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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / b7e5b2741f9cbd68e7be54e3445d28d266477c92 / video_engine / include
  1. d1d198b Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics(). by stefan@webrtc.org · 10 years ago
  2. 15097fc Remove the VPM denoiser. by pbos@webrtc.org · 10 years ago
  3. 55b0f2e Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 10 years ago
  4. 09da1a7 Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 10 years ago
  5. 2fd91bd Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 10 years ago
  6. 07dc4be Removed old code and default implementations. by asapersson@webrtc.org · 10 years ago
  7. f6eaabf Increased kMaxRampUpDelayMs (120 to 240s). by asapersson@webrtc.org · 10 years ago
  8. 6845de7 Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 10 years ago
  9. 9cd8281 Add additional metric (relative standard deviation of encode time) for overuse detection. by asapersson@webrtc.org · 10 years ago
  10. 5101f84 AppRTCDemo(android): support app (UI) & capture rotation. by fischman@webrtc.org · 10 years ago
  11. 4e95436 Added api for getting cpu measures using a struct. by asapersson@webrtc.org · 10 years ago
  12. 9d10769 Propagate capture ntp timestamp from rtp to renderer. by wu@webrtc.org · 10 years ago
  13. 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
  14. 5d8c954 Adding API for setting bandwidth estimation configurations. by stefan@webrtc.org · 10 years ago
  15. 154951d Add configuration for ability to use the encode usage measure for triggering overuse/underuse. by asapersson@webrtc.org · 10 years ago
  16. 2d3624c Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. by solenberg@webrtc.org · 10 years ago
  17. fec6b6e VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
  18. 9da327c Add ability to configure cpu overuse options via an API. by asapersson@webrtc.org · 10 years ago
  19. 3f83f9c Implement minimum transmit bitrate. by pbos@webrtc.org · 10 years ago
  20. 55a2a27 Adds APIs for reporting pacer queuing delay. by jiayl@webrtc.org · 10 years ago
  21. 4a15560 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 10 years ago
  22. 800136d Remove ViE external encryption API. by solenberg@webrtc.org · 10 years ago
  23. d1e7fac Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
  24. efeb8ce Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  25. 12553ad Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  26. 984bee2 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  27. 4fcb2f5 Remove overloaded CpuOveruseMeasure function. by asapersson@webrtc.org · 11 years ago
  28. c33d37c Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
  29. ee234be Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
  30. db04941 Remove default implementations for SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  31. f1630b1 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
  32. 5a669d5 Lock frame in ViECapturer::IncomingFrameI420. by pbos@webrtc.org · 11 years ago
  33. 163393e Create default implementation to fix issue in libjingle by sprang@webrtc.org · 11 years ago
  34. 2e98d45 Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
  35. dd4f866 Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api. by asapersson@webrtc.org · 11 years ago
  36. 4ee440a Remove const in vie_rtp_rtcp, where there is conflict with by sprang@webrtc.org · 11 years ago
  37. 50293f5 Replace VideoFrameI420 with I420VideoFrame. by pbos@webrtc.org · 11 years ago
  38. 82b883c Renaming ViEEncoderObserver::VideoSuspended by henrik.lundin@webrtc.org · 11 years ago
  39. 3dc7ff3 Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
  40. 4673674 Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
  41. 4747585 Added ViE API for getting overuse measure. by asapersson@webrtc.org · 11 years ago
  42. 3051951 Deliver I420VideoFrames from VideoRender module. by pbos@webrtc.org · 11 years ago
  43. 4590177 Rename AutoMute to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  44. af92d3e Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
  45. a191cb0 Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
  46. 6baaf30 Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
  47. 7773eec Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
  48. f00942a Drop ViEDecoderObserver::DecoderTiming impl now that WebRtcDecoderObserver rolled in r5038. by fischman@webrtc.org · 11 years ago
  49. 4ce7590 Removing the threshold from the auto-mute APIs by henrik.lundin@webrtc.org · 11 years ago
  50. ecfef19 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. by fischman@webrtc.org · 11 years ago
  51. 7c46e95 AutoMute: Adding channel_id parameter to callback. by henrik.lundin@webrtc.org · 11 years ago
  52. 63301bd Implement I420FrameCallbacks in Call. by pbos@webrtc.org · 11 years ago
  53. 81cd5ca VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. by fischman@webrtc.org · 11 years ago
  54. 499392c Minor fix to avoid breakage by henrik.lundin@webrtc.org · 11 years ago
  55. 39079d1 Piping AutoMuter interface through to ViE API by henrik.lundin@webrtc.org · 11 years ago
  56. 9b7bdee Revert r4562 by elham@webrtc.org · 11 years ago
  57. ece3d35 Added choice of decode error mode to loopback test. by agalusza@google.com · 11 years ago
  58. 7fc75bb Update talk to 50918584. by wu@webrtc.org · 11 years ago
  59. ea7b33e * Update libjingle to 50389769. by wu@webrtc.org · 11 years ago
  60. bf76ae2 Hooking up first simple CPU adaptation version. by mflodman@webrtc.org · 11 years ago
  61. 0ba496b Revert r4301 by tnakamura@webrtc.org · 11 years ago
  62. a32d18f Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
  63. 0291c80 Removed ViE file API. by mflodman@webrtc.org · 11 years ago
  64. f40e9b6 - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. by solenberg@webrtc.org · 11 years ago
  65. 453f9c0 Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
  66. 281cff8 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
  67. 7645e4d Remove SetOverUseDetectorOptions and cleaned ViESharedData. by mflodman@webrtc.org · 11 years ago
  68. 0425392 Adding a factory to remote bitrate estimator and allow it to be set via config. by andresp@webrtc.org · 11 years ago
  69. ac6d919 Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. by andresp@webrtc.org · 11 years ago
  70. 7bc7e02 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
  71. 9b53152 Change capture interface to use NTP capture time. by stefan@webrtc.org · 11 years ago
  72. 67879bc WebRtc_Word32 -> int32_t in video_engine/ by pbos@webrtc.org · 11 years ago
  73. 065b64d Remove UDP transport API from ViE by pwestin@webrtc.org · 11 years ago
  74. 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 · 11 years ago
  75. dca71b2 Add interface to signal a network down event. by stefan@webrtc.org · 11 years ago
  76. 912b7f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
  77. 2daec4c Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. by pwestin@webrtc.org · 11 years ago
  78. 9d6fcb3 Adding a receive side API for buffering mode. by mikhal@webrtc.org · 11 years ago
  79. cd1ac8b Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 11 years ago
  80. 1619664 Adding a send side API for streaming by mikhal@webrtc.org · 11 years ago
  81. 05d046c Removing outdated comment by mikhal@webrtc.org · 12 years ago
  82. 71f3f68 Enable external encoders with internal picture source. by stefan@webrtc.org · 12 years ago
  83. f314c80 Added API to get receive side video delay. by mflodman@webrtc.org · 12 years ago
  84. e7d1c04 Removed ViEBaseObserver. by mflodman@webrtc.org · 12 years ago
  85. 3bbed74 Switching to I420VideoFrame by mikhal@webrtc.org · 12 years ago
  86. b015cbe Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago