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gerrit.omnirom.org / android_frameworks_av / 8211da9eabe28037ebcc23bbfbbe20cb21cb0cdc
  1. 8211da9 Merge "Vorbis files may have more samples encoded that should be used, i.e. we have to trim samples at the end of the stream. This is crucial for proper looping of some audio files." into gingerbread by Andreas Huber · 16 years ago
  2. 640d660 Merge "Squashed commit of the following:" into gingerbread by Andreas Huber · 16 years ago
  3. db62222 Vorbis files may have more samples encoded that should be used, i.e. we have to trim samples at the end of the stream. This is crucial for proper looping of some audio files. by Andreas Huber · 16 years ago
  4. 02fa834 Fix media.player dumpsys to output open/mapped files correctly. Bug 2866669. by Dave Sparks · 16 years ago
  5. 0da4dab Squashed commit of the following: by Andreas Huber · 16 years ago
  6. 4a73f3d The old overlay should be destroyed if orientation changes. by Wu-cheng Li · 16 years ago
  7. e2dc4aa Merge "Fix track duration calculation if the start timestamp is non-zero" into gingerbread by James Dong · 16 years ago
  8. 8428af5 Fix track duration calculation if the start timestamp is non-zero by James Dong · 16 years ago
  9. 676570f Merge "Support other kinds of HTTP redirect in NuHTTPDataSource" into gingerbread by Andreas Huber · 16 years ago
  10. 7539a05 Support other kinds of HTTP redirect in NuHTTPDataSource by Andreas Huber · 16 years ago
  11. 0fa449c Fix error in AudioEffect command status reporting. by Eric Laurent · 16 years ago
  12. 76e2c7b Merge "Instead of just writing one track to the .mp4 file, write all of them (at most 1 video and 1 audio track). Also support httplive URIs in the stagefright commandline tool." into gingerbread by Andreas Huber · 16 years ago
  13. ea314ac Instead of just writing one track to the .mp4 file, write all of them (at most 1 video and 1 audio track). Also support httplive URIs in the stagefright commandline tool. by Andreas Huber · 16 years ago
  14. 35dd00b Merge "Fix issue 3007862" into gingerbread by Eric Laurent · 16 years ago
  15. 84e9a10 Fix issue 3007862 by Eric Laurent · 16 years ago
  16. bbc3831 Proper sync-frame detection for sources that don't already provide it. by Andreas Huber · 16 years ago
  17. 1f78bad 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 · 16 years ago
  18. 95fcef2 Depending on our preference to write 2-byte or 4-byte NALs, patch the codec specific data 'avcC' accordingly. by Andreas Huber · 16 years ago
  19. 59f63db Ignore errors from correction parameter query and config for M4v and H263 encoders by James Dong · 16 years ago
  20. 97e0fcc Use the advertised profile and level from M4V and H263 video encoders by James Dong · 16 years ago
  21. fac4895 Request permission for global audio effects. by Eric Laurent · 16 years ago
  22. 9fb467e 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 · 16 years ago
  23. 5a808f8 Make sure we drain the avc software decoder's output queue once we run out of input data. by Andreas Huber · 16 years ago
  24. 5ee6bb5 Merge "Fix issue 2913071." into gingerbread by Eric Laurent · 16 years ago
  25. a312142 Merge "This log message is codec specific." into gingerbread by Andreas Huber · 16 years ago
  26. 43d4f74 Merge "Remove stagefright foundation's incompatible logging interface and update callsites." into gingerbread by Andreas Huber · 16 years ago
  27. 6e4c5c4 Remove stagefright foundation's incompatible logging interface and update callsites. by Andreas Huber · 16 years ago
  28. 0e75f0f Fix issue 2913071. by Eric Laurent · 16 years ago
  29. 955194d This log message is codec specific. by Andreas Huber · 16 years ago
  30. e936413 Merge "Allow record to set input color format as a command line option" into gingerbread by James Dong · 16 years ago
  31. 425587d Merge "Another attempt for fixing AAC+/eAAC+ related issue" into gingerbread by James Dong · 16 years ago
  32. ac4205c Rename FOCUS_MODE_CONTINUOUS to FOCUS_MODE_CONTINUOUS_VIDEO. by Wu-cheng Li · 16 years ago
  33. 1826945 Another attempt for fixing AAC+/eAAC+ related issue by James Dong · 16 years ago
  34. a733679 Allow record to set input color format as a command line option by James Dong · 16 years ago
  35. 1c1503c Add a check to track a problem the monkey script has been triggering. by Marco Nelissen · 16 years ago
  36. 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 · 16 years ago
  37. 095916d Register the new OMX components. by Andreas Huber · 16 years ago
  38. 876742d Merge "Make sure the .wav extractor does not read data outside the bounds of the 'data' box." into gingerbread by Andreas Huber · 16 years ago
  39. 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 · 16 years ago
  40. c225da9 Make sure stagefright -o terminates even if we're using a raw audio source (such as .wav pcm) by Andreas Huber · 16 years ago
  41. 104fcb8 Make sure the .wav extractor does not read data outside the bounds of the 'data' box. by Andreas Huber · 16 years ago
  42. 0270f47 Merge "Fixed a bug in the query to the supported profiles and levels" into gingerbread by James Dong · 16 years ago
  43. f01691f Fixed a bug in the query to the supported profiles and levels by James Dong · 16 years ago
  44. 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 · 16 years ago
  45. 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 · 16 years ago
  46. 70fb57d Merge "Fix problem in lvm effect bundle wrapper" into gingerbread by Eric Laurent · 16 years ago
  47. 29cc743 Fix problem in lvm effect bundle wrapper by Eric Laurent · 16 years ago
  48. eae6193 Merge "Upgrade to the latest .webm project code." into gingerbread by Andreas Huber · 16 years ago
  49. acf67ea Upgrade to the latest .webm project code. by Andreas Huber · 16 years ago
  50. d790910 Merge "Add some explicit error log messages" into gingerbread by James Dong · 16 years ago
  51. e78d3bb Merge "Fix audio input sample timestamp when audio driver loses audio samples" into gingerbread by James Dong · 16 years ago
  52. 3b93208 Merge "Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting." into gingerbread by Andreas Huber · 16 years ago
  53. 6f85dba Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting. by Andreas Huber · 16 years ago
  54. a1abc1a Add some explicit error log messages by James Dong · 16 years ago
  55. 67e9269 Fix audio input sample timestamp when audio driver loses audio samples by James Dong · 16 years ago
  56. e0aed6d Fix volume problems with insert revert by Eric Laurent · 16 years ago
  57. a175413 Merge "LVM release 1.09 delivery" into gingerbread by Eric Laurent · 16 years ago
  58. acb5621 TimedEventQueue now explicitly sets its scheduling policy to foreground as it should. by Andreas Huber · 16 years ago
  59. 5185b01 LVM release 1.09 delivery by Eric Laurent · 16 years ago
  60. 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
  61. 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
  62. 49110ce Instead of asserting return a runtime error if the maximum sample size cannot be determined. by Andreas Huber · 16 years ago
  63. 772bcc2 Instead of asserting, publish no tracks if an MP3Extractor is used on non-mp3 content. by Andreas Huber · 16 years ago
  64. d2518e0 When 32-bit offset is used, by James Dong · 16 years ago
  65. fbf7162 Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 16 years ago
  66. 3c3763d HW audio encoder expects timestamp via kKeyTime from each input buffer by James Dong · 16 years ago
  67. 54c38fd Modify type of some environmental reverb parameters by Eric Laurent · 16 years ago
  68. f9c0ae8 Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 16 years ago
  69. ddba3f0 Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 16 years ago
  70. 2d3bf53 LVM release 1.08 delivery. by Eric Laurent · 16 years ago
  71. 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
  72. 1a4c79e Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 16 years ago
  73. 8650e19 Properly buffer a certain amount of data on streaming sources before finishing prepare(). by Andreas Huber · 16 years ago
  74. caa68a5 Not all audio source has the drift time information by James Dong · 16 years ago
  75. b4d5320 Remove unused/debugging code from MP4 file writer by James Dong · 16 years ago
  76. 1f90c4b Better file size estimate by James Dong · 16 years ago
  77. bd05775 Merge "Calculate audio media drift time from AudioSource" into gingerbread by James Dong · 16 years ago
  78. 34c8d61 Merge "Fix problem in AudioEffect::command() status." into gingerbread by Eric Laurent · 16 years ago
  79. aeae3de Fix problem in AudioEffect::command() status. by Eric Laurent · 16 years ago
  80. d707fcb Calculate audio media drift time from AudioSource by James Dong · 16 years ago
  81. 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
  82. e91b462 Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 16 years ago
  83. 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
  84. 6e20bdf Make sure that if initialization fails, AudioSource still behaves well. by James Dong · 16 years ago
  85. bcbe5af Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 16 years ago
  86. 82f7321 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. by Andreas Huber · 16 years ago
  87. 389636c Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 16 years ago
  88. 8f45bd7 Audio Effects: fix problems in volume control. by Eric Laurent · 16 years ago
  89. 0612475 Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 16 years ago
  90. 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
  91. 4dba3e9 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 16 years ago
  92. f74c8f9 Add some encoding parameters for the "record" utility by James Dong · 16 years ago
  93. 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
  94. 5edae61 fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 16 years ago
  95. 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 · 16 years ago
  96. 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 · 16 years ago
  97. 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 · 16 years ago
  98. 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 · 16 years ago
  99. 44eb096 Merge "Fix issue 2952766." into gingerbread by Eric Laurent · 16 years ago
  100. 541d765 Remove camera metering mode API. by Wu-cheng Li · 16 years ago