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review.evervolv.com / android_frameworks_av / 1a02fabca518806222212bdc708b36a37a7c3925 / media
  1. 1a02fab Merge "Adding hasProfile." by Nipun Kwatra · 16 years ago
  2. a0d472f Merge "Added specific resolution and time lapse profiles." by Nipun Kwatra · 16 years ago
  3. 8bb5603 Adding hasProfile. by Nipun Kwatra · 16 years ago
  4. c0a8478 Added specific resolution and time lapse profiles. by Nipun Kwatra · 16 years ago
  5. 275c732 am ea4680f0: am 2814ad25: Merge "LVM release 1.09 delivery" into gingerbread by Eric Laurent · 16 years ago
  6. e2af4f7 am 2814ad25: Merge "LVM release 1.09 delivery" into gingerbread by Eric Laurent · 16 years ago
  7. a175413 Merge "LVM release 1.09 delivery" into gingerbread by Eric Laurent · 16 years ago
  8. a01b794 am 0d661b5e: am ff4c57ad: Merge "TimedEventQueue now explicitly sets its scheduling policy to foreground as it should." into gingerbread by Andreas Huber · 16 years ago
  9. f391593 am ff4c57ad: Merge "TimedEventQueue now explicitly sets its scheduling policy to foreground as it should." into gingerbread by Andreas Huber · 16 years ago
  10. acb5621 TimedEventQueue now explicitly sets its scheduling policy to foreground as it should. by Andreas Huber · 16 years ago
  11. 1999787 am 8986f62b: am f3de053c: Merge "Instead of asserting return a runtime error if the maximum sample size cannot be determined." into gingerbread by Andreas Huber · 16 years ago
  12. ea16215 am e19df3dc: 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 · 16 years ago
  13. 1432a00 am f0eab98c: am a063cd64: Merge "Instead of asserting, publish no tracks if an MP3Extractor is used on non-mp3 content." into gingerbread by Andreas Huber · 16 years ago
  14. 5185b01 LVM release 1.09 delivery by Eric Laurent · 16 years ago
  15. e3ae15e am f3de053c: Merge "Instead of asserting return a runtime error if the maximum sample size cannot be determined." into gingerbread by Andreas Huber · 16 years ago
  16. 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 · 16 years ago
  17. 31d2a4b Merge "Instead of asserting return a runtime error if the maximum sample size cannot be determined." into gingerbread by Andreas Huber · 16 years ago
  18. 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 · 16 years ago
  19. 49110ce Instead of asserting return a runtime error if the maximum sample size cannot be determined. by Andreas Huber · 16 years ago
  20. 84cd8ad am a063cd64: Merge "Instead of asserting, publish no tracks if an MP3Extractor is used on non-mp3 content." into gingerbread by Andreas Huber · 16 years ago
  21. 772bcc2 Instead of asserting, publish no tracks if an MP3Extractor is used on non-mp3 content. by Andreas Huber · 16 years ago
  22. 9b2cef2 am 306cd58d: am d353c840: Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 16 years ago
  23. d2518e0 When 32-bit offset is used, by James Dong · 16 years ago
  24. 368b56e am d353c840: Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 16 years ago
  25. fbf7162 Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 16 years ago
  26. 3c3763d HW audio encoder expects timestamp via kKeyTime from each input buffer by James Dong · 16 years ago
  27. 87e6079 am e126119c: am 95d86480: Merge "Modify type of some environmental reverb parameters" into gingerbread by Eric Laurent · 16 years ago
  28. e92e213 am 95d86480: Merge "Modify type of some environmental reverb parameters" into gingerbread by Eric Laurent · 16 years ago
  29. 54c38fd Modify type of some environmental reverb parameters by Eric Laurent · 16 years ago
  30. 5de4999 Don't assert on unexpected surface flinger dequeue/enqueueBuffer errors, log a warning and ignore them instead. by Andreas Huber · 16 years ago
  31. 5d7d90a Merge changes I71f5b0fc,I92c7accb by Nipun Kwatra · 16 years ago
  32. 155e833 Moving decision to use still camera to CameraSourceTimeLapse by Nipun Kwatra · 16 years ago
  33. 4d3f97e am ef9e508c: am 7e427934: Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 16 years ago
  34. 734c3c9 am d6fd133d: am 9077f8ec: Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 16 years ago
  35. aa8d119 am 7e427934: Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 16 years ago
  36. f9c0ae8 Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 16 years ago
  37. 77682db am 9077f8ec: Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 16 years ago
  38. ddba3f0 Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 16 years ago
  39. 2d3bf53 LVM release 1.08 delivery. by Eric Laurent · 16 years ago
  40. 49ce029 am 8e11c822: am 9fee0b2a: Ogg files can be tagged to be automatically looping, this setting always overrides the MediaPlayer\'s setLooping setting. by Andreas Huber · 16 years ago
  41. 1d816a9 am 9fee0b2a: Ogg files can be tagged to be automatically looping, this setting always overrides the MediaPlayer\'s setLooping setting. by Andreas Huber · 16 years ago
  42. 24beeac Merge "Add the new Stagefright ANativeWindow OMX codec API." by Jamie Gennis · 16 years ago
  43. 83750ea Add the new Stagefright ANativeWindow OMX codec API. by Jamie Gennis · 16 years ago
  44. 8ae49d8 Ogg files can be tagged to be automatically looping, this setting always overrides the MediaPlayer's setLooping setting. by Andreas Huber · 16 years ago
  45. 511ea98 am af7a7c34: am cc4a38c6: Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 16 years ago
  46. 6f6bc92 am cc4a38c6: Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 16 years ago
  47. 1a4c79e Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 16 years ago
  48. 8650e19 Properly buffer a certain amount of data on streaming sources before finishing prepare(). by Andreas Huber · 16 years ago
  49. caa68a5 Not all audio source has the drift time information by James Dong · 16 years ago
  50. 35b8b48 am bc1452a3: am 7755cdd6: Remove unused/debugging code from MP4 file writer by James Dong · 16 years ago
  51. 52c006e am 7755cdd6: Remove unused/debugging code from MP4 file writer by James Dong · 16 years ago
  52. b4d5320 Remove unused/debugging code from MP4 file writer by James Dong · 16 years ago
  53. 92be4e1 am 3c3fc97e: am 46e63b34: Merge "Better file size estimate" into gingerbread by James Dong · 16 years ago
  54. de428f1 am 46e63b34: Merge "Better file size estimate" into gingerbread by James Dong · 16 years ago
  55. 1f90c4b Better file size estimate by James Dong · 16 years ago
  56. fbc6db2 am bb64e554: am 7ed7668b: Merge "Calculate audio media drift time from AudioSource" into gingerbread by James Dong · 16 years ago
  57. ea0fe65 am 7ed7668b: Merge "Calculate audio media drift time from AudioSource" into gingerbread by James Dong · 16 years ago
  58. e3e76c4 MTP: Implement support for getting/setting device properties by Mike Lockwood · 16 years ago
  59. d707fcb Calculate audio media drift time from AudioSource by James Dong · 16 years ago
  60. 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 · 16 years ago
  61. 955c5d0 am 3fd01c4d: am d3c1bae4: Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 16 years ago
  62. 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 · 16 years ago
  63. 4cd45f8 am d3c1bae4: Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 16 years ago
  64. 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 · 16 years ago
  65. ab8920a Merge "Support for auxiliary video parameters." by Nipun Kwatra · 16 years ago
  66. e91b462 Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 16 years ago
  67. 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 · 16 years ago
  68. 9f2d258 Revert "Merge "Add the new Stagefright ANativeWindow OMX codec API."" by Jamie Gennis · 16 years ago
  69. 6e20bdf Make sure that if initialization fails, AudioSource still behaves well. by James Dong · 16 years ago
  70. 9dd4931 Merge "Add the new Stagefright ANativeWindow OMX codec API." by Jamie Gennis · 16 years ago
  71. cfe88a2 Support for auxiliary video parameters. by Nipun Kwatra · 16 years ago
  72. 56f2c6e am 47f2cf62: am 412fc7cd: Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 16 years ago
  73. 002b34c am 412fc7cd: Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 16 years ago
  74. bcbe5af Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 16 years ago
  75. 8852dbb am 021a822e: am de2b1615: Merge "Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer." into gingerbread by Andreas Huber · 16 years ago
  76. 559dc60 Add the new Stagefright ANativeWindow OMX codec API. by Jamie Gennis · 16 years ago
  77. d8c48ad am de2b1615: Merge "Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer." into gingerbread by Andreas Huber · 16 years ago
  78. 82f7321 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. by Andreas Huber · 16 years ago
  79. 389636c Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 16 years ago
  80. 1040127 am 55e79374: am f560ceab: Merge "Audio Effects: fix problems in volume control." into gingerbread by Eric Laurent · 16 years ago
  81. 2e0448f am f560ceab: Merge "Audio Effects: fix problems in volume control." into gingerbread by Eric Laurent · 16 years ago
  82. 8f45bd7 Audio Effects: fix problems in volume control. by Eric Laurent · 16 years ago
  83. 3ef9f98 am 6b52911c: am 48ac68e1: Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 16 years ago
  84. dc344e5 am 48ac68e1: Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 16 years ago
  85. 0612475 Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 16 years ago
  86. 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 · 16 years ago
  87. 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 · 16 years ago
  88. 23e6126 Merge "MTP: Send an Intent after an MTP session that resulted in media database modifications" by Mike Lockwood · 16 years ago
  89. 4dba3e9 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 16 years ago
  90. 6b3a9d1 MTP: Send an Intent after an MTP session that resulted in media database modifications by Mike Lockwood · 16 years ago
  91. 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 · 16 years ago
  92. ad452ff Added VideoSourceDownSampler by Nipun Kwatra · 16 years ago
  93. 5d6aca5 Added setAuxiliaryOutputFile to MediaRecorder and JNI by Nipun Kwatra · 16 years ago
  94. 9c075bc Adding support for parallel recording sessions. by Nipun Kwatra · 16 years ago
  95. 6865ddb am 1577e629: am 12006013: fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 16 years ago
  96. d0e5b40 Merge "Make sure we only reallocate buffers on a genuine port definition change." by Andreas Huber · 16 years ago
  97. eebcf36 am 12006013: fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 16 years ago
  98. 5edae61 fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 16 years ago
  99. 5e9d8a2 Make sure we only reallocate buffers on a genuine port definition change. by Andreas Huber · 16 years ago
  100. 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 · 16 years ago