Google Git
Sign in
webrtc / src / 1380e266ff48be9718ce0867cfd65058cb09c5fc / webrtc / common_types.h
  1. dce40cf Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 10 years ago
  2. 4cee419 Separating voice activity flag from audio level in RtpHeaderExtension. by Minyue · 10 years ago
  3. a9455ab Integration of VP9 packetization. by asapersson · 10 years ago
  4. 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
  5. c62642c Make the BWE threshold adaptive. by stefan · 10 years ago
  6. 3093390 Parsing of transport wide sequence number rtp extension header. by sprang@webrtc.org · 10 years ago
  7. 4536289 Add CVO support to RTP sender side. by guoweis@webrtc.org · 10 years ago
  8. 14665ff Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro by kjellander@webrtc.org · 10 years ago
  9. e8f50df Remove avi recorder and corresponding enable_video flags. by andresp@webrtc.org · 10 years ago
  10. ac2d27d Fix style violations in common_types.h and config.h by kwiberg@webrtc.org · 10 years ago
  11. 1d0fa5d Add RtcpPacketTypeCounter stats to new API. by pbos@webrtc.org · 10 years ago
  12. c0bd7be Adding two new stats to VoiceReceiverInfo by minyue@webrtc.org · 10 years ago
  13. 4414939 Add method for incrementing RtpPacketCounter. Removes duplicate code. by asapersson@webrtc.org · 10 years ago
  14. 273fbbb Update StreamDataCounter with FEC bytes. by asapersson@webrtc.org · 10 years ago
  15. cfd82df Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. by asapersson@webrtc.org · 10 years ago
  16. ce4e9a3 Refactor some receive-side stats. by pbos@webrtc.org · 10 years ago
  17. d08d389 Add field to counters for when first rtp/rtcp packet is sent/received. by asapersson@webrtc.org · 10 years ago
  18. 97d0489 Add video send bitrates to histogram stats: by asapersson@webrtc.org · 10 years ago
  19. d952c40 Add receive bitrates to histogram stats: by asapersson@webrtc.org · 10 years ago
  20. 4591fbd Use size_t more consistently for packet/payload lengths. by pkasting@chromium.org · 10 years ago
  21. 0bae1fa Wire up bandwidth stats to the new API and webrtcvideoengine2. by stefan@webrtc.org · 10 years ago
  22. 5b88317 Add VP9 codec to VCM and vie_auto_test. by marpan@webrtc.org · 10 years ago
  23. 2dd3134 Add stats for duplicate sent and received NACK requests. by asapersson@webrtc.org · 10 years ago
  24. b1dac33 Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..." by henrike@webrtc.org · 10 years ago
  25. 573c78e Add VP9 codec to VCM and vie_auto_test. by marpan@webrtc.org · 10 years ago
  26. 8768f16 Fix comments in common_types.h by henrik.lundin@webrtc.org · 10 years ago
  27. 168f23f Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 11 years ago
  28. 72491b9 Count total bytes sent in RTPSender::Bytes(). by pbos@webrtc.org · 11 years ago
  29. b9f5453 Add boilerplate code for H.264. by stefan@webrtc.org · 11 years ago
  30. 1e92b0a Add ToString() to VideoSendStream::Config. by pbos@webrtc.org · 11 years ago
  31. 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
  32. 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
  33. 2a77082 Remove usage of webrtc trace in video processing modules. by asapersson@webrtc.org · 11 years ago
  34. 681d448 Removing VideoCodecDerived and moving methods inside VideoCodec. by mallinath@webrtc.org · 11 years ago
  35. 3c412b2 Add targetBitrate to VideoCodec struct. by pbos@webrtc.org · 11 years ago
  36. b1f5010 VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 11 years ago
  37. 0209e56 Adding operator== and != methods for CodecInst and VideoCodec structures. by mallinath@webrtc.org · 11 years ago
  38. f577ae9 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 11 years ago
  39. 8098e07 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 11 years ago
  40. a079233 Remove external encryption API for VoE. by solenberg@webrtc.org · 11 years ago
  41. 0931570 Wire up statistics in video receive stream of new API by sprang@webrtc.org · 11 years ago
  42. 39fcfd7 Remove empty VideoCodecGeneric struct. by pbos@webrtc.org · 11 years ago
  43. ccd4284 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  44. 5ab7567 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  45. 41e2615 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  46. 341e914 Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  47. 24301a6 Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  48. 6811b6e Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  49. a989080 Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  50. 2018269 Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  51. a129b6c Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  52. 096e8d9 Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  53. 2656cf9 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  54. ebad765 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
  55. a6ad6e5 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
  56. 71f055f Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
  57. 72964bd Make interface destructor virtual by sprang@webrtc.org · 11 years ago
  58. dc50aae Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
  59. fe5d36b Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well. by sprang@webrtc.org · 11 years ago
  60. eda189b Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 12 years ago
  61. 678cf29 webrtc/common_types.h: Document bitrate fields' units. by fischman@webrtc.org · 12 years ago
  62. 185bae4 Replace ExtraCodecOptions with new Config class that supports multiple settings at once. by andresp@webrtc.org · 12 years ago
  63. 77f6b21 Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..." by pbos@webrtc.org · 12 years ago
  64. 68e5a68 Revert 3933 "Remove traces of deprecated WebRtc_Word types." by pbos@webrtc.org · 12 years ago
  65. 265a5d2 Remove traces of deprecated WebRtc_Word types. by pbos@webrtc.org · 12 years ago
  66. b5eeaa9 Adding extra options to interact with external encoder/decoder. by andresp@webrtc.org · 12 years ago
  67. 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
  68. 94bc4cf Add min and target bitrate to VideoCodec. by marpan@webrtc.org · 12 years ago
  69. 8911ce4 Generic video-codec support. by pbos@webrtc.org · 12 years ago
  70. b7edd06 Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 12 years ago
  71. 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
  72. eb91792 Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings. by stefan@webrtc.org · 12 years ago
  73. e07c661 VP8: Making key frame interval a tunnable parameter by mikhal@webrtc.org · 12 years ago
  74. b8ba4d8 Add number of inserted samples to NetEq statistics. by roosa@google.com · 12 years ago
  75. b718619 Expose NetEq playout mode off through VoiceEngine. by roosa@google.com · 12 years ago
  76. 655d8f5 Add a kTraceTerseInfo level for non-verbose logging. by andrew@webrtc.org · 12 years ago
  77. 23ec30b Clean up TraceCallback::Print. by andrew@webrtc.org · 12 years ago
  78. 50419b0 Add libjingle-style stream-style logging. by andrew@webrtc.org · 12 years ago
  79. 14b43be Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago[Renamed from src/common_types.h]
  80. 21ab3ba Add external media hook for preprocessed audio. by andrew@webrtc.org · 12 years ago
  81. c58be0d Tune for faster ramp-up. by stefan@webrtc.org · 13 years ago
  82. 3387b88 Makes it possible to disable frame dropping in the VP8 codec. by henrike@webrtc.org · 13 years ago
  83. 88b8b0d Disable "new behavior: elements of array will be default initialized" warning. by andrew@webrtc.org · 13 years ago
  84. 3c28674 Makes it possible to disable automatic resizing. by henrike@webrtc.org · 13 years ago
  85. bd7aeba Expose a set of options to the OveruseDetector supporting experiments by astor@webrtc.org · 13 years ago
  86. dc257b5 Add option to configure error concealment and disable by default. by stefan@webrtc.org · 13 years ago
  87. c8e4886 Upgrade libvpx to 6b66c01 and enabling temporal denoising. by stefan@webrtc.org · 13 years ago
  88. 8012474 Use a const rather than macro for EcDefault. by andrew@webrtc.org · 13 years ago
  89. 8bfee84 Initial revision of a ViE fuzz test. The idea is to inject randomized RTP packets and see what the video engine does. by phoglund@webrtc.org · 13 years ago
  90. 5621057 Removing unused code. Review URL: https://webrtc-codereview.appspot.com/349008 by pwestin@webrtc.org · 13 years ago
  91. f75901f Resolves CID 10540: Copy into fixed size buffer (STRING_OVERFLOW). by henrika@webrtc.org · 13 years ago
  92. 053c799 Add minimum waiting time to NetEQ metrics by henrik.lundin@webrtc.org · 13 years ago
  93. d439870 Adding two new network metrics to NetEQ by henrik.lundin@webrtc.org · 13 years ago
  94. c00f91d Adding BGRA as a video type. by mikhal@webrtc.org · 13 years ago
  95. efd0a48 Add error resilient mode options to the VP8 specific VideoCodec struct. by stefan@webrtc.org · 13 years ago
  96. dbba1f9 Packet waiting-time statistics by henrik.lundin@webrtc.org · 13 years ago
  97. db221d2 Fixes to temporal layers, Henrika please review src/common_types.h by pwestin@webrtc.org · 13 years ago
  98. df10de4 Removing statistics API from NetEQ by henrik.lundin@webrtc.org · 13 years ago
  99. 1da1ce0 First implementation of simulcast, adds VP8 simulcast to video engine. by pwestin@webrtc.org · 13 years ago
  100. 470e71d by niklase@google.com · 14 years ago