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webrtc / src / webrtc / ae4b6ecf6f2b832e2c1750a081c4d35e1e565df9 / common_types.h
  1. 48be054 Avoid unnecessary HW video encoder reconfiguration by skvlad · 9 years ago
  2. a6d985c FileWrapper[Impl] modifications and actually remove the "Impl" class. by tommi · 9 years ago
  3. 00cc045 Add sender controlled playout delay limits by isheriff · 9 years ago
  4. 86a285e 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
  5. 47a40a3 Remove webrtc/stream.h and unutilized inheritance. by pbos · 9 years ago
  6. 84d41a3 Tune BWE to be a bit less sensitive to spurious delay events. by stefan · 9 years ago
  7. 01cff72 Move RTP stats histograms from VieChannel to SendStatisticsProxy. by Erik Språng · 9 years ago
  8. cd71dc6 Move RTP module activation into PayloadRouter. by Peter Boström · 9 years ago
  9. 22c956d Move RTCP histograms from vie_channel to video channel stats proxies. by sprang · 9 years ago
  10. d146002 Remove extra_options from VideoCodec. by Peter Boström · 9 years ago
  11. 80590d9 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  12. 8b9ae39 Remove unused enum RTPDirections. by terelius · 9 years ago
  13. 12764bc Revert of Make overuse estimator one dimensional. (patchset #5 id:80001 of https://codereview.webrtc.org/1376423002/ ) by stefan · 9 years ago
  14. 4123a7e Make overuse estimator one dimensional. by Stefan Holmer · 9 years ago
  15. 0ba16d1 Adding support for simulcast and spatial layers into VideoQualityTest by sprang · 9 years ago
  16. 9511e46 Remove VideoFrameType aliases for FrameType. by Peter Boström · 9 years ago
  17. b63d8ad Unify FrameType and VideoFrameType. by pbos · 9 years ago
  18. 2d8a1b5 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
  19. 3eea752 Adding support for simulcast and spatial layers into VideoQualityTest by ivica · 9 years ago
  20. ba01e95 Unify newapi::RtcpMode and RTCPMethod. by pbos · 9 years ago
  21. d5bdda3 Unify Transport and newapi::Transport interfaces. by pbos · 10 years ago
  22. 00a7fa4 Remove channel ids from various interfaces. by Peter Boström · 10 years ago
  23. e8b7e11 VP9: Add automaticeResize to codec setting. by Marco · 10 years ago
  24. a0ad248 Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 10 years ago
  25. da4c0f0 Separating voice activity flag from audio level in RtpHeaderExtension. by Minyue · 10 years ago
  26. 6240797 Integration of VP9 packetization. by asapersson · 10 years ago
  27. 847b12a Revert the process noise co-variance of the bitrate over-use estimator to its value prior to r9545. by Stefan Holmer · 10 years ago
  28. b2f1caf Make the BWE threshold adaptive. by stefan · 10 years ago
  29. 25ec20f Parsing of transport wide sequence number rtp extension header. by sprang@webrtc.org · 10 years ago
  30. 5f74fcf Add CVO support to RTP sender side. by guoweis@webrtc.org · 10 years ago
  31. 860ac53 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro by kjellander@webrtc.org · 10 years ago
  32. 51bd14d Remove avi recorder and corresponding enable_video flags. by andresp@webrtc.org · 10 years ago
  33. c4e2cd0 Fix style violations in common_types.h and config.h by kwiberg@webrtc.org · 10 years ago
  34. eb1c2b5 Add RtcpPacketTypeCounter stats to new API. by pbos@webrtc.org · 10 years ago
  35. 5f8d288 Adding two new stats to VoiceReceiverInfo by minyue@webrtc.org · 10 years ago
  36. 1a8794b Add method for incrementing RtpPacketCounter. Removes duplicate code. by asapersson@webrtc.org · 10 years ago
  37. 08c57f7 Update StreamDataCounter with FEC bytes. by asapersson@webrtc.org · 10 years ago
  38. 96763da Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. by asapersson@webrtc.org · 10 years ago
  39. 1a36f78 Refactor some receive-side stats. by pbos@webrtc.org · 10 years ago
  40. c76c553 Add field to counters for when first rtp/rtcp packet is sent/received. by asapersson@webrtc.org · 10 years ago
  41. 96568c2 Add video send bitrates to histogram stats: by asapersson@webrtc.org · 10 years ago
  42. 2755e82 Add receive bitrates to histogram stats: by asapersson@webrtc.org · 10 years ago
  43. 0ab923a Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 10 years ago
  44. 5232267 Wire up bandwidth stats to the new API and webrtcvideoengine2. by stefan@webrtc.org · 10 years ago
  45. 6637388 Add VP9 codec to VCM and vie_auto_test. by marpan@webrtc.org · 10 years ago
  46. 2ba45ee Add stats for duplicate sent and received NACK requests. by asapersson@webrtc.org · 10 years ago
  47. 63d5096 Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..." by henrike@webrtc.org · 10 years ago
  48. e172b11 Add VP9 codec to VCM and vie_auto_test. by marpan@webrtc.org · 10 years ago
  49. 6e7480a Fix comments in common_types.h by henrik.lundin@webrtc.org · 10 years ago
  50. 55b0f2e Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 11 years ago
  51. 9aa3497 Count total bytes sent in RTPSender::Bytes(). by pbos@webrtc.org · 11 years ago
  52. 2d4a80c Add boilerplate code for H.264. by stefan@webrtc.org · 11 years ago
  53. 7e68693 Add ToString() to VideoSendStream::Config. by pbos@webrtc.org · 11 years ago
  54. 93ae821 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
  55. 692224a 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
  56. 9402619 Remove usage of webrtc trace in video processing modules. by asapersson@webrtc.org · 11 years ago
  57. 2a0cbfc Removing VideoCodecDerived and moving methods inside VideoCodec. by mallinath@webrtc.org · 11 years ago
  58. 3d6910c Add targetBitrate to VideoCodec struct. by pbos@webrtc.org · 11 years ago
  59. fec6b6e VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 11 years ago
  60. 18c2945 Adding operator== and != methods for CodecInst and VideoCodec structures. by mallinath@webrtc.org · 11 years ago
  61. e2a7a77 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 11 years ago
  62. 4a15560 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 11 years ago
  63. a56c5b4 Remove external encryption API for VoE. by solenberg@webrtc.org · 11 years ago
  64. c8ab721 Wire up statistics in video receive stream of new API by sprang@webrtc.org · 11 years ago
  65. 84350a9 Remove empty VideoCodecGeneric struct. by pbos@webrtc.org · 11 years ago
  66. 49812e6 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  67. 46f7288 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  68. c5a5713 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  69. 3bbc91e Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  70. 79d6daf Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  71. b70db6d Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  72. efeb8ce Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  73. 12553ad Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  74. 984bee2 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  75. ffea4ce Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  76. 1430bc3 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  77. b113981 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
  78. 9b30fd3 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
  79. 5fdd10a Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
  80. 21dc10d Make interface destructor virtual by sprang@webrtc.org · 11 years ago
  81. 4673674 Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
  82. 2714c79 Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well. by sprang@webrtc.org · 11 years ago
  83. 5cf83f4 Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 12 years ago
  84. 5dcb4e3 webrtc/common_types.h: Document bitrate fields' units. by fischman@webrtc.org · 12 years ago
  85. ee6f8a2 Replace ExtraCodecOptions with new Config class that supports multiple settings at once. by andresp@webrtc.org · 12 years ago
  86. 52b2ee5 Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..." by pbos@webrtc.org · 12 years ago
  87. 2d6f0df Revert 3933 "Remove traces of deprecated WebRtc_Word types." by pbos@webrtc.org · 12 years ago
  88. e422d12 Remove traces of deprecated WebRtc_Word types. by pbos@webrtc.org · 12 years ago
  89. f292306 Adding extra options to interact with external encoder/decoder. by andresp@webrtc.org · 12 years ago
  90. fa2dd22 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
  91. e1198e6 Add min and target bitrate to VideoCodec. by marpan@webrtc.org · 12 years ago
  92. e3339fc Generic video-codec support. by pbos@webrtc.org · 12 years ago
  93. 15a03fd Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 12 years ago
  94. 8665399 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
  95. f4d3788 Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings. by stefan@webrtc.org · 12 years ago
  96. ca0e88a VP8: Making key frame interval a tunnable parameter by mikhal@webrtc.org · 12 years ago
  97. 0049a76 Add number of inserted samples to NetEq statistics. by roosa@google.com · 12 years ago
  98. 90d333e Expose NetEq playout mode off through VoiceEngine. by roosa@google.com · 12 years ago
  99. bc687c5 Add a kTraceTerseInfo level for non-verbose logging. by andrew@webrtc.org · 12 years ago
  100. d75680a Clean up TraceCallback::Print. by andrew@webrtc.org · 12 years ago