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gerrit-public.fairphone.software / platform / frameworks / base / f83d4f1779eb4801275a56023c343459a95009af / media
  1. 53d7765 am fd0eed00: am a2511da9: 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
  2. 5aa0adc am 3fd01c4d: am d3c1bae4: Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 14 years ago
  3. fd0eed0 am a2511da9: 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
  4. 3fd01c4 am d3c1bae4: Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 14 years ago
  5. 5b98831 Merge "resolved conflicts for merge of 71c908c4 to master" by Andreas Huber · 14 years ago
  6. ee9c3db resolved conflicts for merge of 71c908c4 to master by Andreas Huber · 14 years ago
  7. 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
  8. 8d89cda Merge "Support for auxiliary video parameters." by Nipun Kwatra · 14 years ago
  9. d3c1bae Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 14 years ago
  10. 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
  11. 2198d75 Revert "Merge "Add the new Stagefright ANativeWindow OMX codec API."" by Jamie Gennis · 14 years ago
  12. a87544b Make sure that if initialization fails, AudioSource still behaves well. by James Dong · 14 years ago
  13. 8a643b4 Merge "Add the new Stagefright ANativeWindow OMX codec API." by Jamie Gennis · 14 years ago
  14. 71c908c am 6c33904a: 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
  15. 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
  16. 239f2e5 Support for auxiliary video parameters. by Nipun Kwatra · 14 years ago
  17. 52d14be am 47f2cf62: am 412fc7cd: Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 14 years ago
  18. 47f2cf6 am 412fc7cd: Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 14 years ago
  19. 412fc7c Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 14 years ago
  20. 8d7d413 Now that AmrInputStream no longer relies on opencore, make sure it's registered in non-opencore builds. by Andreas Huber · 14 years ago
  21. 564a9f2 am 021a822e: am de2b1615: Merge "Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer." into gingerbread by Andreas Huber · 14 years ago
  22. dab357b Add the new Stagefright ANativeWindow OMX codec API. by Jamie Gennis · 14 years ago
  23. 021a822 am de2b1615: Merge "Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer." into gingerbread by Andreas Huber · 14 years ago
  24. 4dcc6a1 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. by Andreas Huber · 14 years ago
  25. 27b9c8e Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 14 years ago
  26. 288db3b am 55e79374: am f560ceab: Merge "Audio Effects: fix problems in volume control." into gingerbread by Eric Laurent · 14 years ago
  27. 55e7937 am f560ceab: Merge "Audio Effects: fix problems in volume control." into gingerbread by Eric Laurent · 14 years ago
  28. a92ebfa Audio Effects: fix problems in volume control. by Eric Laurent · 14 years ago
  29. 03cf220 am 6b52911c: am 48ac68e1: Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 14 years ago
  30. 6b52911 am 48ac68e1: Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 14 years ago
  31. 48ac68e Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 14 years ago
  32. 5f39972 am e1a3cddd: am 99fa510e: 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 · 14 years ago
  33. e1a3cdd am 99fa510e: 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 · 14 years ago
  34. f06a449 Merge "MTP: Send an Intent after an MTP session that resulted in media database modifications" by Mike Lockwood · 14 years ago
  35. e536f80 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 14 years ago
  36. d20e802 Merge "Add settings option for running MTP server in PTP mode." by Mike Lockwood · 14 years ago
  37. 2837eef MTP: Send an Intent after an MTP session that resulted in media database modifications by Mike Lockwood · 14 years ago
  38. eabe8bf Add settings option for running MTP server in PTP mode. by Mike Lockwood · 14 years ago
  39. 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
  40. 06a1d61 Added VideoSourceDownSampler by Nipun Kwatra · 14 years ago
  41. b33a5ae Added setAuxiliaryOutputFile to MediaRecorder and JNI by Nipun Kwatra · 14 years ago
  42. d7e7a3f Adding support for parallel recording sessions. by Nipun Kwatra · 14 years ago
  43. 2deeff1 am 1577e629: am 12006013: fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 14 years ago
  44. c855deba Merge "Make sure we only reallocate buffers on a genuine port definition change." by Andreas Huber · 14 years ago
  45. 1577e62 am 12006013: fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 14 years ago
  46. 1200601 fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 14 years ago
  47. 29c03c6 Make sure we only reallocate buffers on a genuine port definition change. by Andreas Huber · 14 years ago
  48. 47416bc am 03e83d4a: am 68ae91cb: 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
  49. 03e83d4 am 68ae91cb: 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
  50. 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
  51. 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
  52. 6924563 am 987556bc: am abb8398e: 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 · 14 years ago
  53. 987556b am abb8398e: 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 · 14 years ago
  54. 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
  55. 2527da0 am 9aa05ec2: am 681c5ff2: 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
  56. 9d01dcb am 5762dc19: am 858bb4f6: Merge "LVM release 1.07 delivery." into gingerbread by Eric Laurent · 14 years ago
  57. 3678668 am 7ed9104c: am f6639c46: Finetune some rtsp timeout constants. by Andreas Huber · 14 years ago
  58. 631025e am 6df6d606: am df992ac9: Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 14 years ago
  59. 9aa05ec am 681c5ff2: 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
  60. 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
  61. 30cfa20 Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore. by Andreas Huber · 14 years ago
  62. 5762dc1 am 858bb4f6: Merge "LVM release 1.07 delivery." into gingerbread by Eric Laurent · 14 years ago
  63. 7ed9104 am f6639c46: Finetune some rtsp timeout constants. by Andreas Huber · 14 years ago
  64. 6df6d60 am df992ac9: Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 14 years ago
  65. 858bb4f Merge "LVM release 1.07 delivery." into gingerbread by Eric Laurent · 14 years ago
  66. f6639c4 Finetune some rtsp timeout constants. by Andreas Huber · 14 years ago
  67. 5367ab6 MTP: Move all MTP protocol constants for Java to a central location. by Mike Lockwood · 14 years ago
  68. df992ac Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 14 years ago
  69. daeeddc MTP: Fix build break due to misuse of "git add -i" command by Mike Lockwood · 14 years ago
  70. af2ba83 Merge "MTP: Add try/finally handling to finalize overrides" by Mike Lockwood · 14 years ago
  71. dbead32 MTP: Add try/finally handling to finalize overrides by Mike Lockwood · 14 years ago
  72. 453f2ef Merge "client_id->clientId, bugfix for signaling of read abort on stop." by Nipun Kwatra · 14 years ago
  73. ea434da client_id->clientId, bugfix for signaling of read abort on stop. by Nipun Kwatra · 14 years ago
  74. 206bf9d am a5fe77d0: am df8356ff: Merge "Workaround for a QCOM issue where the output buffer size advertised by the AVC encoder is occasionally too small." into gingerbread by James Dong · 14 years ago
  75. c6ff7a9 am 7d3ff384: am b86365ad: Merge "Suppress the video recording start signal - bug 2950297" into gingerbread by James Dong · 14 years ago
  76. 8abd425 am 05c1cada: am 577615c9: 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 · 14 years ago
  77. 84ecebb am e25e0361: am e250c220: Merge "We accidentally always aborted after 10 secs, even if the connection was fine." into gingerbread by Andreas Huber · 14 years ago
  78. c4e0b70 ALoopers can now be named (useful to distinguish threads). by Andreas Huber · 14 years ago
  79. a5fe77d am df8356ff: Merge "Workaround for a QCOM issue where the output buffer size advertised by the AVC encoder is occasionally too small." into gingerbread by James Dong · 14 years ago
  80. 90862e2 Workaround for a QCOM issue where the output buffer size advertised by the AVC encoder by James Dong · 14 years ago
  81. 7d3ff38 am b86365ad: Merge "Suppress the video recording start signal - bug 2950297" into gingerbread by James Dong · 14 years ago
  82. cf66e47 Merge "Added MediaSourceSplitter to split single source to multiple ones." by Nipun Kwatra · 14 years ago
  83. b86365a Merge "Suppress the video recording start signal - bug 2950297" into gingerbread by James Dong · 14 years ago
  84. f83cba7 Added MediaSourceSplitter to split single source to multiple ones. by Nipun Kwatra · 14 years ago
  85. 05c1cad am 577615c9: 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 · 14 years ago
  86. e25e036 am e250c220: Merge "We accidentally always aborted after 10 secs, even if the connection was fine." into gingerbread by Andreas Huber · 14 years ago
  87. 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
  88. adecf1c LVM release 1.07 delivery. by Eric Laurent · 14 years ago
  89. d6a4004 We accidentally always aborted after 10 secs, even if the connection was fine. by Andreas Huber · 14 years ago
  90. d7f1c3d Suppress the video recording start signal - bug 2950297 by James Dong · 14 years ago
  91. 178e1d0 am 74ae6973: am 17a765a1: Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 14 years ago
  92. 0a84244 am d0faf5a2: am c6d7a895: Merge "Changed type of reverb presets from int to short" into gingerbread by Eric Laurent · 14 years ago
  93. 8ed880d Merge "Disable all the hardware decoders except for h.264 video decode, since the software decoders are faster." by Andreas Huber · 14 years ago
  94. d222c84 Disable all the hardware decoders except for h.264 video decode, since the software decoders are faster. by Andreas Huber · 14 years ago
  95. 74ae697 am 17a765a1: Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 14 years ago
  96. 17a765a Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 14 years ago
  97. 832aa58 am 2db73305: am dfded35b: Merge "Added automated tests for reverb audio effect." into gingerbread by Eric Laurent · 14 years ago
  98. 8c26c12 am 9509a0ce: am 318a759e: Merge "Make sure that timestamp does not go backward in MP4 file writer" into gingerbread by James Dong · 14 years ago
  99. 0416da7 Support for RTP packets arriving interleaved with RTSP responses. by Andreas Huber · 14 years ago
  100. d0faf5a am c6d7a895: Merge "Changed type of reverb presets from int to short" into gingerbread by Eric Laurent · 14 years ago