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gerrit-public.fairphone.software / fp2-dev / platform / frameworks / av / ecba7488c5cceaacef95836b1b476433451e7761 / media / libstagefright / rtsp
  1. 190cdba Identify network servers and clients with a OS version related string by Andreas Huber · 11 years ago
  2. 4f4c265 am 59ac7b30: am 66abe3ae: Merge "Fix for crash if no content in DESCRIBE response" by Andreas Huber · 11 years ago
  3. d32b7b4 Fix for crash if no content in DESCRIBE response by Xuefei Chen · 12 years ago
  4. 0955986 Avoid rebuffering after RTSP pause by Roger Jönsson · 12 years ago
  5. 1a37ee3 EOS fixes for RTSP streams by joakim johansson · 12 years ago
  6. b6ec588 RTSP: Parse session level control attribute from SDP by Måns Zigher · 12 years ago
  7. 46d13e3 Enable pause/resume for RTSP streaming by Roger Jönsson · 12 years ago
  8. cfc3083 RTSP buffering improvements by Roger Jönsson · 12 years ago
  9. 7f475c3 RTSP now properly publishes its "seekable" flags after connection by Andreas Huber · 11 years ago
  10. ec29a2b Detect live streams by Roger Jönsson · 12 years ago
  11. 81dd60e Added HTTP support for SDP files. by Oscar Rydhé · 12 years ago
  12. cc4e609 Merge "Use default values when MPEG4 audio config parsing fails." by James Dong · 12 years ago
  13. b90b748 Fix bad checks that causes crash when streaming H.263 content. by Roger1 Jonsson · 14 years ago
  14. 78cc49b Unsolicited server responses cause RTSP streaming to crash by Lena Magnusson · 13 years ago
  15. e1a31d1 Crash in android::MyHandler::parsePlayResponse by Patric Frederiksen · 13 years ago
  16. a45a600 Use default values when MPEG4 audio config parsing fails. by Erik Rydgren · 13 years ago
  17. fa0e033 ALooper::GetNowUs() now relies on systemTime instead of gettimeofday. by Andreas Huber · 12 years ago
  18. 4969468 Add support for mpeg2 transport streams to the RTSP implementation. by Andreas Huber · 12 years ago
  19. 3677437 Fixed member access into incomplete type build error by Tareq A. Siraj · 12 years ago
  20. 8033393 h264 streaming: make profile-level-id optional by Patrik2 Carlsson · 12 years ago
  21. 3d51d5c Add NOTICE and MODULE_LICENSE_APACH2 to libs build under /frameworks/av/ by James Dong · 12 years ago
  22. 8647bbe Prefix MPEG4-generic audio data with ADTS headers by Andreas Huber · 12 years ago
  23. f95439a Changes to add support for H263-1999/2000 formats for streaming by Andreas Huber · 12 years ago
  24. 559bf28 AV Android make files changes by James Dong · 12 years ago
  25. 3ee2694 Remove JNI in LOCAL_C_INCLUDE from non-JNI related Android.mk files. by James Dong · 12 years ago
  26. 6c6b4d0 Switched to use the header files in /frameworks/native by James Dong · 12 years ago
  27. 2d8bedd Add new APIs AMessage::(set|find)Buffer to make it safer to pass by Andreas Huber · 12 years ago
  28. 7e73e44 Starhub RTSP apparently does not establish time on all tracks by Andreas Huber · 13 years ago
  29. 29357bc Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF) DO NOT MERGE by Steve Block · 13 years ago
  30. 5ff1dd5 Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGE by Steve Block · 13 years ago
  31. df64d15 Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGE by Steve Block · 13 years ago
  32. 6af1e76 Merge "Support for parsing of "folded" RTSP header values and some tweaks to the AMR assembler" by Andreas Huber · 13 years ago
  33. aa5ba9a am aa82c39b: am 0ba9380a: Merge "Fix Bitreader "putBits" implementation, make sure we emulate timestamps" into ics-mr1 by Andreas Huber · 13 years ago
  34. 4aae77c Support for parsing of "folded" RTSP header values and some tweaks to the AMR assembler by Andreas Huber · 13 years ago
  35. 1906e5c Fix Bitreader "putBits" implementation, make sure we emulate timestamps by Andreas Huber · 13 years ago
  36. 78ff828 am 8a065423: am 23217182: Merge "Didn\'t mean to check this in..." into ics-mr1 by Andreas Huber · 13 years ago
  37. 516fb1d 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
  38. a36d8ca am 9c981cd3: am d9f25bc9: Merge "Disconnect on socket error on the RTSP control connection." into ics-mr1 by Andreas Huber · 13 years ago
  39. 91f2304 Didn't mean to check this in... by Andreas Huber · 13 years ago
  40. 73b1fd5 Merge "Instead of asserting, signal a runtime error if the session doesn't contain" into ics-mr1 by Andreas Huber · 13 years ago
  41. 4ab3045 Merge "DO NOT MERGE: Instead of asserting, remove active streams if their sockets" into ics-mr1 by Andreas Huber · 13 years ago
  42. 0fbe057 Disconnect on socket error on the RTSP control connection. by Andreas Huber · 13 years ago
  43. 19de627 DO NOT MERGE: Instead of asserting, remove active streams if their sockets by Andreas Huber · 13 years ago
  44. f0c86a8 Instead of asserting, signal a runtime error if the session doesn't contain by Andreas Huber · 13 years ago
  45. 7cad0b4 am 9e2949c6: am 2375d163: Merge "Send RTSP control connection keep-alive requests" into ics-mr1 by Andreas Huber · 13 years ago
  46. 8c308ff Instead of asserting, remove active streams if their sockets return failure by Andreas Huber · 13 years ago
  47. 908dbde Send RTSP control connection keep-alive requests by Andreas Huber · 13 years ago
  48. 3856b09 Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE by Steve Block · 13 years ago
  49. 2bfdd42 NuPlayer is now taking on the task of streaming over RTSP. by Andreas Huber · 13 years ago
  50. a23456b Network traffic accounting for chromium stack support in mediaserver. by Ashish Sharma · 13 years ago
  51. f89d780 Eliminate superfluous memcpys by wrapping an ABuffer in a MediaBuffer by Andreas Huber · 13 years ago
  52. dab718b Remove legacy http support from stagefright, chromium is the new hotness. by Andreas Huber · 13 years ago
  53. 9b80c2b Charge network traffic to the uid of the process using the MediaPlayer. by Andreas Huber · 13 years ago
  54. ac5767a Revert "Parse RTP-Info even for live streams." by Andreas Huber · 13 years ago
  55. a6925e6 Parse RTP-Info even for live streams. by Andreas Huber · 13 years ago
  56. 386d609 Support mpeg1,2 audio and mpeg1,2,4 video content extraction from .ts streams. by Andreas Huber · 13 years ago
  57. e681b91 Add a user-agent header to our RTSP requests. by Andreas Huber · 13 years ago
  58. fcea8f7 Support for PCMA and PCMU raw audio data in RTP/RTSP. by Andreas Huber · 13 years ago
  59. 55e2619 Support more MPEG4-LATM audio functionality. by Andreas Huber · 13 years ago
  60. 5ef1521 Respond to RTSP server->client requests. by Andreas Huber · 13 years ago
  61. de9a20c Derive the Transport "source" attribute from the RTSP endpoint address if necessary by Andreas Huber · 13 years ago
  62. dc468c5 Work around several issues with non-compliant RTSP servers. by Andreas Huber · 13 years ago
  63. f1958f9 Enable cancelling the rtsp connection process early. by Andreas Huber · 13 years ago
  64. 864d066 Fix the build. by Andreas Huber · 13 years ago
  65. 100a440 Change timestamp handling in RTSP, remove unused, experimental, gtalk support by Andreas Huber · 13 years ago
  66. 783e5cd More robust parsing of NPT time ranges in RTSP. by Andreas Huber · 13 years ago
  67. 9202cca This particular RTSP server streams MPEG4-LATM audio with extra trailing bytes. by Andreas Huber · 13 years ago
  68. 21a6f9f Implement parsing of vbv buffering info in RTSP. by Andreas Huber · 14 years ago
  69. 934ca8c Fail to parse duration instead of asserting, if the server response cannot be parsed. by Andreas Huber · 14 years ago
  70. 674ebd0 Removed uncessary FILE structure pointer for I/O by James Dong · 14 years ago
  71. fc9ac98 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
  72. c211436 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
  73. 4579b7d 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
  74. 8ac0cb9 Merge fb474872 from gingerbread-plus-aosp by Jean-Baptiste Queru · 14 years ago
  75. 56cfa23 Include the framework copy of the OpenMAX headers instead of referencing external/opencore. by Andreas Huber · 14 years ago
  76. a44501e 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
  77. f61551f Some webcams output rtp streams but never send any rtcp data in violation of by Andreas Huber · 14 years ago
  78. 43a2b3b 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
  79. 2bc940b 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
  80. 250e051 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
  81. e31aa74 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
  82. 1c8ef86 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
  83. 0dcd837 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
  84. c68a48c 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
  85. a9d9dd2 Disable the access unit timeout temporarily while a seek operation is in progress. by Andreas Huber · 14 years ago
  86. 3f94dac am af909581: am 67738486: Merge "Remove stagefright foundation\'s incompatible logging interface and update callsites." into gingerbread by Andreas Huber · 14 years ago
  87. 6e4c5c4 Remove stagefright foundation's incompatible logging interface and update callsites. by Andreas Huber · 14 years ago
  88. ac5f724 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
  89. 6f85dba Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting. by Andreas Huber · 14 years ago
  90. 6faf0cd 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
  91. c9e8948 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
  92. 56f2c6e am 47f2cf62: am 412fc7cd: Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 14 years ago
  93. 389636c Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 14 years ago
  94. 3ef9f98 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
  95. 0612475 Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 14 years ago
  96. 16c4e8c 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
  97. 4dba3e9 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 14 years ago
  98. e7d3e90 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
  99. ca999e0 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
  100. 5d5f5df 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