0763d Make the new jitter buffer the default jitter buffer. by philipel · 8 years ago
  42. 022b54e Wire up H264 fmtp sprop-parameter-sets with H264SpsPpsTracker. by philipel · 8 years ago
  43. 07e276c Rename RtpStreamReceiver::SetCodec() to ::AddCodec(). by johan · 8 years ago
  44. f7c6d72 Rename RtpStreamReceiver::IsFecEnabled to RtpStreamReceiver::IsUlpfecEnabled. by brandtr · 8 years ago
  45. fd5a20f New jitter buffer experiment. by philipel · 8 years ago
  46. e6f98c7 Remove RED/RTX workaround from sender/receiver and VideoEngine2. by brandtr · 8 years ago
  47. d55c3f6 Rename FecReceiver to UlpfecReceiver. by brandtr · 8 years ago
  48. 7056be9 Delete old video defines in engine config. by mflodman · 8 years ago
  49. 737336d Add NACK rate throttling for audio channels. by Erik Språng · 8 years ago
  50. ec4f068 Style cleanups in RtpSender. by Sergey Ulanov · 8 years ago
  51. 0208322 GN: Add video_engine_tests by Peter Boström · 9 years ago
  52. 733b547 Movable support for VideoReceiveStream::Config and avoid copies. by Tommi · 9 years ago
  53. dc7d0d2 Move, almost, all receive side references to RTP to RtpStreamReceiver. by mflodman · 9 years ago
  54. cfc8e3b Removed all RTP dependencies from ViEChannel and renamed class. by mflodman · 9 years ago
  55. fa66659 Rename ViEReceiver and move ownership to VideoReceiveStream. by mflodman · 9 years ago[Renamed (84%) from webrtc/video/vie_receiver.h]
  56. 4485ffb #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 9 years ago
  57. c0e58a3 Move receive RtpRtcp ownership from ViEChannel to ViEReceiver. by mflodman · 9 years ago
  58. 0b25072 Use vcm::VideoReceiver on the receive side. by Peter Boström · 9 years ago
  59. 83d0910 Move Ownership of RtpModules to VideoSendStream from VieChannel and remove use of vie_channel and vie_receiver from video_send_stream. by Per · 9 years ago
  60. 4fa7eca Remove add/removal of ViEReceiver RTP modules. by Peter Boström · 9 years ago
  61. 27f982b Replace scoped_ptr with unique_ptr in webrtc/video/ by kwiberg · 9 years ago
  62. 9c01725 Simplify registration of RTP-header extensions. by Peter Boström · 9 years ago
  63. 029e220 Removes use of DeRegister Rtp Header Extension for video by Danil Chapovalov · 9 years ago
  64. 59c634b Re-add RemoteBitrateEstimator::GetStats. by Stefan Holmer · 9 years ago
  65. d1d66ba Remove ViEChannel calls for VideoReceiveStream. by Peter Boström · 9 years ago
  66. 97888bd Swap use of CriticalSectionWrapper for rtc::CriticalSection in webrtc/video. by Tommi · 9 years ago
  67. 7623ce4 Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ ) by Peter Boström · 9 years ago[Renamed (96%) from webrtc/video_engine/vie_receiver.h]
  68. 8237abf Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) by kjellander · 9 years ago[Renamed (96%) from webrtc/video/vie_receiver.h]
  69. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago[Renamed (96%) from webrtc/video_engine/vie_receiver.h]
  70. 0fcaf99 Enable cpplint for webrtc/video_engine by kjellander@webrtc.org · 9 years ago
  71. c4a1c37 Removed vie_defines.h by mflodman · 9 years ago
  72. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  73. 65220a7 Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists. by noahric · 9 years ago
  74. ac547a6 Remove channel ids from various interfaces. by Peter Boström · 9 years ago
  75. 867fb52 Add support for transport wide sequence numbers by sprang · 9 years ago
  76. d6f1a38 Remove ViEChannel simulcast lock. by Peter Boström · 9 years ago
  77. c3f4dbc Remove rtp_rtcp/ dump functionality. by Peter Boström · 10 years ago
  78. 300eeb6 Remove VideoEngine interfaces. by Peter Boström · 10 years ago
  79. e62202f Support handling multiple RTX but only generate SDP with RTX associated with VP8. by Shao Changbin · 10 years ago
  80. 6cff9cf Revert "Remove simulcast modules from ViEReceiver." by Peter Boström · 10 years ago
  81. 14a97f0 Remove simulcast modules from ViEReceiver. by Peter Boström · 10 years ago
  82. fdd1057 Add CVO support to Vie layer. by guoweis@webrtc.org · 10 years ago
  83. 14665ff Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro by kjellander@webrtc.org · 10 years ago
  84. 058b1f1 Remove GetReceiveBandwidthEstimatorStats. by pbos@webrtc.org · 10 years ago
  85. 00b8f6b Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 10 years ago
  86. 37c0559 Notify jitter buffer about received FEC packets (to avoid sending NACK request for these packets). by asapersson@webrtc.org · 10 years ago
  87. 273fbbb Update StreamDataCounter with FEC bytes. by asapersson@webrtc.org · 10 years ago
  88. 0800db7 Add percentage of fec packets and recovered media packets to histogram stats: by asapersson@webrtc.org · 10 years ago
  89. 8f27fcc Revert 8028 "Support associated payload type when registering Rt..." by andrew@webrtc.org · 10 years ago
  90. 2a16964 Support associated payload type when registering Rtx payload type. by pbos@webrtc.org · 10 years ago
  91. d952c40 Add receive bitrates to histogram stats: by asapersson@webrtc.org · 10 years ago
  92. 4591fbd Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 10 years ago
  93. eb24b04 Add periodic logging of received RTP headers and estimated clock offsets for e2e delay. by stefan@webrtc.org · 10 years ago
  94. 6071b06 Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such. by stefan@webrtc.org · 10 years ago
  95. ef92755 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. by stefan@webrtc.org · 11 years ago
  96. 88abf11 Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe. by wu@webrtc.org · 11 years ago
  97. 4e2806d Remove WEBRTC_TRACE uses in video_engine/ by pbos@webrtc.org · 11 years ago
  98. 66773a0 Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine. by wu@webrtc.org · 11 years ago
  99. 24bd364 Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels. by stefan@webrtc.org · 11 years ago
  100. cd70119 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 11 years ago