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gerrit-public.fairphone.software / platform / frameworks / base / ae7752798a98fc81ff5e6ae69dde2137692106be / media / libstagefright / rtsp
  1. b672373 Support for PCMA and PCMU raw audio data in RTP/RTSP. by Andreas Huber · 13 years ago
  2. 8c7c6dc Support more MPEG4-LATM audio functionality. by Andreas Huber · 13 years ago
  3. e066616 Respond to RTSP server->client requests. by Andreas Huber · 13 years ago
  4. 27db53d Derive the Transport "source" attribute from the RTSP endpoint address if necessary by Andreas Huber · 13 years ago
  5. 0407269 Work around several issues with non-compliant RTSP servers. by Andreas Huber · 13 years ago
  6. cb21879 Enable cancelling the rtsp connection process early. by Andreas Huber · 13 years ago
  7. c2f9a26 Fix the build. by Andreas Huber · 13 years ago
  8. b2934b1 Change timestamp handling in RTSP, remove unused, experimental, gtalk support by Andreas Huber · 13 years ago
  9. a2edd7d8 More robust parsing of NPT time ranges in RTSP. by Andreas Huber · 13 years ago
  10. 0f535af This particular RTSP server streams MPEG4-LATM audio with extra trailing bytes. by Andreas Huber · 13 years ago
  11. 77034e6 Implement parsing of vbv buffering info in RTSP. by Andreas Huber · 13 years ago
  12. 4bca5e1 Fail to parse duration instead of asserting, if the server response cannot be parsed. by Andreas Huber · 14 years ago
  13. 2747e0e Removed uncessary FILE structure pointer for I/O by James Dong · 14 years ago
  14. b0d25a0 Better support for MP4A-LATM RTP disassembly. This used to fail if mNumSubFrames > 1 and the sub frames did not align with RTP packet boundaries. by Andreas Huber · 14 years ago
  15. cf74754 We don't have access to the md5 implementation on the simulator, let's disable digest authentication in rtsp for simulator targets. by Andreas Huber · 14 years ago
  16. 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
  17. 5582cc3 Merge fb474872 from gingerbread-plus-aosp by Jean-Baptiste Queru · 14 years ago
  18. a4f391c Include the framework copy of the OpenMAX headers instead of referencing external/opencore. by Andreas Huber · 14 years ago
  19. 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
  20. cc5fb1d Some webcams output rtp streams but never send any rtcp data in violation of by Andreas Huber · 14 years ago
  21. 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
  22. 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
  23. 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
  24. 023266c am e0c8545a: am 0fd4e216: Merge "Refactor some more h.264 utility code out into avc_utils. Work around a hardware decoder issue by making sure the first access unit submitted to a decoder at startup or after seek is an IDR." into gingerbread by Andreas Huber · 14 years ago
  25. 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
  26. 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
  27. 38285db Refactor some more h.264 utility code out into avc_utils. Work around a hardware decoder issue by making sure the first access unit submitted to a decoder at startup or after seek is an IDR. 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. 53d7765 am fd0eed00: am a2511da9: Merge "Better support for buffered streaming of rtsp content, if buffer drops below a certain threshold we will temporarily pause playback until we have sufficient data." into gingerbread by Andreas Huber · 14 years ago
  34. 4d8f66b Better support for buffered streaming of rtsp content, if buffer drops below a certain threshold we will temporarily pause playback until we have sufficient data. by Andreas Huber · 14 years ago
  35. 52d14be am 47f2cf62: am 412fc7cd: Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 14 years ago
  36. 27b9c8e Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 14 years ago
  37. 03cf220 am 6b52911c: am 48ac68e1: Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 14 years ago
  38. 48ac68e Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 14 years ago
  39. 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
  40. e536f80 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 14 years ago
  41. 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
  42. 47416bc am 03e83d4a: am 68ae91cb: Merge "Recent changes to the rtsp code require every buffer fed to the packet source to have a timestamp, we\'re ignoring timestamps for gtalk videochat but we still have to have a placeholder." into gingerbread by Andreas Huber · 14 years ago
  43. 68ae91c Merge "Recent changes to the rtsp code require every buffer fed to the packet source to have a timestamp, we're ignoring timestamps for gtalk videochat but we still have to have a placeholder." into gingerbread by Andreas Huber · 14 years ago
  44. 0ddf8c0 Recent changes to the rtsp code require every buffer fed to the packet source to have a timestamp, we're ignoring timestamps for gtalk videochat but we still have to have a placeholder. by Andreas Huber · 14 years ago
  45. 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
  46. 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
  47. 3678668 am 7ed9104c: am f6639c46: Finetune some rtsp timeout constants. by Andreas Huber · 14 years ago
  48. 631025e am 6df6d606: am df992ac9: Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 14 years ago
  49. f6639c4 Finetune some rtsp timeout constants. by Andreas Huber · 14 years ago
  50. 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
  51. 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
  52. c4e0b70 ALoopers can now be named (useful to distinguish threads). by Andreas Huber · 14 years ago
  53. 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
  54. d6a4004 We accidentally always aborted after 10 secs, even if the connection was fine. by Andreas Huber · 14 years ago
  55. 178e1d0 am 74ae6973: am 17a765a1: Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 14 years ago
  56. 0416da7 Support for RTP packets arriving interleaved with RTSP responses. by Andreas Huber · 14 years ago
  57. 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
  58. e0dd7d3 A first shot at proper support for seeking of rtsp streams. by Andreas Huber · 14 years ago
  59. 804539b am 31e71131: am 3e22ef1e: Merge "Better handling of rtsp connection and disconnection." into gingerbread by Andreas Huber · 14 years ago
  60. 8370be1 Better handling of rtsp connection and disconnection. by Andreas Huber · 14 years ago
  61. 349250f am c8d2fa70: am cbd038fe: Merge "Make MediaWriter stop and pause return errors if necessary" into gingerbread by James Dong · 14 years ago
  62. 83b3e35 am 873ebfb8: am 223e4f73: Merge "Support for MP4V-ES packetization format according to RFC3016." into gingerbread by Andreas Huber · 14 years ago
  63. fc5d0cf am b29ebd39: am f0ad5484: Merge "In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data." into gingerbread by Andreas Huber · 14 years ago
  64. 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
  65. cbd038f Merge "Make MediaWriter stop and pause return errors if necessary" into gingerbread by James Dong · 14 years ago
  66. d036662 Make MediaWriter stop and pause return errors if necessary by James Dong · 14 years ago
  67. a979ad6 Support for MP4V-ES packetization format according to RFC3016. by Andreas Huber · 14 years ago
  68. eef3c33 In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data. by Andreas Huber · 14 years ago
  69. 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
  70. 08c96ec am 3bf8c342: am ae3a1f45: Merge "Fix the h.263 assembler to properly subset a buffer\'s range if it already has a range applied." into gingerbread by Andreas Huber · 14 years ago
  71. 53fb14f am 53895c6a: am 66aa0f3d: Merge "APacketSource is too verbose." into gingerbread by Andreas Huber · 14 years ago
  72. 00237b7 Fix the h.263 assembler to properly subset a buffer's range if it already has a range applied. by Andreas Huber · 14 years ago
  73. 3f55576 APacketSource is too verbose. by Andreas Huber · 14 years ago
  74. 3965148 am 4dc41bb4: am 18f0174f: Merge "We\'re now going to ignore timestamps completely in gtalk video conferencing, playing video as soon as it comes in. We also make up fake timestamps in the rtp code, ignoring rtcp SR information to enable early startup." into gingerbrea by Andreas Huber · 14 years ago
  75. f88f844 We're now going to ignore timestamps completely in gtalk video conferencing, playing video as soon as it comes in. We also make up fake timestamps in the rtp code, ignoring rtcp SR information to enable early startup. by Andreas Huber · 14 years ago
  76. 2184abf am 870678a9: am 2c37f3d3: Merge "Better support for fake timestamps in RTP, H.263 video now also requests FIR." into gingerbread by Andreas Huber · 14 years ago
  77. b2b1a2d am c6d1519e: am fb861523: Merge "Specification of codec specific data as part of the session description is now optional." into gingerbread by Andreas Huber · 14 years ago
  78. 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
  79. 3eaa300 Better support for fake timestamps in RTP, H.263 video now also requests FIR. by Andreas Huber · 14 years ago
  80. 426b650 Specification of codec specific data as part of the session description is now optional. by Andreas Huber · 14 years ago
  81. 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
  82. 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
  83. 4e4173b Various changes to improve rtsp networking, reduce packet loss and adapt to ALooper API changes. by Andreas Huber · 14 years ago
  84. 5a23f8c Fixes for simulator build on lucid by Mike Lockwood · 14 years ago
  85. 7a747b8 Initial checkin of preliminary rtsp support for stagefright. by Andreas Huber · 14 years ago