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webrtc / src.git / 96b6b8336a543112573fd91a0cf13a3c1f9d83c9 / webrtc / common_types.h
  1. cd349d9 Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ ) by sprang · 9 years ago
  2. 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
  3. 05ce4ae Reland Issue 2061423003: Refactor NACK bitrate allocation by Erik Språng · 9 years ago
  4. e5dd441 Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ ) by sprang · 9 years ago
  5. 5fc59e8 Refactor NACK bitrate allocation by Erik Språng · 9 years ago
  6. 3abb764 Avoid unnecessary HW video encoder reconfiguration by skvlad · 9 years ago
  7. a6219cc FileWrapper[Impl] modifications and actually remove the "Impl" class. by tommi · 9 years ago
  8. 6b4b5f3 Add sender controlled playout delay limits by isheriff · 9 years ago
  9. 35151f3 Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket. by asapersson · 9 years ago
  10. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 9 years ago
  11. 1069cac Tune BWE to be a bit less sensitive to spurious delay events. by stefan · 9 years ago
  12. 22c2b48 Move RTP stats histograms from VieChannel to SendStatisticsProxy. by Erik Språng · 9 years ago
  13. 8b79b07 Move RTP module activation into PayloadRouter. by Peter Boström · 9 years ago
  14. 07fb9be Move RTCP histograms from vie_channel to video channel stats proxies. by sprang · 9 years ago
  15. 7b971e7 Remove extra_options from VideoCodec. by Peter Boström · 9 years ago
  16. 6955870 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  17. 2845a02 Remove unused enum RTPDirections. by terelius · 9 years ago
  18. 64c0a0a Revert of Make overuse estimator one dimensional. (patchset #5 id:80001 of https://codereview.webrtc.org/1376423002/ ) by stefan · 9 years ago
  19. 06e05a8 Make overuse estimator one dimensional. by Stefan Holmer · 9 years ago
  20. ce4aef1 Adding support for simulcast and spatial layers into VideoQualityTest by sprang · 9 years ago
  21. 49e196a Remove VideoFrameType aliases for FrameType. by Peter Boström · 9 years ago
  22. 22993e1 Unify FrameType and VideoFrameType. by pbos · 9 years ago
  23. 7a975f7 Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ ) by sprang · 9 years ago
  24. 87f83a9 Adding support for simulcast and spatial layers into VideoQualityTest by ivica · 9 years ago
  25. da903ea Unify newapi::RtcpMode and RTCPMethod. by pbos · 9 years ago
  26. 2d56668 Unify Transport and newapi::Transport interfaces. by pbos · 10 years ago
  27. ac547a6 Remove channel ids from various interfaces. by Peter Boström · 10 years ago
  28. f350720 VP9: Add automaticeResize to codec setting. by Marco · 10 years ago
  29. dce40cf Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 10 years ago
  30. 4cee419 Separating voice activity flag from audio level in RtpHeaderExtension. by Minyue · 10 years ago
  31. a9455ab Integration of VP9 packetization. by asapersson · 10 years ago
  32. 8647922 Revert the process noise co-variance of the bitrate over-use estimator to its value prior to r9545. by Stefan Holmer · 10 years ago
  33. c62642c Make the BWE threshold adaptive. by stefan · 10 years ago
  34. 3093390 Parsing of transport wide sequence number rtp extension header. by sprang@webrtc.org · 10 years ago
  35. 4536289 Add CVO support to RTP sender side. by guoweis@webrtc.org · 10 years ago
  36. 14665ff Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro by kjellander@webrtc.org · 10 years ago
  37. e8f50df Remove avi recorder and corresponding enable_video flags. by andresp@webrtc.org · 10 years ago
  38. ac2d27d Fix style violations in common_types.h and config.h by kwiberg@webrtc.org · 10 years ago
  39. 1d0fa5d Add RtcpPacketTypeCounter stats to new API. by pbos@webrtc.org · 10 years ago
  40. c0bd7be Adding two new stats to VoiceReceiverInfo by minyue@webrtc.org · 10 years ago
  41. 4414939 Add method for incrementing RtpPacketCounter. Removes duplicate code. by asapersson@webrtc.org · 10 years ago
  42. 273fbbb Update StreamDataCounter with FEC bytes. by asapersson@webrtc.org · 10 years ago
  43. cfd82df Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. by asapersson@webrtc.org · 10 years ago
  44. ce4e9a3 Refactor some receive-side stats. by pbos@webrtc.org · 10 years ago
  45. d08d389 Add field to counters for when first rtp/rtcp packet is sent/received. by asapersson@webrtc.org · 10 years ago
  46. 97d0489 Add video send bitrates to histogram stats: by asapersson@webrtc.org · 10 years ago
  47. d952c40 Add receive bitrates to histogram stats: by asapersson@webrtc.org · 10 years ago
  48. 4591fbd Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 10 years ago
  49. 0bae1fa Wire up bandwidth stats to the new API and webrtcvideoengine2. by stefan@webrtc.org · 10 years ago
  50. 5b88317 Add VP9 codec to VCM and vie_auto_test. by marpan@webrtc.org · 10 years ago
  51. 2dd3134 Add stats for duplicate sent and received NACK requests. by asapersson@webrtc.org · 10 years ago
  52. b1dac33 Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..." by henrike@webrtc.org · 10 years ago
  53. 573c78e Add VP9 codec to VCM and vie_auto_test. by marpan@webrtc.org · 10 years ago
  54. 8768f16 Fix comments in common_types.h by henrik.lundin@webrtc.org · 10 years ago
  55. 168f23f Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 11 years ago
  56. 72491b9 Count total bytes sent in RTPSender::Bytes(). by pbos@webrtc.org · 11 years ago
  57. b9f5453 Add boilerplate code for H.264. by stefan@webrtc.org · 11 years ago
  58. 1e92b0a Add ToString() to VideoSendStream::Config. by pbos@webrtc.org · 11 years ago
  59. 82d3cb6 Made common_types.h PacketTime declaration match https://code.google.com/p/webrtc/source/browse/trunk/talk/base/asyncpacketsocket.h#65 by henrike@webrtc.org · 11 years ago
  60. 6680348 Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 11 years ago
  61. 2a77082 Remove usage of webrtc trace in video processing modules. by asapersson@webrtc.org · 11 years ago
  62. 681d448 Removing VideoCodecDerived and moving methods inside VideoCodec. by mallinath@webrtc.org · 11 years ago
  63. 3c412b2 Add targetBitrate to VideoCodec struct. by pbos@webrtc.org · 11 years ago
  64. b1f5010 VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 11 years ago
  65. 0209e56 Adding operator== and != methods for CodecInst and VideoCodec structures. by mallinath@webrtc.org · 11 years ago
  66. f577ae9 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 11 years ago
  67. 8098e07 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 11 years ago
  68. a079233 Remove external encryption API for VoE. by solenberg@webrtc.org · 11 years ago
  69. 0931570 Wire up statistics in video receive stream of new API by sprang@webrtc.org · 11 years ago
  70. 39fcfd7 Remove empty VideoCodecGeneric struct. by pbos@webrtc.org · 11 years ago
  71. ccd4284 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  72. 5ab7567 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  73. 41e2615 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  74. 341e914 Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  75. 24301a6 Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  76. 6811b6e Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  77. a989080 Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  78. 2018269 Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  79. a129b6c Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  80. 096e8d9 Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  81. 2656cf9 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  82. ebad765 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
  83. a6ad6e5 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
  84. 71f055f Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
  85. 72964bd Make interface destructor virtual by sprang@webrtc.org · 11 years ago
  86. dc50aae Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
  87. fe5d36b Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well. by sprang@webrtc.org · 11 years ago
  88. eda189b Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 12 years ago
  89. 678cf29 webrtc/common_types.h: Document bitrate fields' units. by fischman@webrtc.org · 12 years ago
  90. 185bae4 Replace ExtraCodecOptions with new Config class that supports multiple settings at once. by andresp@webrtc.org · 12 years ago
  91. 77f6b21 Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..." by pbos@webrtc.org · 12 years ago
  92. 68e5a68 Revert 3933 "Remove traces of deprecated WebRtc_Word types." by pbos@webrtc.org · 12 years ago
  93. 265a5d2 Remove traces of deprecated WebRtc_Word types. by pbos@webrtc.org · 12 years ago
  94. b5eeaa9 Adding extra options to interact with external encoder/decoder. by andresp@webrtc.org · 12 years ago
  95. a442d4d Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. by solenberg@webrtc.org · 12 years ago
  96. 94bc4cf Add min and target bitrate to VideoCodec. by marpan@webrtc.org · 12 years ago
  97. 8911ce4 Generic video-codec support. by pbos@webrtc.org · 12 years ago
  98. b7edd06 Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 12 years ago
  99. 24045c5 None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise. by turaj@webrtc.org · 12 years ago
  100. eb91792 Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings. by stefan@webrtc.org · 12 years ago