1. 368b56e am d353c840: Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 15 years ago
  2. fbf7162 Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 15 years ago
  3. 3c3763d HW audio encoder expects timestamp via kKeyTime from each input buffer by James Dong · 15 years ago
  4. e92e213 am 95d86480: Merge "Modify type of some environmental reverb parameters" into gingerbread by Eric Laurent · 15 years ago
  5. 54c38fd Modify type of some environmental reverb parameters by Eric Laurent · 15 years ago
  6. aa8d119 am 7e427934: Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 15 years ago
  7. f9c0ae8 Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 15 years ago
  8. 77682db am 9077f8ec: Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 15 years ago
  9. ddba3f0 Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 15 years ago
  10. 2d3bf53 LVM release 1.08 delivery. by Eric Laurent · 15 years ago
  11. 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
  12. 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
  13. 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
  14. 1a4c79e Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 15 years ago
  15. 8650e19 Properly buffer a certain amount of data on streaming sources before finishing prepare(). by Andreas Huber · 15 years ago
  16. caa68a5 Not all audio source has the drift time information by James Dong · 15 years ago
  17. 52c006e am 7755cdd6: Remove unused/debugging code from MP4 file writer by James Dong · 15 years ago
  18. b4d5320 Remove unused/debugging code from MP4 file writer by James Dong · 15 years ago
  19. de428f1 am 46e63b34: Merge "Better file size estimate" into gingerbread by James Dong · 15 years ago
  20. 1f90c4b Better file size estimate by James Dong · 15 years ago
  21. ea0fe65 am 7ed7668b: Merge "Calculate audio media drift time from AudioSource" into gingerbread by James Dong · 15 years ago
  22. bd05775 Merge "Calculate audio media drift time from AudioSource" into gingerbread by James Dong · 15 years ago
  23. a526824 am 32ec1ad1: Merge "Fix problem in AudioEffect::command() status." into gingerbread by Eric Laurent · 15 years ago
  24. 34c8d61 Merge "Fix problem in AudioEffect::command() status." into gingerbread by Eric Laurent · 15 years ago
  25. aeae3de Fix problem in AudioEffect::command() status. by Eric Laurent · 15 years ago
  26. d707fcb Calculate audio media drift time from AudioSource by James Dong · 15 years ago
  27. 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
  28. 4cd45f8 am d3c1bae4: Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 15 years ago
  29. 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
  30. e91b46246 Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 15 years ago
  31. 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
  32. 6e20bdf Make sure that if initialization fails, AudioSource still behaves well. by James Dong · 15 years ago
  33. 002b34c am 412fc7cd: Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 15 years ago
  34. bcbe5af Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 15 years ago
  35. d8c48ad am de2b1615: Merge "Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer." into gingerbread by Andreas Huber · 15 years ago
  36. 82f7321 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. by Andreas Huber · 15 years ago
  37. 389636c Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 15 years ago
  38. 2e0448f am f560ceab: Merge "Audio Effects: fix problems in volume control." into gingerbread by Eric Laurent · 15 years ago
  39. 8f45bd7 Audio Effects: fix problems in volume control. by Eric Laurent · 15 years ago
  40. 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
  41. 0612475 Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 15 years ago
  42. 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
  43. 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
  44. 4dba3e9 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 15 years ago
  45. eaae2c3 am 6aacad66: Merge "Add some encoding parameters for the "record" utility" into gingerbread by James Dong · 15 years ago
  46. f74c8f9 Add some encoding parameters for the "record" utility by James Dong · 15 years ago
  47. 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
  48. eebcf36 am 12006013: fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 15 years ago
  49. 5edae61 fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 15 years ago
  50. 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
  51. 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
  52. 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
  53. 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
  54. 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
  55. 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
  56. da701fe am ae6bdc23: Merge "Fix issue 2952766." into gingerbread by Eric Laurent · 15 years ago
  57. 44eb096 Merge "Fix issue 2952766." into gingerbread by Eric Laurent · 15 years ago
  58. fabf86f am 7ec7b997: Remove camera metering mode API. by Wu-cheng Li · 15 years ago
  59. 541d765 Remove camera metering mode API. by Wu-cheng Li · 15 years ago
  60. 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
  61. 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
  62. a1ffe49 Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore. by Andreas Huber · 15 years ago
  63. d2ab607 am 858bb4f6: Merge "LVM release 1.07 delivery." into gingerbread by Eric Laurent · 15 years ago
  64. bf5606b am f6639c46: Finetune some rtsp timeout constants. by Andreas Huber · 15 years ago
  65. c28160f am df992ac9: Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 15 years ago
  66. 3849699 Merge "LVM release 1.07 delivery." into gingerbread by Eric Laurent · 15 years ago
  67. e56121b Finetune some rtsp timeout constants. by Andreas Huber · 15 years ago
  68. c01ec02 Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 15 years ago
  69. c1c88e2 Fix issue 2952766. by Eric Laurent · 15 years ago
  70. a814c1f ALoopers can now be named (useful to distinguish threads). by Andreas Huber · 15 years ago
  71. 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
  72. 824c9ff Workaround for a QCOM issue where the output buffer size advertised by the AVC encoder by James Dong · 15 years ago
  73. 23f0d68 am b86365ad: Merge "Suppress the video recording start signal - bug 2950297" into gingerbread by James Dong · 15 years ago
  74. 352c468 Merge "Suppress the video recording start signal - bug 2950297" into gingerbread by James Dong · 15 years ago
  75. 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
  76. 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
  77. 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
  78. 165dc4c Merge "We accidentally always aborted after 10 secs, even if the connection was fine." into gingerbread by Andreas Huber · 15 years ago
  79. 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
  80. 95fd60f am ae66946b: Merge "fix a race in SF buffer management" into gingerbread by Mathias Agopian · 15 years ago
  81. d918324 LVM release 1.07 delivery. by Eric Laurent · 15 years ago
  82. 14cc6fc Merge "fix a race in SF buffer management" into gingerbread by Mathias Agopian · 15 years ago
  83. cc6adf5 We accidentally always aborted after 10 secs, even if the connection was fine. by Andreas Huber · 15 years ago
  84. d7c43d3 fix a race in SF buffer management by Mathias Agopian · 15 years ago
  85. f1ae196 Suppress the video recording start signal - bug 2950297 by James Dong · 15 years ago
  86. f39928d am 17a765a1: Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 15 years ago
  87. 1b07372 Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 15 years ago
  88. 0792ce7 Support for RTP packets arriving interleaved with RTSP responses. by Andreas Huber · 15 years ago
  89. 64b531e am 318a759e: Merge "Make sure that timestamp does not go backward in MP4 file writer" into gingerbread by James Dong · 15 years ago
  90. 640a72e Merge "Make sure that timestamp does not go backward in MP4 file writer" into gingerbread by James Dong · 15 years ago
  91. 90e4b45 am 8ac0983e: Merge "Fix support for per-frame unsynchronization in ID3V2.4 tags." into gingerbread by Andreas Huber · 15 years ago
  92. f302743 Merge "Fix support for per-frame unsynchronization in ID3V2.4 tags." into gingerbread by Andreas Huber · 15 years ago
  93. 3bfb0a0 am c14f9ca6: Merge "Added preset reverb." into gingerbread by Eric Laurent · 15 years ago
  94. f0bfaa8 Merge "Added preset reverb." into gingerbread by Eric Laurent · 15 years ago
  95. b821911 am 23584022: Merge "Ensure that buffering updates eventually hit 100% after we download everything." into gingerbread by Andreas Huber · 15 years ago
  96. ac994df Fix support for per-frame unsynchronization in ID3V2.4 tags. by Andreas Huber · 15 years ago
  97. 81ce489 Merge "Ensure that buffering updates eventually hit 100% after we download everything." into gingerbread by Andreas Huber · 15 years ago
  98. 4243f1a am b8814dce: Merge "Allow sniffers to return a packet of opaque data that the corresponding extractor can take advantage of to not duplicate work already done sniffing. The mp3 extractor takes advantage of this now." into gingerbread by Andreas Huber · 15 years ago
  99. 3f71e8b Merge "Allow sniffers to return a packet of opaque data that the corresponding extractor can take advantage of to not duplicate work already done sniffing. The mp3 extractor takes advantage of this now." into gingerbread by Andreas Huber · 15 years ago
  100. 5a1c352 Allow sniffers to return a packet of opaque data that the corresponding extractor can take advantage of to not duplicate work already done sniffing. The mp3 extractor takes advantage of this now. by Andreas Huber · 15 years ago