Gitiles
Code Review Sign In
review.blissroms.org / platform_frameworks_av / a3121424e0680647e8ab7beb4229db6d8d4f337f
  1. a312142 Merge "This log message is codec specific." into gingerbread by Andreas Huber · 15 years ago
  2. 43d4f74 Merge "Remove stagefright foundation's incompatible logging interface and update callsites." into gingerbread by Andreas Huber · 15 years ago
  3. 6e4c5c4 Remove stagefright foundation's incompatible logging interface and update callsites. by Andreas Huber · 15 years ago
  4. 955194d This log message is codec specific. by Andreas Huber · 15 years ago
  5. e936413 Merge "Allow record to set input color format as a command line option" into gingerbread by James Dong · 15 years ago
  6. 425587d Merge "Another attempt for fixing AAC+/eAAC+ related issue" into gingerbread by James Dong · 15 years ago
  7. ac4205c Rename FOCUS_MODE_CONTINUOUS to FOCUS_MODE_CONTINUOUS_VIDEO. by Wu-cheng Li · 15 years ago
  8. 1826945 Another attempt for fixing AAC+/eAAC+ related issue by James Dong · 15 years ago
  9. a733679 Allow record to set input color format as a command line option by James Dong · 15 years ago
  10. 1c1503c Add a check to track a problem the monkey script has been triggering. by Marco Nelissen · 15 years ago
  11. 00998fb Make sure the message dispatcher stays around until after OMX_FreeHandle is finished in case it posts some more messages during shutdown. Clear the source as soon as possible in OMXCodec's destructor. by Andreas Huber · 15 years ago
  12. 095916d Register the new OMX components. by Andreas Huber · 15 years ago
  13. 876742d Merge "Make sure the .wav extractor does not read data outside the bounds of the 'data' box." into gingerbread by Andreas Huber · 15 years ago
  14. 102dfe0 Merge "Make sure stagefright -o terminates even if we're using a raw audio source (such as .wav pcm)" into gingerbread by Andreas Huber · 15 years ago
  15. c225da9 Make sure stagefright -o terminates even if we're using a raw audio source (such as .wav pcm) by Andreas Huber · 15 years ago
  16. 104fcb8 Make sure the .wav extractor does not read data outside the bounds of the 'data' box. by Andreas Huber · 15 years ago
  17. 0270f47 Merge "Fixed a bug in the query to the supported profiles and levels" into gingerbread by James Dong · 15 years ago
  18. f01691f Fixed a bug in the query to the supported profiles and levels by James Dong · 15 years ago
  19. 72b2749 Sometimes the avc software decoder will signal that a frame is ready but then unexpectedly fail to return the frame... stop asserting on that and return an error instead. by Andreas Huber · 15 years ago
  20. aae3516 A ThreadedSource wraps around an existing MediaSource and reads output buffers on a separate thread. It's now used for the vpx decoder to decode frames ahead of time to improve playback performance. by Andreas Huber · 15 years ago
  21. 70fb57d Merge "Fix problem in lvm effect bundle wrapper" into gingerbread by Eric Laurent · 15 years ago
  22. 29cc743 Fix problem in lvm effect bundle wrapper by Eric Laurent · 15 years ago
  23. eae6193 Merge "Upgrade to the latest .webm project code." into gingerbread by Andreas Huber · 15 years ago
  24. acf67ea Upgrade to the latest .webm project code. by Andreas Huber · 15 years ago
  25. d790910 Merge "Add some explicit error log messages" into gingerbread by James Dong · 15 years ago
  26. e78d3bb Merge "Fix audio input sample timestamp when audio driver loses audio samples" into gingerbread by James Dong · 15 years ago
  27. 3b93208 Merge "Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting." into gingerbread by Andreas Huber · 15 years ago
  28. 6f85dba Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting. by Andreas Huber · 15 years ago
  29. a1abc1a Add some explicit error log messages by James Dong · 15 years ago
  30. 67e9269 Fix audio input sample timestamp when audio driver loses audio samples by James Dong · 15 years ago
  31. e0aed6d Fix volume problems with insert revert by Eric Laurent · 15 years ago
  32. a175413 Merge "LVM release 1.09 delivery" into gingerbread by Eric Laurent · 15 years ago
  33. acb5621 TimedEventQueue now explicitly sets its scheduling policy to foreground as it should. by Andreas Huber · 15 years ago
  34. 5185b01 LVM release 1.09 delivery by Eric Laurent · 15 years ago
  35. 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
  36. 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
  37. 49110ce Instead of asserting return a runtime error if the maximum sample size cannot be determined. by Andreas Huber · 15 years ago
  38. 772bcc2 Instead of asserting, publish no tracks if an MP3Extractor is used on non-mp3 content. by Andreas Huber · 15 years ago
  39. d2518e0 When 32-bit offset is used, by James Dong · 15 years ago
  40. fbf7162 Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 15 years ago
  41. 3c3763d HW audio encoder expects timestamp via kKeyTime from each input buffer by James Dong · 15 years ago
  42. 54c38fd Modify type of some environmental reverb parameters by Eric Laurent · 15 years ago
  43. f9c0ae8 Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 15 years ago
  44. ddba3f0 Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 15 years ago
  45. 2d3bf53 LVM release 1.08 delivery. by Eric Laurent · 15 years ago
  46. 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
  47. 1a4c79e Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 15 years ago
  48. 8650e19 Properly buffer a certain amount of data on streaming sources before finishing prepare(). by Andreas Huber · 15 years ago
  49. caa68a5 Not all audio source has the drift time information by James Dong · 15 years ago
  50. b4d5320 Remove unused/debugging code from MP4 file writer by James Dong · 15 years ago
  51. 1f90c4b Better file size estimate by James Dong · 15 years ago
  52. bd05775 Merge "Calculate audio media drift time from AudioSource" into gingerbread by James Dong · 15 years ago
  53. 34c8d61 Merge "Fix problem in AudioEffect::command() status." into gingerbread by Eric Laurent · 15 years ago
  54. aeae3de Fix problem in AudioEffect::command() status. by Eric Laurent · 15 years ago
  55. d707fcb Calculate audio media drift time from AudioSource by James Dong · 15 years ago
  56. 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
  57. e91b46246 Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 15 years ago
  58. 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
  59. 6e20bdf Make sure that if initialization fails, AudioSource still behaves well. by James Dong · 15 years ago
  60. bcbe5af Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 15 years ago
  61. 82f7321 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. by Andreas Huber · 15 years ago
  62. 389636c Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 15 years ago
  63. 8f45bd7 Audio Effects: fix problems in volume control. by Eric Laurent · 15 years ago
  64. 0612475 Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 15 years ago
  65. 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
  66. 4dba3e9 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 15 years ago
  67. f74c8f9 Add some encoding parameters for the "record" utility by James Dong · 15 years ago
  68. 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
  69. 5edae61 fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 15 years ago
  70. 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
  71. 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
  72. 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
  73. 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
  74. 44eb096 Merge "Fix issue 2952766." into gingerbread by Eric Laurent · 15 years ago
  75. 541d765 Remove camera metering mode API. by Wu-cheng Li · 15 years ago
  76. 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
  77. a1ffe49 Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore. by Andreas Huber · 15 years ago
  78. 3849699 Merge "LVM release 1.07 delivery." into gingerbread by Eric Laurent · 15 years ago
  79. e56121b Finetune some rtsp timeout constants. by Andreas Huber · 15 years ago
  80. c01ec02 Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 15 years ago
  81. c1c88e2 Fix issue 2952766. by Eric Laurent · 15 years ago
  82. a814c1f ALoopers can now be named (useful to distinguish threads). by Andreas Huber · 15 years ago
  83. 824c9ff Workaround for a QCOM issue where the output buffer size advertised by the AVC encoder by James Dong · 15 years ago
  84. 352c468 Merge "Suppress the video recording start signal - bug 2950297" into gingerbread by James Dong · 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. d918324 LVM release 1.07 delivery. by Eric Laurent · 15 years ago
  89. 14cc6fc Merge "fix a race in SF buffer management" into gingerbread by Mathias Agopian · 15 years ago
  90. cc6adf5 We accidentally always aborted after 10 secs, even if the connection was fine. by Andreas Huber · 15 years ago
  91. d7c43d3 fix a race in SF buffer management by Mathias Agopian · 15 years ago
  92. f1ae196 Suppress the video recording start signal - bug 2950297 by James Dong · 15 years ago
  93. 1b07372 Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 15 years ago
  94. 0792ce7 Support for RTP packets arriving interleaved with RTSP responses. by Andreas Huber · 15 years ago
  95. 640a72e Merge "Make sure that timestamp does not go backward in MP4 file writer" into gingerbread by James Dong · 15 years ago
  96. f302743 Merge "Fix support for per-frame unsynchronization in ID3V2.4 tags." into gingerbread by Andreas Huber · 15 years ago
  97. f0bfaa8 Merge "Added preset reverb." into gingerbread by Eric Laurent · 15 years ago
  98. ac994df Fix support for per-frame unsynchronization in ID3V2.4 tags. by Andreas Huber · 15 years ago
  99. 81ce489 Merge "Ensure that buffering updates eventually hit 100% after we download everything." into gingerbread by Andreas Huber · 15 years ago
  100. 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