Gitiles
Code Review Sign In
gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / b7e5b2741f9cbd68e7be54e3445d28d266477c92 / video_engine
  1. f33a674 Make vie/voe_auto_test accept non-supported flags without error. by kjellander@webrtc.org · 10 years ago
  2. c773ded Reduced kMaxSampleDiffMs (limit to 22fps). by asapersson@webrtc.org · 10 years ago
  3. 068cd6f Fix failing test introduced with r6111. by stefan@webrtc.org · 10 years ago
  4. b50d671 Fixes log spam introduced with r6041. by stefan@webrtc.org · 10 years ago
  5. b43e80d Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at e1f0419756d58f744f902fe62fa31da126c55abe by Android Chromium Automerger · 10 years ago
  6. 6ecc773 Raise kViEMaxNumberOfChannels from 32 to 64 by wu@webrtc.org · 10 years ago
  7. e1f0419 Updated WebRTC version to 3.53 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
  8. 5b371a8 Update makefiles after merge of Chromium at 269041 by Android Chromium Automerger · 10 years ago
  9. 0a2aa9a Update makefiles after merge of Chromium at 269030 by Android Chromium Automerger · 10 years ago
  10. 78c3fc4 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at d2fb259b3bc61c68f368929510215a7ee7d00fda by Android Chromium Automerger · 10 years ago
  11. d2fb259 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
  12. 625ace9 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 39d9fa5486157fb4b3ab28ae403aeaa6d651e92b by Android Chromium Automerger · 10 years ago
  13. 39d9fa5 Remove timestamp_extrapolator's dependency to Clock and vcm defines. by wu@webrtc.org · 10 years ago
  14. 0de5f22 Update makefiles after merge of Chromium at 268379 by Android Chromium Automerger · 10 years ago
  15. d885109 Pointers were not dereferenced in GetRtpStatistics. by asapersson@webrtc.org · 10 years ago
  16. 4d61a36 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
  17. 3a802b9 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at cfed80f78803395dd066261afb5c5d99e5048d5d by Android Chromium Automerger · 10 years ago
  18. 5435208 Upping start bitrate to min, if set to a lower value i SetSendCodec. by mflodman@webrtc.org · 10 years ago
  19. 109a4e7 Update makefiles after merge of Chromium at 266543 by Android Chromium Automerger · 10 years ago
  20. c5fccd6 Disable flaky RunsRtpRtcpTestWIthoutErrors. by pbos@webrtc.org · 10 years ago
  21. ba47616 Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
  22. 0c9f7d3 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 1d95c5aecd0a4924f39f24834fb06d06e61f181e by Android Chromium Automerger · 10 years ago
  23. 1d95c5a Casting char to int in logs. by asapersson@webrtc.org · 10 years ago
  24. 093fc0b Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
  25. 0197db2 Update makefiles after merge of Chromium at 265680 by Android Chromium Automerger · 10 years ago
  26. 658bfc4 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at b031016337f09d758a97ed51b67788e574431103 by Android Chromium Automerger · 10 years ago
  27. 9d10769 Propagate capture ntp timestamp from rtp to renderer. by wu@webrtc.org · 10 years ago
  28. 98f8320 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
  29. 8edccce Cleaned up logging in video_coding. by stefan@webrtc.org · 10 years ago
  30. 54b48fa Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 55bc2810c06fe624311518c4502af5ca8a5c085c by Android Chromium Automerger · 10 years ago
  31. c71dd0d Update makefiles after merge of Chromium at 262754 by Android Chromium Automerger · 10 years ago
  32. 9968131 Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 10 years ago
  33. a32583c Updated WebRTC version to 3.52 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
  34. 97a64e2 Update makefiles after merge of Chromium at 262110 by Android Chromium Automerger · 10 years ago
  35. 022615b Log Fixit for parts of video_engine folder. by mflodman@webrtc.org · 10 years ago
  36. e111a22 Update makefiles after merge of Chromium at 261622 by Android Chromium Automerger · 10 years ago
  37. 2ebb6ba Update makefiles after merge of Chromium at 260927 by Android Chromium Automerger · 10 years ago
  38. d7aa228 Re-submit: rev5775 by andresp@webrtc.org · 10 years ago
  39. 2f0c5f7 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
  40. ca28c29 Revert 5775 "Modify bitrate controller to update bitrate based o..." by andrew@webrtc.org · 10 years ago
  41. 2a0cbfc Removing VideoCodecDerived and moving methods inside VideoCodec. by mallinath@webrtc.org · 10 years ago
  42. 60ae794 Updated WebRTC version to 3.51 by elham@webrtc.org · 10 years ago
  43. a0320c2 Modify bitrate controller to update bitrate based on process call and not by andresp@webrtc.org · 10 years ago
  44. 5d8c954 Adding API for setting bandwidth estimation configurations. by stefan@webrtc.org · 10 years ago
  45. 154951d Add configuration for ability to use the encode usage measure for triggering overuse/underuse. by asapersson@webrtc.org · 10 years ago
  46. 2d3624c Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. by solenberg@webrtc.org · 10 years ago
  47. 5374dab Have changes to REMB trigger RTCP to be sent immediately. by stefan@webrtc.org · 10 years ago
  48. fec6b6e VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
  49. b9d0acb Add AIMD option to BWE API. by stefan@webrtc.org · 10 years ago
  50. 3b6c0e5 Refactor in BitrateController module. by andresp@webrtc.org · 10 years ago
  51. 18c2945 Adding operator== and != methods for CodecInst and VideoCodec structures. by mallinath@webrtc.org · 10 years ago
  52. 9da327c Add ability to configure cpu overuse options via an API. by asapersson@webrtc.org · 10 years ago
  53. fbd6f47 Fixes RTX related bugs. by stefan@webrtc.org · 10 years ago
  54. 1d61e3a Simplify pacer interface. by pbos@webrtc.org · 10 years ago
  55. 76cd2f7 Fix a deadlock in ViEEncoder::DeliverFrame. by wuchengli@chromium.org · 10 years ago
  56. 3f83f9c Implement minimum transmit bitrate. by pbos@webrtc.org · 10 years ago
  57. bbbe9b8 Avoid crash in ViEEncoder::DeRegisterExternalEncoder(). by fischman@webrtc.org · 10 years ago
  58. c766098 Adding a new ramp-up-down-up test by henrik.lundin@webrtc.org · 10 years ago
  59. 55a2a27 Adds APIs for reporting pacer queuing delay. by jiayl@webrtc.org · 10 years ago
  60. 9aef225 Fix to get total number of sent and received rtcp packets. by asapersson@webrtc.org · 10 years ago
  61. fd1b455 Updated WebRTC version to 3.50 TBR= wu@webrtc.org by elham@webrtc.org · 10 years ago
  62. 0b95677 Modified overuse detection thresholds. by asapersson@webrtc.org · 10 years ago
  63. 4a15560 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 10 years ago
  64. a56c5b4 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
  65. 9950796 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
  66. 5a6e691 Adding a critical section missing in r5543. by stefan@webrtc.org · 10 years ago
  67. 5909065 Increase overuse and normal use thresholds for Mac. by asapersson@webrtc.org · 10 years ago
  68. 5fcd7df Fixes a race when writing to send_padding_. by stefan@webrtc.org · 10 years ago
  69. 8e1a876 Set pacing bitrates in SetEncoder. by pbos@webrtc.org · 10 years ago
  70. 800136d Remove ViE external encryption API. by solenberg@webrtc.org · 10 years ago
  71. d1e7fac Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
  72. d2f95a8 Connect webrtc::Config to WrappingBitrateEstimator by henrik.lundin@webrtc.org · 10 years ago
  73. 7dc7514 Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..." by mallinath@webrtc.org · 10 years ago
  74. 9afb6cf Revert 5421 "Fix deadlock on register/unregister observer while ..." by mallinath@webrtc.org · 10 years ago
  75. 630939f Fix deadlock on register/unregister observer while there is a an going callback. by andresp@webrtc.org · 11 years ago
  76. 7d99cd4 Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 11 years ago
  77. b4263e0 Add configuration and test for extended RTCP reference time reports to new video api. by asapersson@webrtc.org · 11 years ago
  78. fba4f1c Roll Chromium 238260 -> 243863 by wjia@webrtc.org · 11 years ago
  79. eed1f11 Updated Webrtc version to 3.49 by elham@webrtc.org · 11 years ago
  80. 083049f Removes usage of ListWrapper from several files. by henrike@webrtc.org · 11 years ago
  81. 49812e6 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  82. c9faf10 Add thread_annotations for clang targets. by andresp@webrtc.org · 11 years ago
  83. a9a7327 If the configured start bitrate is higher than the configures max by mflodman@webrtc.org · 11 years ago
  84. 9662535 Race condition in ViECapturer::RegisterObserver by sprang@webrtc.org · 11 years ago
  85. 91cebfc Update WebRTC to version 3.48 by tnakamura@webrtc.org · 11 years ago
  86. 4f1f5fa Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  87. aacdb9f Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
  88. 5596ac6 Remove media_file from VideoEngine dependencies. by pbos@webrtc.org · 11 years ago
  89. 46f7288 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  90. c5a5713 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  91. 3bbc91e Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  92. b70db6d Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  93. dadfc9e Make sure channels in the same call are in the same channel group. by mflodman@webrtc.org · 11 years ago
  94. c49a3fa Making RemoteRateControl::min_configured_bit_rate_ configurable by henrik.lundin@webrtc.org · 11 years ago
  95. efeb8ce Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  96. 12553ad Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  97. 984bee2 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  98. a48c91d Complete rewrite of demo application. by henrike@webrtc.org · 11 years ago
  99. 4fcb2f5 Remove overloaded CpuOveruseMeasure function. by asapersson@webrtc.org · 11 years ago
  100. c33d37c Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago