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gerrit-public.fairphone.software / platform / frameworks / base / 2509976269fbd9fbd0ec3561e20fdfbd9b897a09 / media
  1. d6877fa Merge "AMRNB: use Frame_Type_3GPP defined in frame_type_3gpp.h instead." into gingerbread by Chia-chi Yeh · 14 years ago
  2. 081833d AMRNB: use Frame_Type_3GPP defined in frame_type_3gpp.h instead. by Chia-chi Yeh · 14 years ago
  3. 4fd3ecc Fix several audio effects problems. by Eric Laurent · 14 years ago
  4. 88a995e Merge "Properly flush the AudioTrack/AudioSink on a seek request and make sure that both the mp3 decoder and aac software decoders start fresh after a seek without any dependency on previously decoded content." into gingerbread by Andreas Huber · 14 years ago
  5. ad3fcfe Properly flush the AudioTrack/AudioSink on a seek request and make sure that both the mp3 decoder and aac software decoders start fresh after a seek without any dependency on previously decoded content. by Andreas Huber · 14 years ago
  6. be04506 Merge "Instead of constantly polling the AudioPlayer to see if it reached EOS or finished seeking, initiate the notification from the AudioPlayer when the event happens." into gingerbread by Andreas Huber · 14 years ago
  7. 2b359ed Instead of constantly polling the AudioPlayer to see if it reached EOS or finished seeking, initiate the notification from the AudioPlayer when the event happens. by Andreas Huber · 14 years ago
  8. 4769f57 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 · 14 years ago
  9. 3a9cc8c Merge "Squashed commit of the following:" into gingerbread by Andreas Huber · 14 years ago
  10. 38ae220 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 · 14 years ago
  11. 124a346 Fix media.player dumpsys to output open/mapped files correctly. Bug 2866669. by Dave Sparks · 14 years ago
  12. c751ecc Squashed commit of the following: by Andreas Huber · 14 years ago
  13. bf47092 Merge "Fix track duration calculation if the start timestamp is non-zero" into gingerbread by James Dong · 14 years ago
  14. dacebe6 Fix track duration calculation if the start timestamp is non-zero by James Dong · 14 years ago
  15. abaabb2 Merge "Support other kinds of HTTP redirect in NuHTTPDataSource" into gingerbread by Andreas Huber · 14 years ago
  16. ab2116c Support other kinds of HTTP redirect in NuHTTPDataSource by Andreas Huber · 14 years ago
  17. 4d3fb50 Fix error in AudioEffect command status reporting. by Eric Laurent · 14 years ago
  18. 2fb43ef fix problem in AudioEffect JNI setup. by Eric Laurent · 14 years ago
  19. 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
  20. 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
  21. 1a5149e Fix issue 3022800. by Eric Laurent · 14 years ago
  22. 903fc22 Ignore errors from correction parameter query and config for M4v and H263 encoders by James Dong · 14 years ago
  23. 1e0e166 Use the advertised profile and level from M4V and H263 video encoders by James Dong · 14 years ago
  24. 3efbc55 Merge "Fix more audio effects auto tests" into gingerbread by Eric Laurent · 14 years ago
  25. 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
  26. ec1e9c7 Fix more audio effects auto tests by Eric Laurent · 14 years ago
  27. 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
  28. 4f29455 Merge "Fix broken insert reverb auto tests." into gingerbread by Eric Laurent · 14 years ago
  29. 71fe631 Fix broken insert reverb auto tests. by Eric Laurent · 14 years ago
  30. a6dc469 Merge "Fix issue 2913071." into gingerbread by Eric Laurent · 14 years ago
  31. 87d208f Merge "This log message is codec specific." into gingerbread by Andreas Huber · 14 years ago
  32. 6773848 Merge "Remove stagefright foundation's incompatible logging interface and update callsites." into gingerbread by Andreas Huber · 14 years ago
  33. 6e3fa44 Remove stagefright foundation's incompatible logging interface and update callsites. by Andreas Huber · 14 years ago
  34. 672c0dc Fix issue 2913071. by Eric Laurent · 14 years ago
  35. af0a188 This log message is codec specific. by Andreas Huber · 14 years ago
  36. 8d1513e Merge "Made audio effect control panel intents public." into gingerbread by Eric Laurent · 14 years ago
  37. 92cf2d6 Made audio effect control panel intents public. by Eric Laurent · 14 years ago
  38. 40da64f Another attempt for fixing AAC+/eAAC+ related issue by James Dong · 14 years ago
  39. 3c473ea Add a check to track a problem the monkey script has been triggering. by Marco Nelissen · 14 years ago
  40. 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
  41. 524e6f6 Register the new OMX components. by Andreas Huber · 14 years ago
  42. 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
  43. 4f5bb1e Make sure the .wav extractor does not read data outside the bounds of the 'data' box. by Andreas Huber · 14 years ago
  44. a7f5e47 Merge "Fixed a bug in the query to the supported profiles and levels" into gingerbread by James Dong · 14 years ago
  45. dfb8991 Fixed a bug in the query to the supported profiles and levels by James Dong · 14 years ago
  46. 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
  47. 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
  48. 37de5da Merge "Fix problem in lvm effect bundle wrapper" into gingerbread by Eric Laurent · 14 years ago
  49. f0f95b8 Fix problem in lvm effect bundle wrapper by Eric Laurent · 14 years ago
  50. 24a2c2b Merge "Upgrade to the latest .webm project code." into gingerbread by Andreas Huber · 14 years ago
  51. e5f8539 Upgrade to the latest .webm project code. by Andreas Huber · 14 years ago
  52. 82a39f4 Merge "Add some explicit error log messages" into gingerbread by James Dong · 14 years ago
  53. 9f882c0 Merge "Fix audio input sample timestamp when audio driver loses audio samples" into gingerbread by James Dong · 14 years ago
  54. 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
  55. f3d2bdf Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting. by Andreas Huber · 14 years ago
  56. b5024da Add some explicit error log messages by James Dong · 14 years ago
  57. 7589ebf Fix audio input sample timestamp when audio driver loses audio samples by James Dong · 14 years ago
  58. d09af7d Added intents for audio effects control application by Eric Laurent · 14 years ago
  59. 27a2fdf Fix volume problems with insert revert by Eric Laurent · 14 years ago
  60. 4709c7f Merge "Fixed a copy and paste error" into gingerbread by James Dong · 14 years ago
  61. 2814ad2 Merge "LVM release 1.09 delivery" into gingerbread by Eric Laurent · 14 years ago
  62. 8d3b910 Fixed a copy and paste error by James Dong · 14 years ago
  63. 4b3d32b TimedEventQueue now explicitly sets its scheduling policy to foreground as it should. by Andreas Huber · 14 years ago
  64. 305443c LVM release 1.09 delivery by Eric Laurent · 14 years ago
  65. 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
  66. 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
  67. d7f2225 Instead of asserting return a runtime error if the maximum sample size cannot be determined. by Andreas Huber · 14 years ago
  68. 3e0f2be Instead of asserting, publish no tracks if an MP3Extractor is used on non-mp3 content. by Andreas Huber · 14 years ago
  69. a4fb816 When 32-bit offset is used, by James Dong · 14 years ago
  70. d353c84 Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 14 years ago
  71. d015ccf HW audio encoder expects timestamp via kKeyTime from each input buffer by James Dong · 14 years ago
  72. 95d5de0 Modify type of some environmental reverb parameters by Eric Laurent · 14 years ago
  73. 7e42793 Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 14 years ago
  74. 9077f8e Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 14 years ago
  75. 5fa6df6 LVM release 1.08 delivery. by Eric Laurent · 14 years ago
  76. 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
  77. cc4a38c Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 14 years ago
  78. 87ab9cd Properly buffer a certain amount of data on streaming sources before finishing prepare(). by Andreas Huber · 14 years ago
  79. 3caa714 Not all audio source has the drift time information by James Dong · 14 years ago
  80. 7755cdd Remove unused/debugging code from MP4 file writer by James Dong · 14 years ago
  81. cb7e65c Better file size estimate by James Dong · 14 years ago
  82. 4c23815 Calculate audio media drift time from AudioSource by James Dong · 14 years ago
  83. 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
  84. d3c1bae Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 14 years ago
  85. 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
  86. a87544b Make sure that if initialization fails, AudioSource still behaves well. by James Dong · 14 years ago
  87. 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
  88. 412fc7c Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 14 years ago
  89. 8d7d413 Now that AmrInputStream no longer relies on opencore, make sure it's registered in non-opencore builds. by Andreas Huber · 14 years ago
  90. 4dcc6a1 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. by Andreas Huber · 14 years ago
  91. 27b9c8e Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 14 years ago
  92. a92ebfa Audio Effects: fix problems in volume control. by Eric Laurent · 14 years ago
  93. 48ac68e Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 14 years ago
  94. e536f80 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 14 years ago
  95. 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
  96. 1200601 fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 14 years ago
  97. 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
  98. 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
  99. 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
  100. 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