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webrtc / src / 5acd9d0393260b63e80b435fed9e5024ea6f4498 / webrtc / video_send_stream.h
  1. 440b6d9 Move video send/receive stream headers to webrtc/call. by aleloi · 8 years ago
  2. e5c4a81 Move RTP keep-alive config from VideoSendStream::Config to Call::Config by sprang · 8 years ago
  3. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  4. 168794c Implement RTP keepalive in native stack. by sprang · 8 years ago
  5. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  6. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  7. 3c81a1a Add field trial for balanced degradation preference. by asapersson · 8 years ago
  8. 09f0561 Update adaptation stats to support degradations in both resolution and framerate. by asapersson · 8 years ago
  9. fab6707 Add number of quality adapt changes to VideoSendStream stats. by asapersson · 8 years ago
  10. c5d62e2 Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ ) by sprang · 8 years ago
  11. f9ed235 Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ ) by lliuu · 8 years ago
  12. 3ea3c77 Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ ) by sprang · 8 years ago
  13. 8b45b11 Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ ) by skvlad · 8 years ago
  14. 72acf25 Add framerate to VideoSinkWants and ability to signal on overuse by sprang · 8 years ago
  15. 3d200bd Remove FlexfecConfig and replace with specific struct in VideoSendStream. by brandtr · 8 years ago
  16. 80ed35e Implement periodic bandwidth probing in application-limited region. by sergeyu · 8 years ago
  17. a8eb756 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
  18. a6a699a Sent bitrate stats are incorrect if FlexFEC is configured: by asapersson · 8 years ago
  19. e950cad Wire up FlexfecSender in RTP module and VideoSendStream. by brandtr · 8 years ago
  20. 803d97f Let ViEEncoder express resolution requests as Sinkwants. by perkj · 8 years ago
  21. 87da404 Implement qpSum stat for video send ssrc stats. by sakal · 8 years ago
  22. 43536c3 Implement framesEncoded stat in video send ssrc stats. by sakal · 8 years ago
  23. b5f2c3f Rename FecConfig to UlpfecConfig in config.h. by brandtr · 8 years ago
  24. a48ddb7 Add VideoSendStream::Stats::prefered_media_bitrate_bps by Per · 9 years ago
  25. e75f204 Expose Ivf logging through the native API by palmkvist · 9 years ago
  26. a49cbd3 Replace VideoCapturerInput with VideoSinkInterface. by perkj · 9 years ago
  27. 9fdbda6 Revert of Replace interface VideoCapturerInput with VideoSinkInterface. (patchset #13 id:280001 of https://codereview.webrtc.org/2257413002/ ) by perkj · 9 years ago
  28. 95a226f Replace VideoCapturerInput with VideoSinkInterface. by perkj · 9 years ago
  29. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 9 years ago
  30. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 9 years ago
  31. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 9 years ago
  32. 2e5cfcd Add periodic logging of video stats. by asapersson · 9 years ago
  33. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 9 years ago
  34. a96b60b Move frame_callback.h to common_video/include. by pbos · 9 years ago
  35. d30a111 Change pre_encode_callback to get a const frame. by nisse · 9 years ago
  36. 7ade7b3 Delete class webrtc::VideoRenderer and its header file. by nisse · 9 years ago
  37. 905f8e7 Make ReconfigureVideoEncoder void. by Peter Boström · 9 years ago
  38. e449915 Measure encoding time on encode callbacks. by Peter Boström · 9 years ago
  39. b7d9a97 Expose codec implementation names in stats. by Peter Boström · 9 years ago
  40. 17821db Wire up bandwidth limitation info to GetStats and adapt_reason. by asapersson · 9 years ago
  41. 1387149 Adding reduced size RTCP configuration down to the video stream level. by deadbeef · 9 years ago
  42. 8d15bd6 Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ ) by ivica · 9 years ago
  43. 8104479 Revert of Collecting encode_time_ms for each frame (patchset #13 id:220001 of https://codereview.webrtc.org/1374233002/ ) by kjellander · 9 years ago
  44. 092b133 Collecting encode_time_ms for each frame. by ivica · 9 years ago
  45. 2d56668 Unify Transport and newapi::Transport interfaces. by pbos · 10 years ago
  46. e526974 Make LoadObserver settable per video send stream. Gives client flexibility and makes the implementation slightly simpler. See discussion in: https://codereview.webrtc.org/1269863005/ by solenberg · 10 years ago
  47. 47d78cc Pass the encoder's internal source property through to video_sender to request a keyframe from the external encoder by sophiechang · 10 years ago
  48. 4fbae2b Add send transports to individual webrtc::Call streams. by solenberg · 10 years ago
  49. cd67022 Define Stream base classes by Jelena Marusic · 10 years ago
  50. 4b91bd0 Move frame input (ViECapturer) to webrtc/video/. by Peter Boström · 10 years ago
  51. 78fb3b3 C++11 in-class member initialization in Call configs. by Fredrik Solenberg · 10 years ago
  52. 4765070 Rename I420VideoFrame to VideoFrame. by Miguel Casas-Sanchez · 10 years ago
  53. af612d5 Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."" by perkj@webrtc.org · 10 years ago
  54. d7452a0 Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame." by magjed@webrtc.org · 10 years ago
  55. bcead30 Make the entry point for VideoFrames to webrtc const ref I420VideoFrame. by perkj@webrtc.org · 10 years ago
  56. 891d483 Wire up target_media_bitrate in VideoSendStream. by pbos@webrtc.org · 10 years ago
  57. 3e6e271 Implement CpuOveruseMetrics as callbacks. by pbos@webrtc.org · 10 years ago
  58. 09c77b9 Add decoder-timing stats to VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  59. 32e8528 Log configs when creating video streams in Call. by pbos@webrtc.org · 10 years ago
  60. 742386a Enable payload-based padding by default and remove the API. by stefan@webrtc.org · 10 years ago
  61. 273a414 Report encoded frame size in VideoSendStream. by pbos@webrtc.org · 10 years ago
  62. 0bae1fa Wire up bandwidth stats to the new API and webrtcvideoengine2. by stefan@webrtc.org · 10 years ago
  63. ad3b5a5 Move min transmit bitrate to VideoEncoderConfig. by pbos@webrtc.org · 10 years ago
  64. bbe0a85 Config struct for VideoEncoder. by pbos@webrtc.org · 11 years ago
  65. 168f23f Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 11 years ago
  66. 4ef438e Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 11 years ago
  67. cb254aa Enable pacing by default and remove the option to disable it from the new API. by stefan@webrtc.org · 11 years ago
  68. fbb567d Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 11 years ago
  69. 6ae48c6 Make VideoSendStream/VideoReceiveStream configs const. by pbos@webrtc.org · 11 years ago
  70. ef92755 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. by stefan@webrtc.org · 11 years ago
  71. 1566ee2 Revert "Revert "Remove VideoSendStreamInput::PutFrame."" by pbos@webrtc.org · 11 years ago
  72. 2cdd433 Revert "Remove VideoSendStreamInput::PutFrame." by pbos@webrtc.org · 11 years ago
  73. f3085e4 Remove VideoSendStreamInput::PutFrame. by pbos@webrtc.org · 11 years ago
  74. 1e92b0a Add ToString() to VideoSendStream::Config. by pbos@webrtc.org · 11 years ago
  75. a5c8d2c Rename Start/Stop in Video{Send,Receive}Streams. by pbos@webrtc.org · 11 years ago
  76. 709e297 Simplify pacer interface. by pbos@webrtc.org · 11 years ago
  77. f577ae9 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 11 years ago
  78. b10363f3 Re-landing "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 11 years ago
  79. 3349ae0 Implement minimum transmit bitrate. by pbos@webrtc.org · 11 years ago
  80. be39470 Revert "Routing SuspendChange to VideoSendStream::Stats" by henrik.lundin@webrtc.org · 11 years ago
  81. 1598b80 Routing SuspendChange to VideoSendStream::Stats by henrik.lundin@webrtc.org · 11 years ago
  82. 0931570 Wire up statistics in video receive stream of new API by sprang@webrtc.org · 11 years ago
  83. c279a5d Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  84. ccd4284 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  85. b429e51 cpplint cleaning new API and its implementation files. by mflodman@webrtc.org · 11 years ago
  86. 724947b Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
  87. 4070935 Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
  88. 331d440 Connect pacer/padding to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  89. 53c8573 Rename video streams' start/stop methods. by pbos@webrtc.org · 11 years ago
  90. ce8e093 Rename AutoMute to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  91. 6488761 Implement VideoSendStream::SetCodec(). by pbos@webrtc.org · 11 years ago
  92. 16e03b7 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago[Renamed (96%) from webrtc/video_engine/new_include/video_send_stream.h]
  93. 1a3a6e5 Removing the threshold from the auto-mute APIs by henrik.lundin@webrtc.org · 11 years ago
  94. ba975e2 Porting auto mute to new ViE API by henrik.lundin@webrtc.org · 11 years ago
  95. c11148b Compound/reduced-size RTCP in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  96. 25fce9a Fixed issue with how MTU is calculated. by sprang@webrtc.org · 11 years ago
  97. 5d957e2 Wired up max packet size and added simple test. by sprang@webrtc.org · 11 years ago
  98. 74fa489 Remove newapi:: namespace for typenames without overlap. by pbos@webrtc.org · 12 years ago
  99. 360e376 Adds two tests for verifying padding and ramp-up behavior. by stefan@webrtc.org · 12 years ago
  100. 7f1b0ae Fix init list for VideoSendStream::Config::Rtp. by pbos@webrtc.org · 12 years ago