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webrtc / src / webrtc / 68337ca2dc9568763d00a933cba190c0a3bb2dab / config.h
  1. 39e693a Implement timing frames. by ilnik · 8 years ago
  2. cb436f6 Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ ) by ilnik · 8 years ago
  3. 8f162b3 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ ) by ilnik · 8 years ago
  4. 1087c1e Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ ) by ilnik · 8 years ago
  5. f888bbf Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2811963002/ ) by ilnik · 8 years ago
  6. 6b65c51 Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ ) by ilnik · 8 years ago
  7. fa3ebb0 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ ) by ilnik · 8 years ago
  8. e849cea Add content type information to Encoded Images and add corresponding RTP extension header. by ilnik · 8 years ago
  9. ad5d1e6 Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 8 years ago
  10. 6f2447a Remove FlexfecConfig and replace with specific struct in VideoSendStream. by brandtr · 8 years ago
  11. 02b7ecd Keep all codec parameters in VideoReceiveStream::Decoder by magjed · 8 years ago
  12. 056bdda Wire up FlexFEC in VideoEngine2. by brandtr · 8 years ago
  13. 91c6f34 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  14. 77b3ea6 Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ ) by terelius · 8 years ago
  15. b8bff80 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  16. 4e97439 Add FlexfecReceiveStream. by brandtr · 8 years ago
  17. d984c57 Rename FecConfig to UlpfecConfig in config.h. by brandtr · 8 years ago
  18. 730a9e7 Releand of Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  19. 29f813b Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ ) by perkj · 8 years ago
  20. 68eaf60 Let ViEEncoder handle resolution changes. by perkj · 8 years ago
  21. 4cc4a8d This is a resubmission of https://codereview.webrtc.org/2047513002/ by kthelgason · 9 years ago
  22. a686d5e Moving/renaming webrtc/common.h. by solenberg · 9 years ago
  23. 3f65eaf This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 9 years ago
  24. bdd04aa Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 9 years ago
  25. cd76a16 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 9 years ago
  26. 246856b Add decoder-specific settings with proper lifetime. by johan · 9 years ago
  27. 48be054 Avoid unnecessary HW video encoder reconfiguration by skvlad · 9 years ago
  28. 5a37e3e Configure VoE NACK through AudioSendStream::Config, for send streams. by solenberg · 9 years ago
  29. 00cc045 Add sender controlled playout delay limits by isheriff · 9 years ago
  30. c4921f4 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  31. bf4689d Re-land: "Use an explicit identifier in Config" by aluebs · 9 years ago
  32. da2fece Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ ) by tommi · 9 years ago
  33. 770ba12 Use an explicit identifier in Config by aluebs · 9 years ago
  34. f95302f Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  35. ffe1ce0 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  36. 0ba16d1 Adding support for simulcast and spatial layers into VideoQualityTest by sprang · 9 years ago
  37. 2d8a1b5 Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ ) by sprang · 9 years ago
  38. 3eea752 Adding support for simulcast and spatial layers into VideoQualityTest by ivica · 9 years ago
  39. 374a570 Add support for transport wide sequence numbers by sprang · 10 years ago
  40. 87b0c58 Add options for NetEq fast accelerate mode through libjingle by Henrik Lundin · 10 years ago
  41. 3780a57 Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..." by Henrik Lundin · 10 years ago
  42. a8d7a5b Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..." by Henrik Lundin · 10 years ago
  43. 408aa7d Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..." by Henrik Lundin · 10 years ago
  44. 31eb279 Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..." by Henrik Lundin · 10 years ago
  45. 4cdbdd6 Adding a new constraint to set NetEq buffer capacity from peerconnection by Henrik Lundin · 10 years ago
  46. ad86786 Add AudioReceiveStream to Call API. by Fredrik Solenberg · 10 years ago
  47. fc398fe Set correct encoder-specific settings for vpx in the new API. by Erik Språng · 10 years ago
  48. 0f07171 Support handling multiple RTX but only generate SDP with RTX associated with VP8. by Shao Changbin · 10 years ago
  49. 42e1d11 Add CVO support to Vie layer. by guoweis@webrtc.org · 10 years ago
  50. c4e2cd0 Fix style violations in common_types.h and config.h by kwiberg@webrtc.org · 10 years ago
  51. c0cf6d3 Add decoder-timing stats to VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  52. eb1c2b5 Add RtcpPacketTypeCounter stats to new API. by pbos@webrtc.org · 10 years ago
  53. 378a41a Revert 8028 "Support associated payload type when registering Rt..." by andrew@webrtc.org · 10 years ago
  54. 846164f Support associated payload type when registering Rtx payload type. by pbos@webrtc.org · 10 years ago
  55. 1a36f78 Refactor some receive-side stats. by pbos@webrtc.org · 10 years ago
  56. ee9497c Report encoded frame size in VideoSendStream. by pbos@webrtc.org · 10 years ago
  57. f800961 Remove unused RtpStatistics struct. by pbos@webrtc.org · 10 years ago
  58. 5232267 Wire up bandwidth stats to the new API and webrtcvideoengine2. by stefan@webrtc.org · 10 years ago
  59. ddb84aa Implement conference-mode temporal-layer screencast. by pbos@webrtc.org · 10 years ago
  60. 2366875 Move min transmit bitrate to VideoEncoderConfig. by pbos@webrtc.org · 10 years ago
  61. 58b5140 Config struct for VideoEncoder. by pbos@webrtc.org · 11 years ago
  62. f0a119f Check before send/receive rtp header extensions. by pbos@webrtc.org · 11 years ago
  63. 55b0f2e Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 11 years ago
  64. 09da1a7 Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 11 years ago
  65. 7e68693 Add ToString() to VideoSendStream::Config. by pbos@webrtc.org · 11 years ago
  66. c8ab721 Wire up statistics in video receive stream of new API by sprang@webrtc.org · 11 years ago
  67. c71929d Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  68. 5b08052 Set up receiver RTX config using a std::map. by pbos@webrtc.org · 11 years ago
  69. 49812e6 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  70. 346dbe7 Rename RTP-extension constants. by pbos@webrtc.org · 11 years ago
  71. 24e2089 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago