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gerrit-public.fairphone.software / platform / frameworks / base / 8cc9a8e132ebbf852e055ed17ab91f35b02fd974 / media / libstagefright / rtsp / MyHandler.h
  1. 61a8b53 am aa82c39b: am 0ba9380a: Merge "Fix Bitreader "putBits" implementation, make sure we emulate timestamps" into ics-mr1 by Andreas Huber · 13 years ago
  2. f6ae711 Fix Bitreader "putBits" implementation, make sure we emulate timestamps by Andreas Huber · 13 years ago
  3. 3dc44d9 am 8a065423: am 23217182: Merge "Didn\'t mean to check this in..." into ics-mr1 by Andreas Huber · 13 years ago
  4. d8ad2fa am 40461ee7: am cd556b82: Merge "Instead of asserting, signal a runtime error if the session doesn\'t contain" into ics-mr1 by Andreas Huber · 13 years ago
  5. 54c7efa Didn't mean to check this in... by Andreas Huber · 13 years ago
  6. 57cc14f Instead of asserting, signal a runtime error if the session doesn't contain by Andreas Huber · 13 years ago
  7. 62c0c32 am 9e2949c6: am 2375d163: Merge "Send RTSP control connection keep-alive requests" into ics-mr1 by Andreas Huber · 13 years ago
  8. a1b3e3d Send RTSP control connection keep-alive requests by Andreas Huber · 13 years ago
  9. 71f2cf1 Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE by Steve Block · 13 years ago
  10. a6be6dc NuPlayer is now taking on the task of streaming over RTSP. by Andreas Huber · 13 years ago
  11. d5a20d8 Network traffic accounting for chromium stack support in mediaserver. by Ashish Sharma · 13 years ago
  12. c311933 Remove legacy http support from stagefright, chromium is the new hotness. by Andreas Huber · 13 years ago
  13. 603d739 Charge network traffic to the uid of the process using the MediaPlayer. by Andreas Huber · 13 years ago
  14. d4a6cac Revert "Parse RTP-Info even for live streams." by Andreas Huber · 13 years ago
  15. d873413 Parse RTP-Info even for live streams. by Andreas Huber · 13 years ago
  16. 27db53d Derive the Transport "source" attribute from the RTSP endpoint address if necessary by Andreas Huber · 13 years ago
  17. 0407269 Work around several issues with non-compliant RTSP servers. by Andreas Huber · 13 years ago
  18. b2934b1 Change timestamp handling in RTSP, remove unused, experimental, gtalk support by Andreas Huber · 13 years ago
  19. a2edd7d8 More robust parsing of NPT time ranges in RTSP. by Andreas Huber · 14 years ago
  20. a0b442e Support for BASIC and DIGEST authentication schemes in RTSP. Support for malformed packet descriptions that end lines in LF only, instead of CRLF. by Andreas Huber · 14 years ago
  21. dc0728f am 8e4f3c76: am 646e0d5a: Merge "Some webcams output rtp streams but never send any rtcp data in violation of the specs. Attempt to use fake timestamps to be able to play these..." into gingerbread by Andreas Huber · 14 years ago
  22. cc5fb1d Some webcams output rtp streams but never send any rtcp data in violation of by Andreas Huber · 14 years ago
  23. fefcc9c am 5b0d0630: am 1010da2e: Merge "Just in case we\'re behind a NAT router/firewall, attempt to poke holes into it for future incoming RTP/RTCP packets to pass through." into gingerbread by Andreas Huber · 14 years ago
  24. 0dc6403 Just in case we're behind a NAT router/firewall, attempt to poke holes into it for future incoming RTP/RTCP packets to pass through. by Andreas Huber · 14 years ago
  25. aee02b8 am cac43e8a: am beffefa2: Merge "RTSP seeking is now asynchronous, MediaPlayer is not notified that the seek is complete until it actually is. Ignore seek requests on live streams." into gingerbread by Andreas Huber · 14 years ago
  26. 2ddbd7d am 14ea1048: am c5912acc: Merge "Disable the access unit timeout temporarily while a seek operation is in progress." into gingerbread by Andreas Huber · 14 years ago
  27. 0c46b69 RTSP seeking is now asynchronous, MediaPlayer is not notified that the seek is complete until it actually is. Ignore seek requests on live streams. by Andreas Huber · 14 years ago
  28. e51e809 Disable the access unit timeout temporarily while a seek operation is in progress. by Andreas Huber · 14 years ago
  29. 41fd0a0 am af909581: am 67738486: Merge "Remove stagefright foundation\'s incompatible logging interface and update callsites." into gingerbread by Andreas Huber · 14 years ago
  30. 6e3fa44 Remove stagefright foundation's incompatible logging interface and update callsites. by Andreas Huber · 14 years ago
  31. 4d8024c am 7ff94577: am 9909b948: Merge "Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting." into gingerbread by Andreas Huber · 14 years ago
  32. f3d2bdf Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting. by Andreas Huber · 14 years ago
  33. 5f39972 am e1a3cddd: am 99fa510e: Merge "Better detection of connection problems - timeout if no rtcp packets arrive within a certain time, not a final frame (which may take longer)" into gingerbread by Andreas Huber · 14 years ago
  34. 3a48d4d Better detection of connection problems - timeout if no rtcp packets arrive within a certain time, not a final frame (which may take longer) by Andreas Huber · 14 years ago
  35. 6924563 am 987556bc: am abb8398e: Merge "Instead of closing the connection altogether if no UDP packets arrive after a certain time, try changing transports (to interleaved TCP). Also properly close the sockets on disconnection." into gingerbread by Andreas Huber · 14 years ago
  36. f88ca7a0 Instead of closing the connection altogether if no UDP packets arrive after a certain time, try changing transports (to interleaved TCP). Also properly close the sockets on disconnection. by Andreas Huber · 14 years ago
  37. 3678668 am 7ed9104c: am f6639c46: Finetune some rtsp timeout constants. by Andreas Huber · 14 years ago
  38. 631025e am 6df6d606: am df992ac9: Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 14 years ago
  39. f6639c4 Finetune some rtsp timeout constants. by Andreas Huber · 14 years ago
  40. 8abd425 am 05c1cada: am 577615c9: Merge "Better support for rtsp (normal play-)time display. Better seek support, timeout if no packets arrive for too long." into gingerbread by Andreas Huber · 14 years ago
  41. 84ecebb am e25e0361: am e250c220: Merge "We accidentally always aborted after 10 secs, even if the connection was fine." into gingerbread by Andreas Huber · 14 years ago
  42. c4e0b70 ALoopers can now be named (useful to distinguish threads). by Andreas Huber · 14 years ago
  43. eeb97d9 Better support for rtsp (normal play-)time display. Better seek support, timeout if no packets arrive for too long. by Andreas Huber · 14 years ago
  44. d6a4004 We accidentally always aborted after 10 secs, even if the connection was fine. by Andreas Huber · 14 years ago
  45. 178e1d0 am 74ae6973: am 17a765a1: Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 14 years ago
  46. 0416da7 Support for RTP packets arriving interleaved with RTSP responses. by Andreas Huber · 14 years ago
  47. 8d9d751 am 67ca90b3: am 6b6ae996: Merge "A first shot at proper support for seeking of rtsp streams." into gingerbread by Andreas Huber · 14 years ago
  48. e0dd7d3 A first shot at proper support for seeking of rtsp streams. by Andreas Huber · 14 years ago
  49. 804539b am 31e71131: am 3e22ef1e: Merge "Better handling of rtsp connection and disconnection." into gingerbread by Andreas Huber · 14 years ago
  50. 8370be1 Better handling of rtsp connection and disconnection. by Andreas Huber · 14 years ago
  51. a6a0fe4 am 6bcffcd2: am 8c192fe9: Merge "Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description." into gingerbread by Andreas Huber · 14 years ago
  52. af063a6 Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description. by Andreas Huber · 14 years ago
  53. 0a3858b am 1f513d88: am c17f35dd: Merge "Support for Gtalk video, includes AMR/H.263 assembler and packetization support, extensions to MediaRecorder to stream via RTP over a pair of UDP sockets as well as various fixes to the RTP implementation." into gingerbread by Andreas Huber · 14 years ago
  54. 57648e4 Support for Gtalk video, includes AMR/H.263 assembler and packetization support, extensions to MediaRecorder to stream via RTP over a pair of UDP sockets as well as various fixes to the RTP implementation. by Andreas Huber · 14 years ago
  55. 8dac3bf am b72d3180: am 81046c8c: Merge "Various changes to improve rtsp networking, reduce packet loss and adapt to ALooper API changes." into gingerbread by Andreas Huber · 14 years ago
  56. 4e4173b Various changes to improve rtsp networking, reduce packet loss and adapt to ALooper API changes. by Andreas Huber · 14 years ago
  57. 5a23f8c Fixes for simulator build on lucid by Mike Lockwood · 14 years ago
  58. 7a747b8 Initial checkin of preliminary rtsp support for stagefright. by Andreas Huber · 14 years ago