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gerrit-public.fairphone.software / platform / frameworks / base / 8f5a1211dc14acb2ef1879b9dad12e3dbaf92e9e / media
  1. 2fb43ef fix problem in AudioEffect JNI setup. by Eric Laurent · 14 years ago
  2. 7b2ed5d Merge "Depending on our preference to write 2-byte or 4-byte NALs, patch the codec specific data 'avcC' accordingly." into gingerbread by Andreas Huber · 14 years ago
  3. e763593 Depending on our preference to write 2-byte or 4-byte NALs, patch the codec specific data 'avcC' accordingly. by Andreas Huber · 14 years ago
  4. 1a5149e Fix issue 3022800. by Eric Laurent · 14 years ago
  5. 903fc22 Ignore errors from correction parameter query and config for M4v and H263 encoders by James Dong · 14 years ago
  6. 1e0e166 Use the advertised profile and level from M4V and H263 video encoders by James Dong · 14 years ago
  7. 3efbc55 Merge "Fix more audio effects auto tests" into gingerbread by Eric Laurent · 14 years ago
  8. 9c37da7 Raise the amount of memory set aside for omx buffer allocations in the test harness to accomodate the new requirements of some codecs. by Andreas Huber · 14 years ago
  9. ec1e9c7 Fix more audio effects auto tests by Eric Laurent · 14 years ago
  10. 55f8aee Make sure we drain the avc software decoder's output queue once we run out of input data. by Andreas Huber · 14 years ago
  11. 4f29455 Merge "Fix broken insert reverb auto tests." into gingerbread by Eric Laurent · 14 years ago
  12. 71fe631 Fix broken insert reverb auto tests. by Eric Laurent · 14 years ago
  13. a6dc469 Merge "Fix issue 2913071." into gingerbread by Eric Laurent · 14 years ago
  14. 87d208f Merge "This log message is codec specific." into gingerbread by Andreas Huber · 14 years ago
  15. 6773848 Merge "Remove stagefright foundation's incompatible logging interface and update callsites." into gingerbread by Andreas Huber · 14 years ago
  16. 6e3fa44 Remove stagefright foundation's incompatible logging interface and update callsites. by Andreas Huber · 14 years ago
  17. 672c0dc Fix issue 2913071. by Eric Laurent · 14 years ago
  18. af0a188 This log message is codec specific. by Andreas Huber · 14 years ago
  19. 8d1513e Merge "Made audio effect control panel intents public." into gingerbread by Eric Laurent · 14 years ago
  20. 92cf2d6 Made audio effect control panel intents public. by Eric Laurent · 14 years ago
  21. 40da64f Another attempt for fixing AAC+/eAAC+ related issue by James Dong · 14 years ago
  22. 3c473ea Add a check to track a problem the monkey script has been triggering. by Marco Nelissen · 14 years ago
  23. f98197a 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 · 14 years ago
  24. 524e6f6 Register the new OMX components. by Andreas Huber · 14 years ago
  25. a7516e9 Merge "Make sure the .wav extractor does not read data outside the bounds of the 'data' box." into gingerbread by Andreas Huber · 14 years ago
  26. 4f5bb1e Make sure the .wav extractor does not read data outside the bounds of the 'data' box. by Andreas Huber · 14 years ago
  27. a7f5e47 Merge "Fixed a bug in the query to the supported profiles and levels" into gingerbread by James Dong · 14 years ago
  28. dfb8991 Fixed a bug in the query to the supported profiles and levels by James Dong · 14 years ago
  29. 45922df 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 · 14 years ago
  30. 8946ab2 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 · 14 years ago
  31. 37de5da Merge "Fix problem in lvm effect bundle wrapper" into gingerbread by Eric Laurent · 14 years ago
  32. f0f95b8 Fix problem in lvm effect bundle wrapper by Eric Laurent · 14 years ago
  33. 24a2c2b Merge "Upgrade to the latest .webm project code." into gingerbread by Andreas Huber · 14 years ago
  34. e5f8539 Upgrade to the latest .webm project code. by Andreas Huber · 14 years ago
  35. 82a39f4 Merge "Add some explicit error log messages" into gingerbread by James Dong · 14 years ago
  36. 9f882c0 Merge "Fix audio input sample timestamp when audio driver loses audio samples" into gingerbread by James Dong · 14 years ago
  37. 9909b94 Merge "Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting." into gingerbread by Andreas Huber · 14 years ago
  38. f3d2bdf Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting. by Andreas Huber · 14 years ago
  39. b5024da Add some explicit error log messages by James Dong · 14 years ago
  40. 7589ebf Fix audio input sample timestamp when audio driver loses audio samples by James Dong · 14 years ago
  41. d09af7d Added intents for audio effects control application by Eric Laurent · 14 years ago
  42. 27a2fdf Fix volume problems with insert revert by Eric Laurent · 14 years ago
  43. 4709c7f Merge "Fixed a copy and paste error" into gingerbread by James Dong · 14 years ago
  44. 2814ad2 Merge "LVM release 1.09 delivery" into gingerbread by Eric Laurent · 14 years ago
  45. 8d3b910 Fixed a copy and paste error by James Dong · 14 years ago
  46. 4b3d32b TimedEventQueue now explicitly sets its scheduling policy to foreground as it should. by Andreas Huber · 14 years ago
  47. 305443c LVM release 1.09 delivery by Eric Laurent · 14 years ago
  48. f3de053 Merge "Instead of asserting return a runtime error if the maximum sample size cannot be determined." into gingerbread by Andreas Huber · 14 years ago
  49. 5c43a7a 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 · 14 years ago
  50. d7f2225 Instead of asserting return a runtime error if the maximum sample size cannot be determined. by Andreas Huber · 14 years ago
  51. 3e0f2be Instead of asserting, publish no tracks if an MP3Extractor is used on non-mp3 content. by Andreas Huber · 14 years ago
  52. a4fb816 When 32-bit offset is used, by James Dong · 14 years ago
  53. d353c84 Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 14 years ago
  54. d015ccf HW audio encoder expects timestamp via kKeyTime from each input buffer by James Dong · 14 years ago
  55. 95d5de0 Modify type of some environmental reverb parameters by Eric Laurent · 14 years ago
  56. 7e42793 Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 14 years ago
  57. 9077f8e Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 14 years ago
  58. 5fa6df6 LVM release 1.08 delivery. by Eric Laurent · 14 years ago
  59. 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
  60. cc4a38c Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 14 years ago
  61. 87ab9cd Properly buffer a certain amount of data on streaming sources before finishing prepare(). by Andreas Huber · 14 years ago
  62. 3caa714 Not all audio source has the drift time information by James Dong · 14 years ago
  63. 7755cdd Remove unused/debugging code from MP4 file writer by James Dong · 14 years ago
  64. cb7e65c Better file size estimate by James Dong · 14 years ago
  65. 4c23815 Calculate audio media drift time from AudioSource by James Dong · 14 years ago
  66. 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
  67. d3c1bae Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 14 years ago
  68. 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
  69. a87544b Make sure that if initialization fails, AudioSource still behaves well. by James Dong · 14 years ago
  70. 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
  71. 412fc7c Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 14 years ago
  72. 8d7d413 Now that AmrInputStream no longer relies on opencore, make sure it's registered in non-opencore builds. by Andreas Huber · 14 years ago
  73. 4dcc6a1 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. by Andreas Huber · 14 years ago
  74. 27b9c8e Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 14 years ago
  75. a92ebfa Audio Effects: fix problems in volume control. by Eric Laurent · 14 years ago
  76. 48ac68e Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 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. 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
  79. 1200601 fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 14 years ago
  80. 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
  81. 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
  82. 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
  83. 681c5ff 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
  84. 30cfa20 Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore. by Andreas Huber · 14 years ago
  85. 858bb4f Merge "LVM release 1.07 delivery." into gingerbread by Eric Laurent · 14 years ago
  86. f6639c4 Finetune some rtsp timeout constants. by Andreas Huber · 14 years ago
  87. df992ac Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 14 years ago
  88. c4e0b70 ALoopers can now be named (useful to distinguish threads). by Andreas Huber · 14 years ago
  89. 90862e2 Workaround for a QCOM issue where the output buffer size advertised by the AVC encoder by James Dong · 14 years ago
  90. b86365a Merge "Suppress the video recording start signal - bug 2950297" into gingerbread by James Dong · 14 years ago
  91. eeb97d9 Better support for rtsp (normal play-)time display. Better seek support, timeout if no packets arrive for too long. by Andreas Huber · 14 years ago
  92. adecf1c LVM release 1.07 delivery. by Eric Laurent · 14 years ago
  93. d6a4004 We accidentally always aborted after 10 secs, even if the connection was fine. by Andreas Huber · 14 years ago
  94. d7f1c3d Suppress the video recording start signal - bug 2950297 by James Dong · 14 years ago
  95. 17a765a Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 14 years ago
  96. 0416da7 Support for RTP packets arriving interleaved with RTSP responses. by Andreas Huber · 14 years ago
  97. 71450f8 Changed type of reverb presets from int to short by Eric Laurent · 14 years ago
  98. dfded35 Merge "Added automated tests for reverb audio effect." into gingerbread by Eric Laurent · 14 years ago
  99. 318a759 Merge "Make sure that timestamp does not go backward in MP4 file writer" into gingerbread by James Dong · 14 years ago
  100. 391e2d0 Added automated tests for reverb audio effect. by Eric Laurent · 14 years ago