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review.evervolv.com / android_frameworks_av / a01b7943b59eb869911f6a43b906f44ddefdebed
  1. 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
  2. f391593 am ff4c57ad: Merge "TimedEventQueue now explicitly sets its scheduling policy to foreground as it should." into gingerbread by Andreas Huber · 16 years ago
  3. f50e91e Merge "Only enable CAMERA_MSG_PREVIEW_FRAME when necessary." by Wu-cheng Li · 16 years ago
  4. 0667de7 Only enable CAMERA_MSG_PREVIEW_FRAME when necessary. by Wu-cheng Li · 16 years ago
  5. acb5621 TimedEventQueue now explicitly sets its scheduling policy to foreground as it should. by Andreas Huber · 16 years ago
  6. 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
  7. 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
  8. 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
  9. 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
  10. 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
  11. 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
  12. 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
  13. 49110ce Instead of asserting return a runtime error if the maximum sample size cannot be determined. by Andreas Huber · 16 years ago
  14. 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
  15. 772bcc2 Instead of asserting, publish no tracks if an MP3Extractor is used on non-mp3 content. by Andreas Huber · 16 years ago
  16. 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
  17. d2518e0 When 32-bit offset is used, by James Dong · 16 years ago
  18. 368b56e am d353c840: Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 16 years ago
  19. fbf7162 Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 16 years ago
  20. 3c3763d HW audio encoder expects timestamp via kKeyTime from each input buffer by James Dong · 16 years ago
  21. 87e6079 am e126119c: am 95d86480: Merge "Modify type of some environmental reverb parameters" into gingerbread by Eric Laurent · 16 years ago
  22. e92e213 am 95d86480: Merge "Modify type of some environmental reverb parameters" into gingerbread by Eric Laurent · 16 years ago
  23. 54c38fd Modify type of some environmental reverb parameters by Eric Laurent · 16 years ago
  24. 5de4999 Don't assert on unexpected surface flinger dequeue/enqueueBuffer errors, log a warning and ignore them instead. by Andreas Huber · 16 years ago
  25. 5d7d90a Merge changes I71f5b0fc,I92c7accb by Nipun Kwatra · 16 years ago
  26. 155e833 Moving decision to use still camera to CameraSourceTimeLapse by Nipun Kwatra · 16 years ago
  27. 4d3f97e am ef9e508c: am 7e427934: Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 16 years ago
  28. 734c3c9 am d6fd133d: am 9077f8ec: Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 16 years ago
  29. aa8d119 am 7e427934: Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 16 years ago
  30. f9c0ae8 Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 16 years ago
  31. 77682db am 9077f8ec: Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 16 years ago
  32. ddba3f0 Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 16 years ago
  33. 2d3bf53 LVM release 1.08 delivery. by Eric Laurent · 16 years ago
  34. 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
  35. 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
  36. 24beeac Merge "Add the new Stagefright ANativeWindow OMX codec API." by Jamie Gennis · 16 years ago
  37. 83750ea Add the new Stagefright ANativeWindow OMX codec API. by Jamie Gennis · 16 years ago
  38. 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
  39. 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
  40. 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
  41. 1a4c79e Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 16 years ago
  42. 8650e19 Properly buffer a certain amount of data on streaming sources before finishing prepare(). by Andreas Huber · 16 years ago
  43. caa68a5 Not all audio source has the drift time information by James Dong · 16 years ago
  44. 35b8b48 am bc1452a3: am 7755cdd6: Remove unused/debugging code from MP4 file writer by James Dong · 16 years ago
  45. 52c006e am 7755cdd6: Remove unused/debugging code from MP4 file writer by James Dong · 16 years ago
  46. b4d5320 Remove unused/debugging code from MP4 file writer by James Dong · 16 years ago
  47. 92be4e1 am 3c3fc97e: am 46e63b34: Merge "Better file size estimate" into gingerbread by James Dong · 16 years ago
  48. de428f1 am 46e63b34: Merge "Better file size estimate" into gingerbread by James Dong · 16 years ago
  49. 1f90c4b Better file size estimate by James Dong · 16 years ago
  50. fbc6db2 am bb64e554: am 7ed7668b: Merge "Calculate audio media drift time from AudioSource" into gingerbread by James Dong · 16 years ago
  51. ea0fe65 am 7ed7668b: Merge "Calculate audio media drift time from AudioSource" into gingerbread by James Dong · 16 years ago
  52. bd05775 Merge "Calculate audio media drift time from AudioSource" into gingerbread by James Dong · 16 years ago
  53. e5fe260 am 937056c1: am 32ec1ad1: Merge "Fix problem in AudioEffect::command() status." into gingerbread by Eric Laurent · 16 years ago
  54. a526824 am 32ec1ad1: Merge "Fix problem in AudioEffect::command() status." into gingerbread by Eric Laurent · 16 years ago
  55. 34c8d61 Merge "Fix problem in AudioEffect::command() status." into gingerbread by Eric Laurent · 16 years ago
  56. e3e76c4 MTP: Implement support for getting/setting device properties by Mike Lockwood · 16 years ago
  57. aeae3de Fix problem in AudioEffect::command() status. by Eric Laurent · 16 years ago
  58. d707fcb Calculate audio media drift time from AudioSource by James Dong · 16 years ago
  59. 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
  60. 955c5d0 am 3fd01c4d: am d3c1bae4: Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 16 years ago
  61. 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
  62. 4cd45f8 am d3c1bae4: Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 16 years ago
  63. 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
  64. ab8920a Merge "Support for auxiliary video parameters." by Nipun Kwatra · 16 years ago
  65. e91b462 Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 16 years ago
  66. 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
  67. 9f2d258 Revert "Merge "Add the new Stagefright ANativeWindow OMX codec API."" by Jamie Gennis · 16 years ago
  68. 6e20bdf Make sure that if initialization fails, AudioSource still behaves well. by James Dong · 16 years ago
  69. 9dd4931 Merge "Add the new Stagefright ANativeWindow OMX codec API." by Jamie Gennis · 16 years ago
  70. cfe88a2 Support for auxiliary video parameters. by Nipun Kwatra · 16 years ago
  71. 56f2c6e am 47f2cf62: am 412fc7cd: Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 16 years ago
  72. 002b34c am 412fc7cd: Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 16 years ago
  73. bcbe5af Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 16 years ago
  74. 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
  75. 559dc60 Add the new Stagefright ANativeWindow OMX codec API. by Jamie Gennis · 16 years ago
  76. d8c48ad am de2b1615: Merge "Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer." into gingerbread by Andreas Huber · 16 years ago
  77. 82f7321 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. by Andreas Huber · 16 years ago
  78. 389636c Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 16 years ago
  79. 84284e7 Merge "Remove setPreviewDisplay in CameraService::Client::~Client()." by Wu-cheng Li · 16 years ago
  80. 3abe8d2 Remove setPreviewDisplay in CameraService::Client::~Client(). by Wu-cheng Li · 16 years ago
  81. 1040127 am 55e79374: am f560ceab: Merge "Audio Effects: fix problems in volume control." into gingerbread by Eric Laurent · 16 years ago
  82. 2e0448f am f560ceab: Merge "Audio Effects: fix problems in volume control." into gingerbread by Eric Laurent · 16 years ago
  83. 8f45bd7 Audio Effects: fix problems in volume control. by Eric Laurent · 16 years ago
  84. 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
  85. 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
  86. 0612475 Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 16 years ago
  87. 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
  88. 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
  89. 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 · 16 years ago
  90. 23e6126 Merge "MTP: Send an Intent after an MTP session that resulted in media database modifications" by Mike Lockwood · 16 years ago
  91. 4dba3e9 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 16 years ago
  92. 85e28b4 am aa40b834: am 6aacad66: Merge "Add some encoding parameters for the "record" utility" into gingerbread by James Dong · 16 years ago
  93. eaae2c3 am 6aacad66: Merge "Add some encoding parameters for the "record" utility" into gingerbread by James Dong · 16 years ago
  94. 6b3a9d1 MTP: Send an Intent after an MTP session that resulted in media database modifications by Mike Lockwood · 16 years ago
  95. f74c8f9 Add some encoding parameters for the "record" utility by James Dong · 16 years ago
  96. 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
  97. ad452ff Added VideoSourceDownSampler by Nipun Kwatra · 16 years ago
  98. 5d6aca5 Added setAuxiliaryOutputFile to MediaRecorder and JNI by Nipun Kwatra · 16 years ago
  99. 9c075bc Adding support for parallel recording sessions. by Nipun Kwatra · 16 years ago
  100. 6865ddb am 1577e629: am 12006013: fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 16 years ago