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webrtc / src / 2a615fc760b112d1cf6011b0e96ea936fa169dea / webrtc / modules / rtp_rtcp / source / rtp_sender_video.cc
  1. 2a615fc Reduce taking locks in RTPSenderVideo::SendVideo by danilchap · 8 years ago
  2. 917a4ee Replace SequencedTaskChecker in RTPSenderVideo by kthelgason · 8 years ago
  3. dbdb3f1 Wire up FlexfecSender in RTPSender and add unit tests. by brandtr · 8 years ago
  4. 131bc49 Wire up FlexfecSender in RTPSenderVideo. by brandtr · 8 years ago
  5. 1743a19 Simplify SetFecParameters signature. by brandtr · 8 years ago
  6. f1bb476 Simplify {,Set}UlpfecConfig interface. by brandtr · 8 years ago
  7. d804895 Rename {,Set}GenericFECStatus to {,Set}UlpfecConfig. by brandtr · 8 years ago
  8. 869e7cd Rename ProducerFec to UlpfecGenerator. by brandtr · 8 years ago
  9. c1600c5 Follow standard sending CVO rtp header extension by danilchap · 8 years ago
  10. cc34833 Remove now unused code in RtpHeaderExtensionMap by danilchap · 8 years ago
  11. 3821399 Centralize deactivation of Unequal Protection. by Rasmus Brandt · 8 years ago
  12. c07ebb3 Simplify public interface of ProducerFec. by Rasmus Brandt · 8 years ago
  13. 7411061 Use RtpPacketToSend in RtpSenderVideo. by danilchap · 8 years ago
  14. 6631e8a Minor fixes in FEC and RtpSender{,Video} by brandtr · 9 years ago
  15. e5b4141 Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData by danilchap · 9 years ago
  16. 74811e5 Style updates to ProducerFec/FecReceiver. by brandtr · 9 years ago
  17. 5fb291a Remove RTPSenderInterface by danilchap · 9 years ago
  18. 525df3f Add EncodedImageCallback::OnEncodedImage(). by Sergey Ulanov · 9 years ago
  19. 51db4dd Revert of Add EncodedImageCallback::OnEncodedImage(). (patchset #14 id:300001 of https://codereview.chromium.org/2089773002/ ) by sergeyu · 9 years ago
  20. 4c7f4cd Add EncodedImageCallback::OnEncodedImage(). by Sergey Ulanov · 9 years ago
  21. ac4dc2c Revert of Add EncodedImageCallback::OnEncodedImage(). (patchset #13 id:280001 of https://codereview.webrtc.org/2089773002/ ) by sergeyu · 9 years ago
  22. ad34dbe Add EncodedImageCallback::OnEncodedImage(). by Sergey Ulanov · 9 years ago
  23. 32cd2c4 Fix issues with RestartingSendStreamPreservesRtpStatesWithRtx by danilchap · 9 years ago
  24. ec4f068 Style cleanups in RtpSender. by Sergey Ulanov · 9 years ago
  25. cd349d9 Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ ) by sprang · 9 years ago
  26. a49f110 Revert of Reland Issue 2061423003: Refactor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131313002/ ) by aluebs · 9 years ago
  27. 05ce4ae Reland Issue 2061423003: Refactor NACK bitrate allocation by Erik Språng · 9 years ago
  28. e5dd441 Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ ) by sprang · 9 years ago
  29. 5fc59e8 Refactor NACK bitrate allocation by Erik Språng · 9 years ago
  30. 6b4b5f3 Add sender controlled playout delay limits by isheriff · 9 years ago
  31. fc715f5 DCHECK that the red payload type doesn't have invalid values when FEC is enabled. by stefan · 9 years ago
  32. 8f4c77f Always send RED headers if configured. by stefan · 9 years ago
  33. 84be511 Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/ by kwiberg · 9 years ago
  34. 52d4e6b Revert of Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/ (patchset #1 id:40001 of https://codereview.webrtc.org/1921233002/ ) by terelius · 9 years ago
  35. 2c27a06 Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/ by kwiberg · 9 years ago
  36. 7c9426c Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module by danilchap · 9 years ago
  37. 98bb664 Added log messages for some important call setup events: by skvlad · 9 years ago
  38. 1e80ce4 webrtc::RtpPacket name freed for better RtpPacket by Danil Chapovalov · 9 years ago
  39. 9d0c432 Remove video-codec max bitrate from TMMBN. by Peter Boström · 9 years ago
  40. c0ae305 Fix null-pointer dereference in RTPSenderVideo. by Peter Boström · 9 years ago
  41. f6975f4 [rtp_rtcp] Lint errors cleaned from rtp_utility by danilchap · 9 years ago
  42. 6db6cdc [rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs by danilchap · 9 years ago
  43. 4654d20 Add test which verifies that the RTP header extensions are set correctly for FEC packets. by Stefan Holmer · 9 years ago
  44. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  45. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  46. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  47. ebc0b4e Use webrtc/base/logging.h for rtp_rtcp. by Peter Boström · 9 years ago
  48. e4f9650 Remove system_wrappers/interface/trace_event.h by tommi · 9 years ago
  49. e23e737 Disable pacer disabling. by Peter Boström · 9 years ago
  50. ebbf8a8 Make sure rtp_rtcp module doesn't directly reference anything in the pacer module, and remove build dependencies on it. by sprang · 9 years ago
  51. 586b19b Enable probing with repeated payload packets by default. by Stefan Holmer · 10 years ago
  52. 91d6ede Add RTC_ prefix to (D)CHECKs and related macros. by henrikg · 10 years ago
  53. a9455ab Integration of VP9 packetization. by asapersson · 10 years ago
  54. fcf54bd Reland "Avoid critsect for protection- and qm setting callbacks in VideoSender." by mflodman · 10 years ago
  55. 64c1e8c Enable CVO by default through webrtc pipeline. by Guo-wei Shieh · 10 years ago
  56. 31331cf Revert "Enable CVO by default through webrtc pipeline." by Minyue · 10 years ago
  57. 1b1c15c Enable CVO by default through webrtc pipeline. by Guo-wei Shieh · 10 years ago
  58. 0828a0c Revert "Avoid critsect for protection- and qm setting callbacks in VideoSender." by mflodman · 10 years ago
  59. 903c0f2 Avoid critsect for protection- and qm setting callbacks in VideoSender. by mflodman · 10 years ago
  60. 779c3d1 Use ByteReader/ByteWriter instead of rtputility and manual shift/add. by sprang@webrtc.org · 10 years ago
  61. 4536289 Add CVO support to RTP sender side. by guoweis@webrtc.org · 10 years ago
  62. 00b8f6b Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 10 years ago
  63. 0200f70 Set webrtc_rtp category to be disabled by default. by sprang@webrtc.org · 10 years ago
  64. 7c4d20f Remove potential deadlock in RTPSenderAudio. by pbos@webrtc.org · 10 years ago
  65. 4591fbd Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 10 years ago
  66. b5e6bfc Make RTPSender/RTPReceiver generic. by pbos@webrtc.org · 11 years ago
  67. 84b9e1e Fix for retransmission. Base layer packets were not retransmitted. by asapersson@webrtc.org · 11 years ago
  68. 2ec5606 Add H.264 packetization. by stefan@webrtc.org · 11 years ago
  69. 9e1acc8 Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 . by tommi@webrtc.org · 11 years ago
  70. 62bafae Some refactoring inside rtp_rtcp/. by pbos@webrtc.org · 11 years ago
  71. b9f5453 Add boilerplate code for H.264. by stefan@webrtc.org · 11 years ago
  72. dc80bae Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 11 years ago
  73. 346094c Incorrect overhead calculation when using FEC + RTP extension headers. by sprang@webrtc.org · 11 years ago
  74. 6811b6e Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  75. 096e8d9 Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  76. 2656cf9 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  77. 822fbd8 Update talk to 50918584. by wu@webrtc.org · 12 years ago
  78. aa4d96a Revert r4301 by tnakamura@webrtc.org · 12 years ago
  79. 1a7b9b9 Cleanup WebRTC tracing by hclam@chromium.org · 12 years ago
  80. 66b2e5c Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 12 years ago
  81. d900e8b Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 12 years ago
  82. 8ccb9f9 Fixes some pacer/padding issues found while testing. by stefan@webrtc.org · 12 years ago
  83. 508a84b Wire up pacer-based padding. by stefan@webrtc.org · 12 years ago
  84. a048d7c Include files from webrtc/.. paths in rtp_rtcp/ by pbos@webrtc.org · 12 years ago
  85. 806dc3b More trace events by hclam@chromium.org · 12 years ago
  86. 2f44673 WebRtc_Word32 => int32_t for rtp_rtcp/ by pbos@webrtc.org · 12 years ago
  87. b5bf54c Permit arbitrary payload names for kVideoCodecGeneric. by pbos@webrtc.org · 12 years ago
  88. bfacda6 Add interface to signal a network down event. by stefan@webrtc.org · 12 years ago
  89. 8911ce4 Generic video-codec support. by pbos@webrtc.org · 12 years ago
  90. 20ed36d Break out RtpClock to system_wrappers and make it more generic. by stefan@webrtc.org · 12 years ago
  91. 14b43be Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago[Renamed from src/modules/rtp_rtcp/source/rtp_sender_video.cc]
  92. 8639fd9 Use correct rtp header size for FEC packets. by marpan@webrtc.org · 13 years ago
  93. 71707aa Add the FEC mask type to FecProtectionParams and set the mask type in the VCM. by marpan@webrtc.org · 13 years ago
  94. ddfdfed Pass capture time (wallclock) to the RTP sender to compute transmission offset by stefan@webrtc.org · 13 years ago
  95. 899baa8 Temporarily disable first partition packet counting to avoid a bug in ProducerFec which doesn't properly handle important packets. by marpan@webrtc.org · 13 years ago
  96. 2853dde Refactor the internal API to the rtp/rtcp module. by pwestin@webrtc.org · 13 years ago
  97. 3c383ab Revert 2211 - Refactor the internal API to the rtp/rtcp module. by turaj@webrtc.org · 13 years ago
  98. 0774838 Refactor the internal API to the rtp/rtcp module. by pwestin@webrtc.org · 13 years ago
  99. b1fbf01 Added timestamp logs, i.e. only tracing. by mflodman@webrtc.org · 13 years ago
  100. c35f5ce Enable multi-frame FEC by default for temporal layers <= 2. For two temporal layers we currently only protect the base layer. by stefan@webrtc.org · 13 years ago