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gerrit-public.fairphone.software / platform / frameworks / base / 3b4bac7a2a940483c81a6bfda21039e02e62157d / media
  1. 3c473ea Add a check to track a problem the monkey script has been triggering. by Marco Nelissen · 14 years ago
  2. 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
  3. 524e6f6 Register the new OMX components. by Andreas Huber · 14 years ago
  4. 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
  5. 4f5bb1e Make sure the .wav extractor does not read data outside the bounds of the 'data' box. by Andreas Huber · 14 years ago
  6. a7f5e47 Merge "Fixed a bug in the query to the supported profiles and levels" into gingerbread by James Dong · 14 years ago
  7. dfb8991 Fixed a bug in the query to the supported profiles and levels by James Dong · 14 years ago
  8. 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
  9. 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
  10. 37de5da Merge "Fix problem in lvm effect bundle wrapper" into gingerbread by Eric Laurent · 14 years ago
  11. f0f95b8 Fix problem in lvm effect bundle wrapper by Eric Laurent · 14 years ago
  12. 24a2c2b Merge "Upgrade to the latest .webm project code." into gingerbread by Andreas Huber · 14 years ago
  13. e5f8539 Upgrade to the latest .webm project code. by Andreas Huber · 14 years ago
  14. 82a39f4 Merge "Add some explicit error log messages" into gingerbread by James Dong · 14 years ago
  15. 9f882c0 Merge "Fix audio input sample timestamp when audio driver loses audio samples" into gingerbread by James Dong · 14 years ago
  16. 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
  17. f3d2bdf Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting. by Andreas Huber · 14 years ago
  18. b5024da Add some explicit error log messages by James Dong · 14 years ago
  19. 7589ebf Fix audio input sample timestamp when audio driver loses audio samples by James Dong · 14 years ago
  20. d09af7d Added intents for audio effects control application by Eric Laurent · 14 years ago
  21. 27a2fdf Fix volume problems with insert revert by Eric Laurent · 14 years ago
  22. 4709c7f Merge "Fixed a copy and paste error" into gingerbread by James Dong · 14 years ago
  23. 2814ad2 Merge "LVM release 1.09 delivery" into gingerbread by Eric Laurent · 14 years ago
  24. 8d3b910 Fixed a copy and paste error by James Dong · 14 years ago
  25. 4b3d32b TimedEventQueue now explicitly sets its scheduling policy to foreground as it should. by Andreas Huber · 14 years ago
  26. 305443c LVM release 1.09 delivery by Eric Laurent · 14 years ago
  27. 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
  28. 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
  29. d7f2225 Instead of asserting return a runtime error if the maximum sample size cannot be determined. by Andreas Huber · 14 years ago
  30. 3e0f2be Instead of asserting, publish no tracks if an MP3Extractor is used on non-mp3 content. by Andreas Huber · 14 years ago
  31. a4fb816 When 32-bit offset is used, by James Dong · 14 years ago
  32. d353c84 Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 14 years ago
  33. d015ccf HW audio encoder expects timestamp via kKeyTime from each input buffer by James Dong · 14 years ago
  34. 95d5de0 Modify type of some environmental reverb parameters by Eric Laurent · 14 years ago
  35. 7e42793 Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 14 years ago
  36. 9077f8e Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 14 years ago
  37. 5fa6df6 LVM release 1.08 delivery. by Eric Laurent · 14 years ago
  38. 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
  39. cc4a38c Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 14 years ago
  40. 87ab9cd Properly buffer a certain amount of data on streaming sources before finishing prepare(). by Andreas Huber · 14 years ago
  41. 3caa714 Not all audio source has the drift time information by James Dong · 14 years ago
  42. 7755cdd Remove unused/debugging code from MP4 file writer by James Dong · 14 years ago
  43. cb7e65c Better file size estimate by James Dong · 14 years ago
  44. 4c23815 Calculate audio media drift time from AudioSource by James Dong · 14 years ago
  45. 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
  46. d3c1bae Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 14 years ago
  47. 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
  48. a87544b Make sure that if initialization fails, AudioSource still behaves well. by James Dong · 14 years ago
  49. 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
  50. 412fc7c Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 14 years ago
  51. 8d7d413 Now that AmrInputStream no longer relies on opencore, make sure it's registered in non-opencore builds. by Andreas Huber · 14 years ago
  52. 4dcc6a1 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. by Andreas Huber · 14 years ago
  53. 27b9c8e Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 14 years ago
  54. a92ebfa Audio Effects: fix problems in volume control. by Eric Laurent · 14 years ago
  55. 48ac68e Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 14 years ago
  56. e536f80 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 14 years ago
  57. 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
  58. 1200601 fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 14 years ago
  59. 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
  60. 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
  61. 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
  62. 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
  63. 30cfa20 Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore. by Andreas Huber · 14 years ago
  64. 858bb4f Merge "LVM release 1.07 delivery." into gingerbread by Eric Laurent · 14 years ago
  65. f6639c4 Finetune some rtsp timeout constants. by Andreas Huber · 14 years ago
  66. df992ac Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 14 years ago
  67. c4e0b70 ALoopers can now be named (useful to distinguish threads). by Andreas Huber · 14 years ago
  68. 90862e2 Workaround for a QCOM issue where the output buffer size advertised by the AVC encoder by James Dong · 14 years ago
  69. b86365a Merge "Suppress the video recording start signal - bug 2950297" into gingerbread by James Dong · 14 years ago
  70. 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
  71. adecf1c LVM release 1.07 delivery. by Eric Laurent · 14 years ago
  72. d6a4004 We accidentally always aborted after 10 secs, even if the connection was fine. by Andreas Huber · 14 years ago
  73. d7f1c3d Suppress the video recording start signal - bug 2950297 by James Dong · 14 years ago
  74. 17a765a Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 14 years ago
  75. 0416da7 Support for RTP packets arriving interleaved with RTSP responses. by Andreas Huber · 14 years ago
  76. 71450f8 Changed type of reverb presets from int to short by Eric Laurent · 14 years ago
  77. dfded35 Merge "Added automated tests for reverb audio effect." into gingerbread by Eric Laurent · 14 years ago
  78. 318a759 Merge "Make sure that timestamp does not go backward in MP4 file writer" into gingerbread by James Dong · 14 years ago
  79. 391e2d0 Added automated tests for reverb audio effect. by Eric Laurent · 14 years ago
  80. 8ac0983 Merge "Fix support for per-frame unsynchronization in ID3V2.4 tags." into gingerbread by Andreas Huber · 14 years ago
  81. c14f9ca Merge "Added preset reverb." into gingerbread by Eric Laurent · 14 years ago
  82. 8735f89 Fix support for per-frame unsynchronization in ID3V2.4 tags. by Andreas Huber · 14 years ago
  83. 2358402 Merge "Ensure that buffering updates eventually hit 100% after we download everything." into gingerbread by Andreas Huber · 14 years ago
  84. b8814dc 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 · 14 years ago
  85. efdd088 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 · 14 years ago
  86. c23296e Ensure that buffering updates eventually hit 100% after we download everything. by Andreas Huber · 14 years ago
  87. cd295c1 Fix the simulator build. by Eric Laurent · 14 years ago
  88. a7e5648 Added preset reverb. by Eric Laurent · 14 years ago
  89. 6b6ae99 Merge "A first shot at proper support for seeking of rtsp streams." into gingerbread by Andreas Huber · 14 years ago
  90. e0dd7d3 A first shot at proper support for seeking of rtsp streams. by Andreas Huber · 14 years ago
  91. 05e80b4 Make sure that timestamp does not go backward in MP4 file writer by James Dong · 14 years ago
  92. b6d7135 Merge "LVM release 1.05 delivery" into gingerbread by Eric Laurent · 14 years ago
  93. 3e22ef1 Merge "Better handling of rtsp connection and disconnection." into gingerbread by Andreas Huber · 14 years ago
  94. 8370be1 Better handling of rtsp connection and disconnection. by Andreas Huber · 14 years ago
  95. 3f51fa7 Runtime dump support for MediaWriter by James Dong · 14 years ago
  96. b80e610 Merge "Visualizer: replace the FFT implementation with a faster one." into gingerbread by Chia-chi Yeh · 14 years ago
  97. a1a96f3 LVM release 1.05 delivery by Eric Laurent · 14 years ago
  98. fb45748 setParamMaxFileDurationUs should allow zero time input as per API of setMaxDuration. by Nipun Kwatra · 14 years ago
  99. 9767dbf Only add 4 bytes offset for the output media buffer when SPS is not received for SW AVC encoder by James Dong · 14 years ago
  100. 0ea4ed3 Don't drop a late frame which may lead to missing I frames in the MP4 file by James Dong · 14 years ago