Gitiles
Code Review Sign In
review.blissroms.org / platform_frameworks_av / 5ee6bb5a8665cf09aa6266012d6ba7d405886724
  1. 5ee6bb5 Merge "Fix issue 2913071." into gingerbread by Eric Laurent · 15 years ago
  2. a312142 Merge "This log message is codec specific." into gingerbread by Andreas Huber · 15 years ago
  3. 43d4f74 Merge "Remove stagefright foundation's incompatible logging interface and update callsites." into gingerbread by Andreas Huber · 15 years ago
  4. 6e4c5c4 Remove stagefright foundation's incompatible logging interface and update callsites. by Andreas Huber · 15 years ago
  5. 0e75f0f Fix issue 2913071. by Eric Laurent · 15 years ago
  6. 955194d This log message is codec specific. by Andreas Huber · 15 years ago
  7. e936413 Merge "Allow record to set input color format as a command line option" into gingerbread by James Dong · 15 years ago
  8. 425587d Merge "Another attempt for fixing AAC+/eAAC+ related issue" into gingerbread by James Dong · 15 years ago
  9. ac4205c Rename FOCUS_MODE_CONTINUOUS to FOCUS_MODE_CONTINUOUS_VIDEO. by Wu-cheng Li · 15 years ago
  10. 1826945 Another attempt for fixing AAC+/eAAC+ related issue by James Dong · 15 years ago
  11. a733679 Allow record to set input color format as a command line option by James Dong · 15 years ago
  12. 1c1503c Add a check to track a problem the monkey script has been triggering. by Marco Nelissen · 15 years ago
  13. 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
  14. 095916d Register the new OMX components. by Andreas Huber · 15 years ago
  15. 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
  16. 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
  17. 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
  18. 104fcb8 Make sure the .wav extractor does not read data outside the bounds of the 'data' box. by Andreas Huber · 15 years ago
  19. 0270f47 Merge "Fixed a bug in the query to the supported profiles and levels" into gingerbread by James Dong · 15 years ago
  20. f01691f Fixed a bug in the query to the supported profiles and levels by James Dong · 15 years ago
  21. 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
  22. 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
  23. 70fb57d Merge "Fix problem in lvm effect bundle wrapper" into gingerbread by Eric Laurent · 15 years ago
  24. 29cc743 Fix problem in lvm effect bundle wrapper by Eric Laurent · 15 years ago
  25. eae6193 Merge "Upgrade to the latest .webm project code." into gingerbread by Andreas Huber · 15 years ago
  26. acf67ea Upgrade to the latest .webm project code. by Andreas Huber · 15 years ago
  27. d790910 Merge "Add some explicit error log messages" into gingerbread by James Dong · 15 years ago
  28. e78d3bb Merge "Fix audio input sample timestamp when audio driver loses audio samples" into gingerbread by James Dong · 15 years ago
  29. 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
  30. 6f85dba Various fixes to improve resilience of the rtsp stack against spurious errors instead of asserting. by Andreas Huber · 15 years ago
  31. a1abc1a Add some explicit error log messages by James Dong · 15 years ago
  32. 67e9269 Fix audio input sample timestamp when audio driver loses audio samples by James Dong · 15 years ago
  33. e0aed6d Fix volume problems with insert revert by Eric Laurent · 15 years ago
  34. a175413 Merge "LVM release 1.09 delivery" into gingerbread by Eric Laurent · 15 years ago
  35. acb5621 TimedEventQueue now explicitly sets its scheduling policy to foreground as it should. by Andreas Huber · 15 years ago
  36. 5185b01 LVM release 1.09 delivery by Eric Laurent · 15 years ago
  37. 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
  38. 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
  39. 49110ce Instead of asserting return a runtime error if the maximum sample size cannot be determined. by Andreas Huber · 15 years ago
  40. 772bcc2 Instead of asserting, publish no tracks if an MP3Extractor is used on non-mp3 content. by Andreas Huber · 15 years ago
  41. d2518e0 When 32-bit offset is used, by James Dong · 15 years ago
  42. fbf7162 Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread by James Dong · 15 years ago
  43. 3c3763d HW audio encoder expects timestamp via kKeyTime from each input buffer by James Dong · 15 years ago
  44. 54c38fd Modify type of some environmental reverb parameters by Eric Laurent · 15 years ago
  45. f9c0ae8 Merge "LVM release 1.08 delivery." into gingerbread by Eric Laurent · 15 years ago
  46. ddba3f0 Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 15 years ago
  47. 2d3bf53 LVM release 1.08 delivery. by Eric Laurent · 15 years ago
  48. 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
  49. 1a4c79e Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 15 years ago
  50. 8650e19 Properly buffer a certain amount of data on streaming sources before finishing prepare(). by Andreas Huber · 15 years ago
  51. caa68a5 Not all audio source has the drift time information by James Dong · 15 years ago
  52. b4d5320 Remove unused/debugging code from MP4 file writer by James Dong · 15 years ago
  53. 1f90c4b Better file size estimate by James Dong · 15 years ago
  54. bd05775 Merge "Calculate audio media drift time from AudioSource" into gingerbread by James Dong · 15 years ago
  55. 34c8d61 Merge "Fix problem in AudioEffect::command() status." into gingerbread by Eric Laurent · 15 years ago
  56. aeae3de Fix problem in AudioEffect::command() status. by Eric Laurent · 15 years ago
  57. d707fcb Calculate audio media drift time from AudioSource by James Dong · 15 years ago
  58. 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
  59. e91b46246 Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 15 years ago
  60. 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
  61. 6e20bdf Make sure that if initialization fails, AudioSource still behaves well. by James Dong · 15 years ago
  62. bcbe5af Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 15 years ago
  63. 82f7321 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. by Andreas Huber · 15 years ago
  64. 389636c Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 15 years ago
  65. 8f45bd7 Audio Effects: fix problems in volume control. by Eric Laurent · 15 years ago
  66. 0612475 Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 15 years ago
  67. 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
  68. 4dba3e9 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 15 years ago
  69. f74c8f9 Add some encoding parameters for the "record" utility by James Dong · 15 years ago
  70. 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
  71. 5edae61 fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 15 years ago
  72. 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
  73. 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
  74. 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
  75. 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
  76. 44eb096 Merge "Fix issue 2952766." into gingerbread by Eric Laurent · 15 years ago
  77. 541d765 Remove camera metering mode API. by Wu-cheng Li · 15 years ago
  78. 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
  79. a1ffe49 Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore. by Andreas Huber · 15 years ago
  80. 3849699 Merge "LVM release 1.07 delivery." into gingerbread by Eric Laurent · 15 years ago
  81. e56121b Finetune some rtsp timeout constants. by Andreas Huber · 15 years ago
  82. c01ec02 Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 15 years ago
  83. c1c88e2 Fix issue 2952766. by Eric Laurent · 15 years ago
  84. a814c1f ALoopers can now be named (useful to distinguish threads). by Andreas Huber · 15 years ago
  85. 824c9ff Workaround for a QCOM issue where the output buffer size advertised by the AVC encoder by James Dong · 15 years ago
  86. 352c468 Merge "Suppress the video recording start signal - bug 2950297" into gingerbread by James Dong · 15 years ago
  87. 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
  88. 165dc4c Merge "We accidentally always aborted after 10 secs, even if the connection was fine." into gingerbread by Andreas Huber · 15 years ago
  89. 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
  90. d918324 LVM release 1.07 delivery. by Eric Laurent · 15 years ago
  91. 14cc6fc Merge "fix a race in SF buffer management" into gingerbread by Mathias Agopian · 15 years ago
  92. cc6adf5 We accidentally always aborted after 10 secs, even if the connection was fine. by Andreas Huber · 15 years ago
  93. d7c43d3 fix a race in SF buffer management by Mathias Agopian · 15 years ago
  94. f1ae196 Suppress the video recording start signal - bug 2950297 by James Dong · 15 years ago
  95. 1b07372 Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 15 years ago
  96. 0792ce7 Support for RTP packets arriving interleaved with RTSP responses. by Andreas Huber · 15 years ago
  97. 640a72e Merge "Make sure that timestamp does not go backward in MP4 file writer" into gingerbread by James Dong · 15 years ago
  98. f302743 Merge "Fix support for per-frame unsynchronization in ID3V2.4 tags." into gingerbread by Andreas Huber · 15 years ago
  99. f0bfaa8 Merge "Added preset reverb." into gingerbread by Eric Laurent · 15 years ago
  100. ac994df Fix support for per-frame unsynchronization in ID3V2.4 tags. by Andreas Huber · 15 years ago