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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 5b4a29777dab756293048ea4de1a37dbafab5c79 / video_engine
  1. abbad27 Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. by andresp@webrtc.org · 10 years ago
  2. 6de25ff Thread annotations for vie_encoder.cc/.h by stefan@webrtc.org · 10 years ago
  3. c2c6375 Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel. by andresp@webrtc.org · 10 years ago
  4. 921f06d Some refactoring inside rtp_rtcp/. by pbos@webrtc.org · 10 years ago
  5. 49e6306 Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 10 years ago
  6. c311a0d Add boilerplate code for H.264. by stefan@webrtc.org · 10 years ago
  7. 6b32f7d Configure RTX send status on new modules. by pbos@webrtc.org · 10 years ago
  8. f047ce8 Introduces PacedVideoSender to test framework and moves the Pacer to use Clock. by stefan@webrtc.org · 10 years ago
  9. ce387cc Removed old code and default implementations. by asapersson@webrtc.org · 10 years ago
  10. 49cfa2e Possibly fix deadlock happening due to unregister/register modules as switching between AST and TSO estimators. by andresp@webrtc.org · 10 years ago
  11. b96302f Reserve RTP/RTCP modules in SetSSRC. by pbos@webrtc.org · 10 years ago
  12. 6851932 Bump version number to 3.55 by tnakamura@webrtc.org · 10 years ago
  13. 97ce37c GN: Add BUILD.gn files + kjellander to OWNERS by kjellander@webrtc.org · 10 years ago
  14. 620f146 Add tests of texture frames in video_send_stream_test. by wuchengli@chromium.org · 10 years ago
  15. b929c5f Add max limit of number for overuses. When limit is reached always apply the rampup delay. by asapersson@webrtc.org · 10 years ago
  16. 0f4e394 Pass GYP DEPTH variable to isolate. by kjellander@webrtc.org · 10 years ago
  17. ab25d49 Enable pacing by default and remove the option to disable it from the new API. by stefan@webrtc.org · 10 years ago
  18. a78c821 Increased kMaxRampUpDelayMs (120 to 240s). by asapersson@webrtc.org · 10 years ago
  19. 508e748 Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 10 years ago
  20. 7d65bab Add additional metric (relative standard deviation of encode time) for overuse detection. by asapersson@webrtc.org · 10 years ago
  21. c3dd427 Add kjellander@webrtc.org as OWNER for *.isolate by kjellander@webrtc.org · 10 years ago
  22. 65cba30 ViEAutoTestAndroid: Unbreak compile by casting void* to jobject. by fischman@webrtc.org · 10 years ago
  23. 0b9b417 AppRTCDemo(android): support app (UI) & capture rotation. by fischman@webrtc.org · 10 years ago
  24. 77cf8de Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. by stefan@webrtc.org · 10 years ago
  25. a7627c4 Android: cleanup gtest_target_type conditions. by henrike@webrtc.org · 10 years ago
  26. ba57781 Add support of texture frames for video capturer. by wuchengli@chromium.org · 10 years ago
  27. f32ce15 Revert "Add support of texture frames for video capturer." by wuchengli@chromium.org · 10 years ago
  28. b4ead22 Add support of texture frames for video capturer. by wuchengli@chromium.org · 10 years ago
  29. 30caf16 Added api for getting cpu measures using a struct. by asapersson@webrtc.org · 10 years ago
  30. 050a15e First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class. by asapersson@webrtc.org · 10 years ago
  31. 1754605 vie_autotest_android.cc: stop referring to undefined functions. by fischman@webrtc.org · 10 years ago
  32. 1878192 Revert "Revert "Remove VideoSendStreamInput::PutFrame."" by pbos@webrtc.org · 10 years ago
  33. 13e6c08 Fix deadlock in RegisterPreDecodeImageCallback. by pbos@webrtc.org · 10 years ago
  34. d64e778 Bump WebRTC version number to 3.54 TBR=wu@webrtc.org by tnakamura@webrtc.org · 10 years ago
  35. 3e2c13a Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
  36. a37d7f6 Revert 6202 "Switch to using base/constructormagic.h and remove ..." by mcasas@webrtc.org · 10 years ago
  37. b186b26 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
  38. 4c31dc2 Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main} by andresp@webrtc.org · 10 years ago
  39. 75364e4 Ignore the return value of UpdateRtcpTimestamp instead of printing warning. Because UpdateRtcpTimestamp may fail when there's no valid RTCP SR, which can happen in the first couple seconds or when the channel is a send only channel. Either case we don't want the warning log. by wu@webrtc.org · 10 years ago
  40. 2a8fb71 Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe. by wu@webrtc.org · 10 years ago
  41. 11fa357 Add webrtc field trials API. by andresp@webrtc.org · 10 years ago
  42. 04a721c Remove WEBRTC_TRACE uses in video_engine/ by pbos@webrtc.org · 10 years ago
  43. 9754a5d Make vie/voe_auto_test accept non-supported flags without error. by kjellander@webrtc.org · 10 years ago
  44. b16a722 Reduced kMaxSampleDiffMs (limit to 22fps). by asapersson@webrtc.org · 10 years ago
  45. a229768 Fix failing test introduced with r6111. by stefan@webrtc.org · 10 years ago
  46. 305fd94 Fixes log spam introduced with r6041. by stefan@webrtc.org · 10 years ago
  47. 0d47fe1 Raise kViEMaxNumberOfChannels from 32 to 64 by wu@webrtc.org · 10 years ago
  48. 569487d Updated WebRTC version to 3.53 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
  49. 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
  50. 73c2412 Remove timestamp_extrapolator's dependency to Clock and vcm defines. by wu@webrtc.org · 10 years ago
  51. 1a9e6ac Pointers were not dereferenced in GetRtpStatistics. by asapersson@webrtc.org · 10 years ago
  52. 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
  53. c6cfc5c Upping start bitrate to min, if set to a lower value i SetSendCodec. by mflodman@webrtc.org · 10 years ago
  54. 73e1a8b Disable flaky RunsRtpRtcpTestWIthoutErrors. by pbos@webrtc.org · 10 years ago
  55. b0079ed Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
  56. 0a5fd54 Casting char to int in logs. by asapersson@webrtc.org · 10 years ago
  57. b991cd0 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
  58. a1626fe Propagate capture ntp timestamp from rtp to renderer. by wu@webrtc.org · 10 years ago
  59. 6b1114a Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
  60. b18bff5 Cleaned up logging in video_coding. by stefan@webrtc.org · 10 years ago
  61. 4b0cd7f Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 10 years ago
  62. ff46b81 Updated WebRTC version to 3.52 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
  63. 44c9b9a Log Fixit for parts of video_engine folder. by mflodman@webrtc.org · 10 years ago
  64. 6c57efd Re-submit: rev5775 by andresp@webrtc.org · 10 years ago
  65. 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
  66. 765ea72 Revert 5775 "Modify bitrate controller to update bitrate based o..." by andrew@webrtc.org · 10 years ago
  67. f50914a Removing VideoCodecDerived and moving methods inside VideoCodec. by mallinath@webrtc.org · 10 years ago
  68. 0ac0bca Updated WebRTC version to 3.51 by elham@webrtc.org · 10 years ago
  69. 539bbde Modify bitrate controller to update bitrate based on process call and not by andresp@webrtc.org · 10 years ago
  70. 1f49208 Adding API for setting bandwidth estimation configurations. by stefan@webrtc.org · 10 years ago
  71. 1a19092 Add configuration for ability to use the encode usage measure for triggering overuse/underuse. by asapersson@webrtc.org · 10 years ago
  72. 50ac4d6 Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. by solenberg@webrtc.org · 10 years ago
  73. 85101db Have changes to REMB trigger RTCP to be sent immediately. by stefan@webrtc.org · 10 years ago
  74. 5f804f8 VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
  75. 9b2b8ec Add AIMD option to BWE API. by stefan@webrtc.org · 10 years ago
  76. 209791d Refactor in BitrateController module. by andresp@webrtc.org · 10 years ago
  77. 40fee00 Adding operator== and != methods for CodecInst and VideoCodec structures. by mallinath@webrtc.org · 10 years ago
  78. 27bd3be Add ability to configure cpu overuse options via an API. by asapersson@webrtc.org · 10 years ago
  79. f9d5709 Fixes RTX related bugs. by stefan@webrtc.org · 10 years ago
  80. bef6e62 Simplify pacer interface. by pbos@webrtc.org · 10 years ago
  81. 3a70e6e Fix a deadlock in ViEEncoder::DeliverFrame. by wuchengli@chromium.org · 10 years ago
  82. 9420a1f Implement minimum transmit bitrate. by pbos@webrtc.org · 10 years ago
  83. f35f098 Avoid crash in ViEEncoder::DeRegisterExternalEncoder(). by fischman@webrtc.org · 10 years ago
  84. 0bf5a2f Adding a new ramp-up-down-up test by henrik.lundin@webrtc.org · 10 years ago
  85. ecee063 Adds APIs for reporting pacer queuing delay. by jiayl@webrtc.org · 10 years ago
  86. c0d56c0 Fix to get total number of sent and received rtcp packets. by asapersson@webrtc.org · 10 years ago
  87. 23c8d6b Updated WebRTC version to 3.50 TBR= wu@webrtc.org by elham@webrtc.org · 10 years ago
  88. 072bab2 Modified overuse detection thresholds. by asapersson@webrtc.org · 10 years ago
  89. 2fa9f7e Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 10 years ago
  90. ae50521 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
  91. 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
  92. 46b22d8 Adding a critical section missing in r5543. by stefan@webrtc.org · 10 years ago
  93. 8e98655 Increase overuse and normal use thresholds for Mac. by asapersson@webrtc.org · 10 years ago
  94. 8cb4c8d Fixes a race when writing to send_padding_. by stefan@webrtc.org · 10 years ago
  95. 6cfc58d Set pacing bitrates in SetEncoder. by pbos@webrtc.org · 10 years ago
  96. ddbd31e Remove ViE external encryption API. by solenberg@webrtc.org · 10 years ago
  97. a68379b Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
  98. 49e9e15 Connect webrtc::Config to WrappingBitrateEstimator by henrik.lundin@webrtc.org · 11 years ago
  99. a1e140d Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..." by mallinath@webrtc.org · 11 years ago
  100. c091c50 Revert 5421 "Fix deadlock on register/unregister observer while ..." by mallinath@webrtc.org · 11 years ago