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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 5b4a29777dab756293048ea4de1a37dbafab5c79 / video / video_send_stream.cc
  1. a7651f8 Support VP8 encoder settings in VideoSendStream. by pbos@webrtc.org · 10 years ago
  2. 49e6306 Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 10 years ago
  3. b96302f Reserve RTP/RTCP modules in SetSSRC. by pbos@webrtc.org · 10 years ago
  4. 212705c Implements start bitrate for new video API. by mflodman@webrtc.org · 10 years ago
  5. ab25d49 Enable pacing by default and remove the option to disable it from the new API. by stefan@webrtc.org · 10 years ago
  6. 508e748 Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 10 years ago
  7. 1553688 Make VideoSendStream/VideoReceiveStream configs const. by pbos@webrtc.org · 10 years ago
  8. 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
  9. 1878192 Revert "Revert "Remove VideoSendStreamInput::PutFrame."" by pbos@webrtc.org · 10 years ago
  10. da214b3 Revert "Remove VideoSendStreamInput::PutFrame." by pbos@webrtc.org · 10 years ago
  11. e62864e Remove VideoSendStreamInput::PutFrame. by pbos@webrtc.org · 10 years ago
  12. c23ed47 Fix Win VideoSendStream::...::ToString() compiles. by pbos@webrtc.org · 10 years ago
  13. b4bc1a6 Add ToString() to VideoSendStream::Config. by pbos@webrtc.org · 10 years ago
  14. c54ff69 Add thread annotations to Call API. by pbos@webrtc.org · 10 years ago
  15. 9d0f79f Rename Start/Stop in Video{Send,Receive}Streams. by pbos@webrtc.org · 10 years ago
  16. bef6e62 Simplify pacer interface. by pbos@webrtc.org · 10 years ago
  17. f39df52 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 10 years ago
  18. 9420a1f Implement minimum transmit bitrate. by pbos@webrtc.org · 10 years ago
  19. 0bf5a2f Adding a new ramp-up-down-up test by henrik.lundin@webrtc.org · 10 years ago
  20. 4b1817f Add configuration for cpu overuse detection to video send stream. by asapersson@webrtc.org · 10 years ago
  21. 48ac0da Drop early packets when not sending in TransportAdapter. by sprang@webrtc.org · 10 years ago
  22. c766775 Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  23. df9f099 Permitting double start/stopping of streams. by pbos@webrtc.org · 11 years ago
  24. ca72300 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  25. 39139dc Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  26. 0af1d21 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  27. ee867fa Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  28. ab6ccbc Adding REMB to receive stream configuration, the send side will always by mflodman@webrtc.org · 11 years ago
  29. e4d538a Make sure channels in the same call are in the same channel group. by mflodman@webrtc.org · 11 years ago
  30. 7123a80 Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
  31. 9105cbd Set up SSRCs correctly after switching codec. by pbos@webrtc.org · 11 years ago
  32. 4a9843f Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
  33. d7d60c8 Connect pacer/padding to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  34. 60108c2 Rename RTP-extension constants. by pbos@webrtc.org · 11 years ago
  35. 48cc9dc Rename video streams' start/stop methods. by pbos@webrtc.org · 11 years ago
  36. 8fdf191 Rename AutoMute to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  37. 8f2997c Implement VideoSendStream::SetCodec(). by pbos@webrtc.org · 11 years ago
  38. b581c90 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago[Renamed (93%) from video_engine/internal/video_send_stream.cc]
  39. d4ec1f5 Removing the threshold from the auto-mute APIs by henrik.lundin@webrtc.org · 11 years ago
  40. ce21c82 Porting auto mute to new ViE API by henrik.lundin@webrtc.org · 11 years ago
  41. 3ba57eb Implement I420FrameCallbacks in Call. by pbos@webrtc.org · 11 years ago
  42. 6133dd5 Fixed issue with how MTU is calculated. by sprang@webrtc.org · 11 years ago
  43. 4fe8543 Wired up max packet size and added simple test. by sprang@webrtc.org · 11 years ago
  44. aa693dd Enable FEC for VideoSendStream. by pbos@webrtc.org · 11 years ago
  45. 26d75f3 Break out glue for old->new Transport. by pbos@webrtc.org · 11 years ago
  46. f952fce Implement NACK over RTX for VideoSendStream. by pbos@webrtc.org · 11 years ago
  47. e22b761 Implement 'abs-send-time' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
  48. 905cebd Implement 'toffset' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
  49. d8e92c9 Remove newapi:: namespace for typenames without overlap. by pbos@webrtc.org · 11 years ago
  50. debc672 Enabling and testing RTCP CNAME in new API. by pbos@webrtc.org · 11 years ago
  51. a0a91d8 Adds two tests for verifying padding and ramp-up behavior. by stefan@webrtc.org · 11 years ago
  52. 30c741a Fix implicit int->bool conversion in VideoSendStream::DeliverRtcp. by pbos@webrtc.org · 11 years ago
  53. bf9bc32 Glue code and tests for NACK in new VideoEngine API. by pbos@webrtc.org · 11 years ago
  54. ecbeb2b Hooking up first simple CPU adaptation version. by mflodman@webrtc.org · 11 years ago
  55. 12d5ede Initial port of FullStackTest to new VideoEngine API. by pbos@webrtc.org · 11 years ago
  56. 2c343fc Stats+Config moved into VideoSend/ReceiveStreams. by pbos@webrtc.org · 11 years ago
  57. d9f9185 FrameGenerator class for future fake capture device. by pbos@webrtc.org · 11 years ago
  58. dc8c883 New VideoEngine API implementation on top of old one, first steps. by pbos@webrtc.org · 11 years ago