1. e3aa6aa am 306cd58d: am d353c840: Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 14 years ago
  2. 306cd58 am d353c840: Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 14 years ago
  3. d353c84 Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 14 years ago
  4. d015ccf HW audio encoder expects timestamp via kKeyTime from each input buffer by James Dong · 14 years ago
  5. 15ff01c am e126119c: am 95d86480: Merge "Modify type of some environmental reverb parameters" into gingerbread by Eric Laurent · 14 years ago
  6. e126119 am 95d86480: Merge "Modify type of some environmental reverb parameters" into gingerbread by Eric Laurent · 14 years ago
  7. 95d5de0 Modify type of some environmental reverb parameters by Eric Laurent · 14 years ago
  8. e521169 MTP: Allow transfering arbitrary file types. by Mike Lockwood · 14 years ago
  9. 3b2a62e Rename the media database's "objects" table to "files" by Mike Lockwood · 14 years ago
  10. 94e6b66 Don't assert on unexpected surface flinger dequeue/enqueueBuffer errors, log a warning and ignore them instead. by Andreas Huber · 14 years ago
  11. f5e1faf Merge changes I71f5b0fc,I92c7accb by Nipun Kwatra · 14 years ago
  12. 4a857e6 Moving decision to use still camera to CameraSourceTimeLapse by Nipun Kwatra · 14 years ago
  13. bcb284c am ef9e508c: am 7e427934: Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 14 years ago
  14. c4e7be5 am d6fd133d: am 9077f8ec: Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 14 years ago
  15. ef9e508 am 7e427934: Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 14 years ago
  16. 7e42793 Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 14 years ago
  17. d6fd133 am 9077f8ec: Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 14 years ago
  18. 9077f8e Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 14 years ago
  19. 5fa6df6 LVM release 1.08 delivery. by Eric Laurent · 14 years ago
  20. 1ab9d12 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 · 14 years ago
  21. 8e11c82 am 9fee0b2a: Ogg files can be tagged to be automatically looping, this setting always overrides the MediaPlayer\'s setLooping setting. by Andreas Huber · 14 years ago
  22. a093659 Merge "Add the new Stagefright ANativeWindow OMX codec API." by Jamie Gennis · 14 years ago
  23. 33a7814 Add the new Stagefright ANativeWindow OMX codec API. by Jamie Gennis · 14 years ago
  24. 9fee0b2 Ogg files can be tagged to be automatically looping, this setting always overrides the MediaPlayer's setLooping setting. by Andreas Huber · 14 years ago
  25. 00c88ea am af7a7c34: am cc4a38c6: Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 14 years ago
  26. af7a7c3 am cc4a38c6: Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 14 years ago
  27. cc4a38c Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 14 years ago
  28. 87ab9cd Properly buffer a certain amount of data on streaming sources before finishing prepare(). by Andreas Huber · 14 years ago
  29. 3caa714 Not all audio source has the drift time information by James Dong · 14 years ago
  30. 9b3569b am bc1452a3: am 7755cdd6: Remove unused/debugging code from MP4 file writer by James Dong · 14 years ago
  31. bc1452a am 7755cdd6: Remove unused/debugging code from MP4 file writer by James Dong · 14 years ago
  32. 7755cdd Remove unused/debugging code from MP4 file writer by James Dong · 14 years ago
  33. 0e60f53 am 3c3fc97e: am 46e63b34: Merge "Better file size estimate" into gingerbread by James Dong · 14 years ago
  34. 3c3fc97 am 46e63b34: Merge "Better file size estimate" into gingerbread by James Dong · 14 years ago
  35. cb7e65c Better file size estimate by James Dong · 14 years ago
  36. 9f20d33 am bb64e554: am 7ed7668b: Merge "Calculate audio media drift time from AudioSource" into gingerbread by James Dong · 14 years ago
  37. bb64e55 am 7ed7668b: Merge "Calculate audio media drift time from AudioSource" into gingerbread by James Dong · 14 years ago
  38. db16e5a Merge "MTP: Implement support for getting/setting device properties" by Mike Lockwood · 14 years ago
  39. 59e3f0d MTP: Implement support for getting/setting device properties by Mike Lockwood · 14 years ago
  40. ab15bce pass auxiliary video parameters. by Nipun Kwatra · 14 years ago
  41. 602bebd MTP: try to fix sim build by Mike Lockwood · 14 years ago
  42. 4c23815 Calculate audio media drift time from AudioSource by James Dong · 14 years ago
  43. 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
  44. 5aa0adc am 3fd01c4d: am d3c1bae4: Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 14 years ago
  45. fd0eed0 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
  46. 3fd01c4 am d3c1bae4: Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 14 years ago
  47. 5b98831 Merge "resolved conflicts for merge of 71c908c4 to master" by Andreas Huber · 14 years ago
  48. ee9c3db resolved conflicts for merge of 71c908c4 to master by Andreas Huber · 14 years ago
  49. a2511da 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
  50. 8d89cda Merge "Support for auxiliary video parameters." by Nipun Kwatra · 14 years ago
  51. d3c1bae Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 14 years ago
  52. 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
  53. 2198d75 Revert "Merge "Add the new Stagefright ANativeWindow OMX codec API."" by Jamie Gennis · 14 years ago
  54. a87544b Make sure that if initialization fails, AudioSource still behaves well. by James Dong · 14 years ago
  55. 8a643b4 Merge "Add the new Stagefright ANativeWindow OMX codec API." by Jamie Gennis · 14 years ago
  56. 71c908c am 6c33904a: Merge "Now that AmrInputStream no longer relies on opencore, make sure it\'s registered in non-opencore builds." into gingerbread by Andreas Huber · 14 years ago
  57. 6c33904 Merge "Now that AmrInputStream no longer relies on opencore, make sure it's registered in non-opencore builds." into gingerbread by Andreas Huber · 14 years ago
  58. 239f2e5 Support for auxiliary video parameters. by Nipun Kwatra · 14 years ago
  59. 52d14be am 47f2cf62: am 412fc7cd: Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 14 years ago
  60. 47f2cf6 am 412fc7cd: Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 14 years ago
  61. 412fc7c Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 14 years ago
  62. 8d7d413 Now that AmrInputStream no longer relies on opencore, make sure it's registered in non-opencore builds. by Andreas Huber · 14 years ago
  63. 564a9f2 am 021a822e: am de2b1615: Merge "Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer." into gingerbread by Andreas Huber · 14 years ago
  64. dab357b Add the new Stagefright ANativeWindow OMX codec API. by Jamie Gennis · 14 years ago
  65. 021a822 am de2b1615: Merge "Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer." into gingerbread by Andreas Huber · 14 years ago
  66. 4dcc6a1 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. by Andreas Huber · 14 years ago
  67. 27b9c8e Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 14 years ago
  68. 288db3b am 55e79374: am f560ceab: Merge "Audio Effects: fix problems in volume control." into gingerbread by Eric Laurent · 14 years ago
  69. 55e7937 am f560ceab: Merge "Audio Effects: fix problems in volume control." into gingerbread by Eric Laurent · 14 years ago
  70. a92ebfa Audio Effects: fix problems in volume control. by Eric Laurent · 14 years ago
  71. 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
  72. 6b52911 am 48ac68e1: Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 14 years ago
  73. 48ac68e Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 14 years ago
  74. 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
  75. e1a3cdd 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
  76. f06a449 Merge "MTP: Send an Intent after an MTP session that resulted in media database modifications" by Mike Lockwood · 14 years ago
  77. e536f80 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 14 years ago
  78. d20e802 Merge "Add settings option for running MTP server in PTP mode." by Mike Lockwood · 14 years ago
  79. 2837eef MTP: Send an Intent after an MTP session that resulted in media database modifications by Mike Lockwood · 14 years ago
  80. eabe8bf Add settings option for running MTP server in PTP mode. by Mike Lockwood · 14 years ago
  81. 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
  82. 06a1d61 Added VideoSourceDownSampler by Nipun Kwatra · 14 years ago
  83. b33a5ae Added setAuxiliaryOutputFile to MediaRecorder and JNI by Nipun Kwatra · 14 years ago
  84. d7e7a3f Adding support for parallel recording sessions. by Nipun Kwatra · 14 years ago
  85. 2deeff1 am 1577e629: am 12006013: fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 14 years ago
  86. c855deba Merge "Make sure we only reallocate buffers on a genuine port definition change." by Andreas Huber · 14 years ago
  87. 1577e62 am 12006013: fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 14 years ago
  88. 1200601 fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 14 years ago
  89. 29c03c6 Make sure we only reallocate buffers on a genuine port definition change. by Andreas Huber · 14 years ago
  90. 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
  91. 03e83d4 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
  92. 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
  93. 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
  94. 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
  95. 987556b 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
  96. 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
  97. 2527da0 am 9aa05ec2: am 681c5ff2: Merge "Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore." into gingerbread by Andreas Huber · 14 years ago
  98. 9d01dcb am 5762dc19: am 858bb4f6: Merge "LVM release 1.07 delivery." into gingerbread by Eric Laurent · 14 years ago
  99. 3678668 am 7ed9104c: am f6639c46: Finetune some rtsp timeout constants. by Andreas Huber · 14 years ago
  100. 631025e am 6df6d606: am df992ac9: Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 14 years ago