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review.blissroms.org / platform_frameworks_av / e3ae15e0ccee7b57a8e8f2cb283b041e97607755
  1. e3ae15e am f3de053c: Merge "Instead of asserting return a runtime error if the maximum sample size cannot be determined." into gingerbread by Andreas Huber · 15 years ago
  2. de32b0f am 5c43a7af: Merge "When 32-bit offset is used, if the requested max file size is greater than the 32-bit offset limit, set the limit to the max 32-bit offset limit." into gingerbread by James Dong · 15 years ago
  3. 31d2a4b Merge "Instead of asserting return a runtime error if the maximum sample size cannot be determined." into gingerbread by Andreas Huber · 15 years ago
  4. 4c73f1f Merge "When 32-bit offset is used, if the requested max file size is greater than the 32-bit offset limit, set the limit to the max 32-bit offset limit." into gingerbread by James Dong · 15 years ago
  5. 49110ce Instead of asserting return a runtime error if the maximum sample size cannot be determined. by Andreas Huber · 15 years ago
  6. 84cd8ad am a063cd64: Merge "Instead of asserting, publish no tracks if an MP3Extractor is used on non-mp3 content." into gingerbread by Andreas Huber · 15 years ago
  7. 772bcc2 Instead of asserting, publish no tracks if an MP3Extractor is used on non-mp3 content. by Andreas Huber · 15 years ago
  8. d2518e0 When 32-bit offset is used, by James Dong · 15 years ago
  9. 368b56e am d353c840: Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 15 years ago
  10. fbf7162 Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 15 years ago
  11. 3c3763d HW audio encoder expects timestamp via kKeyTime from each input buffer by James Dong · 15 years ago
  12. e92e213 am 95d86480: Merge "Modify type of some environmental reverb parameters" into gingerbread by Eric Laurent · 15 years ago
  13. 54c38fd Modify type of some environmental reverb parameters by Eric Laurent · 15 years ago
  14. aa8d119 am 7e427934: Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 15 years ago
  15. f9c0ae8 Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 15 years ago
  16. 77682db am 9077f8ec: Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 15 years ago
  17. ddba3f0 Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 15 years ago
  18. 2d3bf53 LVM release 1.08 delivery. by Eric Laurent · 15 years ago
  19. 1d816a9 am 9fee0b2a: Ogg files can be tagged to be automatically looping, this setting always overrides the MediaPlayer\'s setLooping setting. by Andreas Huber · 15 years ago
  20. 8ae49d8 Ogg files can be tagged to be automatically looping, this setting always overrides the MediaPlayer's setLooping setting. by Andreas Huber · 15 years ago
  21. 6f6bc92 am cc4a38c6: Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 15 years ago
  22. 1a4c79e Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 15 years ago
  23. 8650e19 Properly buffer a certain amount of data on streaming sources before finishing prepare(). by Andreas Huber · 15 years ago
  24. caa68a5 Not all audio source has the drift time information by James Dong · 15 years ago
  25. 52c006e am 7755cdd6: Remove unused/debugging code from MP4 file writer by James Dong · 15 years ago
  26. b4d5320 Remove unused/debugging code from MP4 file writer by James Dong · 15 years ago
  27. de428f1 am 46e63b34: Merge "Better file size estimate" into gingerbread by James Dong · 15 years ago
  28. 1f90c4b Better file size estimate by James Dong · 15 years ago
  29. ea0fe65 am 7ed7668b: Merge "Calculate audio media drift time from AudioSource" into gingerbread by James Dong · 15 years ago
  30. bd05775 Merge "Calculate audio media drift time from AudioSource" into gingerbread by James Dong · 15 years ago
  31. a526824 am 32ec1ad1: Merge "Fix problem in AudioEffect::command() status." into gingerbread by Eric Laurent · 15 years ago
  32. 34c8d61 Merge "Fix problem in AudioEffect::command() status." into gingerbread by Eric Laurent · 15 years ago
  33. aeae3de Fix problem in AudioEffect::command() status. by Eric Laurent · 15 years ago
  34. d707fcb Calculate audio media drift time from AudioSource by James Dong · 15 years ago
  35. 031ecf3 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 · 15 years ago
  36. 4cd45f8 am d3c1bae4: Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 15 years ago
  37. 9b93478 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 · 15 years ago
  38. e91b46246 Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 15 years ago
  39. 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 · 15 years ago
  40. 6e20bdf Make sure that if initialization fails, AudioSource still behaves well. by James Dong · 15 years ago
  41. 002b34c am 412fc7cd: Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 15 years ago
  42. bcbe5af Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 15 years ago
  43. d8c48ad am de2b1615: Merge "Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer." into gingerbread by Andreas Huber · 15 years ago
  44. 82f7321 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. by Andreas Huber · 15 years ago
  45. 389636c Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 15 years ago
  46. 2e0448f am f560ceab: Merge "Audio Effects: fix problems in volume control." into gingerbread by Eric Laurent · 15 years ago
  47. 8f45bd7 Audio Effects: fix problems in volume control. by Eric Laurent · 15 years ago
  48. dc344e5 am 48ac68e1: Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 15 years ago
  49. 0612475 Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 15 years ago
  50. 07e0c92 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 · 15 years ago
  51. 69a4f8b 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 · 15 years ago
  52. 4dba3e9 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 15 years ago
  53. eaae2c3 am 6aacad66: Merge "Add some encoding parameters for the "record" utility" into gingerbread by James Dong · 15 years ago
  54. f74c8f9 Add some encoding parameters for the "record" utility by James Dong · 15 years ago
  55. 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 · 15 years ago
  56. eebcf36 am 12006013: fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 15 years ago
  57. 5edae61 fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 15 years ago
  58. d81ef83 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 · 15 years ago
  59. 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 · 15 years ago
  60. b186054 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 · 15 years ago
  61. f594c64 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 · 15 years ago
  62. e26cd86 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 · 15 years ago
  63. 7aef033 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 · 15 years ago
  64. da701fe am ae6bdc23: Merge "Fix issue 2952766." into gingerbread by Eric Laurent · 15 years ago
  65. 44eb096 Merge "Fix issue 2952766." into gingerbread by Eric Laurent · 15 years ago
  66. fabf86f am 7ec7b997: Remove camera metering mode API. by Wu-cheng Li · 15 years ago
  67. 541d765 Remove camera metering mode API. by Wu-cheng Li · 15 years ago
  68. e83fffc am 681c5ff2: Merge "Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore." into gingerbread by Andreas Huber · 15 years ago
  69. 1c842b2 Merge "Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore." into gingerbread by Andreas Huber · 15 years ago
  70. a1ffe49 Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore. by Andreas Huber · 15 years ago
  71. d2ab607 am 858bb4f6: Merge "LVM release 1.07 delivery." into gingerbread by Eric Laurent · 15 years ago
  72. bf5606b am f6639c46: Finetune some rtsp timeout constants. by Andreas Huber · 15 years ago
  73. c28160f am df992ac9: Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 15 years ago
  74. 3849699 Merge "LVM release 1.07 delivery." into gingerbread by Eric Laurent · 15 years ago
  75. e56121b Finetune some rtsp timeout constants. by Andreas Huber · 15 years ago
  76. c01ec02 Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 15 years ago
  77. c1c88e2 Fix issue 2952766. by Eric Laurent · 15 years ago
  78. a814c1f ALoopers can now be named (useful to distinguish threads). by Andreas Huber · 15 years ago
  79. b354e79 am df8356ff: Merge "Workaround for a QCOM issue where the output buffer size advertised by the AVC encoder is occasionally too small." into gingerbread by James Dong · 15 years ago
  80. 824c9ff Workaround for a QCOM issue where the output buffer size advertised by the AVC encoder by James Dong · 15 years ago
  81. 23f0d68 am b86365ad: Merge "Suppress the video recording start signal - bug 2950297" into gingerbread by James Dong · 15 years ago
  82. 352c468 Merge "Suppress the video recording start signal - bug 2950297" into gingerbread by James Dong · 15 years ago
  83. 368b3ed 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 · 15 years ago
  84. 6adecf4 am e250c220: Merge "We accidentally always aborted after 10 secs, even if the connection was fine." into gingerbread by Andreas Huber · 15 years ago
  85. f8860bf 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 · 15 years ago
  86. 165dc4c Merge "We accidentally always aborted after 10 secs, even if the connection was fine." into gingerbread by Andreas Huber · 15 years ago
  87. 8d34297 Better support for rtsp (normal play-)time display. Better seek support, timeout if no packets arrive for too long. by Andreas Huber · 15 years ago
  88. 95fd60f am ae66946b: Merge "fix a race in SF buffer management" into gingerbread by Mathias Agopian · 15 years ago
  89. d918324 LVM release 1.07 delivery. by Eric Laurent · 15 years ago
  90. 14cc6fc Merge "fix a race in SF buffer management" into gingerbread by Mathias Agopian · 15 years ago
  91. cc6adf5 We accidentally always aborted after 10 secs, even if the connection was fine. by Andreas Huber · 15 years ago
  92. d7c43d3 fix a race in SF buffer management by Mathias Agopian · 15 years ago
  93. f1ae196 Suppress the video recording start signal - bug 2950297 by James Dong · 15 years ago
  94. f39928d am 17a765a1: Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 15 years ago
  95. 1b07372 Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 15 years ago
  96. 0792ce7 Support for RTP packets arriving interleaved with RTSP responses. by Andreas Huber · 15 years ago
  97. 64b531e am 318a759e: Merge "Make sure that timestamp does not go backward in MP4 file writer" into gingerbread by James Dong · 15 years ago
  98. 640a72e Merge "Make sure that timestamp does not go backward in MP4 file writer" into gingerbread by James Dong · 15 years ago
  99. 90e4b45 am 8ac0983e: Merge "Fix support for per-frame unsynchronization in ID3V2.4 tags." into gingerbread by Andreas Huber · 15 years ago
  100. f302743 Merge "Fix support for per-frame unsynchronization in ID3V2.4 tags." into gingerbread by Andreas Huber · 15 years ago