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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 28e9b66210667790f4ce46f5b0bfca1ebd32b212 / video_engine
  1. 0d47fe1 Raise kViEMaxNumberOfChannels from 32 to 64 by wu@webrtc.org · 10 years ago
  2. 569487d Updated WebRTC version to 3.53 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
  3. 60f1422 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 · 10 years ago
  4. 73c2412 Remove timestamp_extrapolator's dependency to Clock and vcm defines. by wu@webrtc.org · 10 years ago
  5. 1a9e6ac Pointers were not dereferenced in GetRtpStatistics. by asapersson@webrtc.org · 10 years ago
  6. 42fe6b3 Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels. by stefan@webrtc.org · 10 years ago
  7. c6cfc5c Upping start bitrate to min, if set to a lower value i SetSendCodec. by mflodman@webrtc.org · 10 years ago
  8. 73e1a8b Disable flaky RunsRtpRtcpTestWIthoutErrors. by pbos@webrtc.org · 10 years ago
  9. b0079ed Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
  10. 0a5fd54 Casting char to int in logs. by asapersson@webrtc.org · 10 years ago
  11. b991cd0 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
  12. a1626fe Propagate capture ntp timestamp from rtp to renderer. by wu@webrtc.org · 10 years ago
  13. 6b1114a Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
  14. b18bff5 Cleaned up logging in video_coding. by stefan@webrtc.org · 10 years ago
  15. 4b0cd7f Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 10 years ago
  16. ff46b81 Updated WebRTC version to 3.52 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
  17. 44c9b9a Log Fixit for parts of video_engine folder. by mflodman@webrtc.org · 10 years ago
  18. 6c57efd Re-submit: rev5775 by andresp@webrtc.org · 10 years ago
  19. 37f807f Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams. by solenberg@webrtc.org · 10 years ago
  20. 765ea72 Revert 5775 "Modify bitrate controller to update bitrate based o..." by andrew@webrtc.org · 10 years ago
  21. f50914a Removing VideoCodecDerived and moving methods inside VideoCodec. by mallinath@webrtc.org · 10 years ago
  22. 0ac0bca Updated WebRTC version to 3.51 by elham@webrtc.org · 10 years ago
  23. 539bbde Modify bitrate controller to update bitrate based on process call and not by andresp@webrtc.org · 10 years ago
  24. 1f49208 Adding API for setting bandwidth estimation configurations. by stefan@webrtc.org · 10 years ago
  25. 1a19092 Add configuration for ability to use the encode usage measure for triggering overuse/underuse. by asapersson@webrtc.org · 10 years ago
  26. 50ac4d6 Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. by solenberg@webrtc.org · 10 years ago
  27. 85101db Have changes to REMB trigger RTCP to be sent immediately. by stefan@webrtc.org · 10 years ago
  28. 5f804f8 VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
  29. 9b2b8ec Add AIMD option to BWE API. by stefan@webrtc.org · 10 years ago
  30. 209791d Refactor in BitrateController module. by andresp@webrtc.org · 10 years ago
  31. 40fee00 Adding operator== and != methods for CodecInst and VideoCodec structures. by mallinath@webrtc.org · 10 years ago
  32. 27bd3be Add ability to configure cpu overuse options via an API. by asapersson@webrtc.org · 10 years ago
  33. f9d5709 Fixes RTX related bugs. by stefan@webrtc.org · 10 years ago
  34. bef6e62 Simplify pacer interface. by pbos@webrtc.org · 10 years ago
  35. 3a70e6e Fix a deadlock in ViEEncoder::DeliverFrame. by wuchengli@chromium.org · 10 years ago
  36. 9420a1f Implement minimum transmit bitrate. by pbos@webrtc.org · 10 years ago
  37. f35f098 Avoid crash in ViEEncoder::DeRegisterExternalEncoder(). by fischman@webrtc.org · 10 years ago
  38. 0bf5a2f Adding a new ramp-up-down-up test by henrik.lundin@webrtc.org · 10 years ago
  39. ecee063 Adds APIs for reporting pacer queuing delay. by jiayl@webrtc.org · 10 years ago
  40. c0d56c0 Fix to get total number of sent and received rtcp packets. by asapersson@webrtc.org · 10 years ago
  41. 23c8d6b Updated WebRTC version to 3.50 TBR= wu@webrtc.org by elham@webrtc.org · 10 years ago
  42. 072bab2 Modified overuse detection thresholds. by asapersson@webrtc.org · 10 years ago
  43. 2fa9f7e Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 10 years ago
  44. ae50521 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
  45. 15e3511 Reset estimate if no frame has been seen for a certain time (to avoid large jitter if stop sending). by asapersson@webrtc.org · 10 years ago
  46. 46b22d8 Adding a critical section missing in r5543. by stefan@webrtc.org · 10 years ago
  47. 8e98655 Increase overuse and normal use thresholds for Mac. by asapersson@webrtc.org · 10 years ago
  48. 8cb4c8d Fixes a race when writing to send_padding_. by stefan@webrtc.org · 10 years ago
  49. 6cfc58d Set pacing bitrates in SetEncoder. by pbos@webrtc.org · 10 years ago
  50. ddbd31e Remove ViE external encryption API. by solenberg@webrtc.org · 10 years ago
  51. a68379b Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
  52. 49e9e15 Connect webrtc::Config to WrappingBitrateEstimator by henrik.lundin@webrtc.org · 11 years ago
  53. a1e140d Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..." by mallinath@webrtc.org · 11 years ago
  54. c091c50 Revert 5421 "Fix deadlock on register/unregister observer while ..." by mallinath@webrtc.org · 11 years ago
  55. aa2c3ae Fix deadlock on register/unregister observer while there is a an going callback. by andresp@webrtc.org · 11 years ago
  56. 224933c Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 11 years ago
  57. 64339f0 Add configuration and test for extended RTCP reference time reports to new video api. by asapersson@webrtc.org · 11 years ago
  58. c1792c5 Roll Chromium 238260 -> 243863 by wjia@webrtc.org · 11 years ago
  59. b95f445 Updated Webrtc version to 3.49 by elham@webrtc.org · 11 years ago
  60. 9c8f391 Removes usage of ListWrapper from several files. by henrike@webrtc.org · 11 years ago
  61. ca72300 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  62. 7e4053c Add thread_annotations for clang targets. by andresp@webrtc.org · 11 years ago
  63. d3f0617 If the configured start bitrate is higher than the configures max by mflodman@webrtc.org · 11 years ago
  64. 4a185e9 Race condition in ViECapturer::RegisterObserver by sprang@webrtc.org · 11 years ago
  65. e83367b Update WebRTC to version 3.48 by tnakamura@webrtc.org · 11 years ago
  66. acc2e43 Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  67. cd117d2 Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
  68. ef1f6c3 Remove media_file from VideoEngine dependencies. by pbos@webrtc.org · 11 years ago
  69. 39139dc Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  70. 0af1d21 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  71. ee867fa Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  72. 0e4512b Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  73. e4d538a Make sure channels in the same call are in the same channel group. by mflodman@webrtc.org · 11 years ago
  74. e6dc4ff Making RemoteRateControl::min_configured_bit_rate_ configurable by henrik.lundin@webrtc.org · 11 years ago
  75. 3a4fc4b Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  76. 9b3d2bf Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  77. cde78d6 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  78. a4670a1 Complete rewrite of demo application. by henrike@webrtc.org · 11 years ago
  79. 0ceb51f Remove overloaded CpuOveruseMeasure function. by asapersson@webrtc.org · 11 years ago
  80. 7123a80 Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
  81. 66e84b0 Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  82. 894dab9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  83. f1d22d4 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  84. ed8c496 Fraction lost statistics not being reported by sprang@webrtc.org · 11 years ago
  85. cf5c552 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
  86. 8db148e Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
  87. adc238a Fixing the android build by henrik.lundin@webrtc.org · 11 years ago
  88. b669e60 Remove default implementations for SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  89. 3bcea52 Fix bug where fraction_lost is always set to 0 when getting received RTCP statistics. by stefan@webrtc.org · 11 years ago
  90. 8911937 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
  91. 5459e0b Remove the long disabled WEBRTC_SVNREVISION define. by andrew@webrtc.org · 11 years ago
  92. 382cfdd Removing DropDeltaAfterKey functionality which is unused. by andresp@webrtc.org · 11 years ago
  93. 9435a17 Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
  94. f2c136b Added a delay measurement, measures the time between an incoming captured frame until the frame is being processed. Measures the delay per second. by asapersson@webrtc.org · 11 years ago
  95. da3ae7c Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
  96. 0e2571d Lock frame in ViECapturer::IncomingFrameI420. by pbos@webrtc.org · 11 years ago
  97. 801822c Ensure that no packet stays in the pacer queue for longer than 2 seconds. by stefan@webrtc.org · 11 years ago
  98. 8f9da30 Create default implementation to fix issue in libjingle by sprang@webrtc.org · 11 years ago
  99. 4a9843f Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
  100. 2622be1 Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api. by asapersson@webrtc.org · 11 years ago