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gerrit-public.fairphone.software / platform / frameworks / base / d5948d23a6a05c1c8126c67b483a08f7a364469a / media / libstagefright
  1. 94e6b66 Don't assert on unexpected surface flinger dequeue/enqueueBuffer errors, log a warning and ignore them instead. by Andreas Huber · 14 years ago
  2. f5e1faf Merge changes I71f5b0fc,I92c7accb by Nipun Kwatra · 14 years ago
  3. 4a857e6 Moving decision to use still camera to CameraSourceTimeLapse by Nipun Kwatra · 14 years ago
  4. c4e7be5 am d6fd133d: am 9077f8ec: Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 14 years ago
  5. 9077f8e Merge "Not all audio source has the drift time information" into gingerbread by James Dong · 14 years ago
  6. 1ab9d12 am 8e11c822: am 9fee0b2a: Ogg files can be tagged to be automatically looping, this setting always overrides the MediaPlayer\'s setLooping setting. by Andreas Huber · 14 years ago
  7. a093659 Merge "Add the new Stagefright ANativeWindow OMX codec API." by Jamie Gennis · 14 years ago
  8. 33a7814 Add the new Stagefright ANativeWindow OMX codec API. by Jamie Gennis · 14 years ago
  9. 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
  10. 00c88ea am af7a7c34: am cc4a38c6: Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 14 years ago
  11. cc4a38c Merge "Properly buffer a certain amount of data on streaming sources before finishing prepare()." into gingerbread by Andreas Huber · 14 years ago
  12. 87ab9cd Properly buffer a certain amount of data on streaming sources before finishing prepare(). by Andreas Huber · 14 years ago
  13. 3caa714 Not all audio source has the drift time information by James Dong · 14 years ago
  14. 9b3569b am bc1452a3: am 7755cdd6: Remove unused/debugging code from MP4 file writer by James Dong · 14 years ago
  15. 7755cdd Remove unused/debugging code from MP4 file writer by James Dong · 14 years ago
  16. 0e60f53 am 3c3fc97e: am 46e63b34: Merge "Better file size estimate" into gingerbread by James Dong · 14 years ago
  17. cb7e65c Better file size estimate by James Dong · 14 years ago
  18. 9f20d33 am bb64e554: am 7ed7668b: Merge "Calculate audio media drift time from AudioSource" into gingerbread by James Dong · 14 years ago
  19. 4c23815 Calculate audio media drift time from AudioSource by James Dong · 14 years ago
  20. 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
  21. 5aa0adc am 3fd01c4d: am d3c1bae4: Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 14 years ago
  22. 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
  23. d3c1bae Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 14 years ago
  24. 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
  25. 2198d75 Revert "Merge "Add the new Stagefright ANativeWindow OMX codec API."" by Jamie Gennis · 14 years ago
  26. a87544b Make sure that if initialization fails, AudioSource still behaves well. by James Dong · 14 years ago
  27. 8a643b4 Merge "Add the new Stagefright ANativeWindow OMX codec API." by Jamie Gennis · 14 years ago
  28. 52d14be am 47f2cf62: am 412fc7cd: Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 14 years ago
  29. 412fc7c Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 14 years ago
  30. 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
  31. dab357b Add the new Stagefright ANativeWindow OMX codec API. by Jamie Gennis · 14 years ago
  32. 4dcc6a1 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. by Andreas Huber · 14 years ago
  33. 27b9c8e Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 14 years ago
  34. 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
  35. 48ac68e Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 14 years ago
  36. 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
  37. e536f80 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 14 years ago
  38. 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
  39. 06a1d61 Added VideoSourceDownSampler by Nipun Kwatra · 14 years ago
  40. d7e7a3f Adding support for parallel recording sessions. by Nipun Kwatra · 14 years ago
  41. c855deba Merge "Make sure we only reallocate buffers on a genuine port definition change." by Andreas Huber · 14 years ago
  42. 29c03c6 Make sure we only reallocate buffers on a genuine port definition change. by Andreas Huber · 14 years ago
  43. 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
  44. 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
  45. 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
  46. 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
  47. 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
  48. 3678668 am 7ed9104c: am f6639c46: Finetune some rtsp timeout constants. by Andreas Huber · 14 years ago
  49. 631025e am 6df6d606: am df992ac9: Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 14 years ago
  50. f6639c4 Finetune some rtsp timeout constants. by Andreas Huber · 14 years ago
  51. df992ac Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 14 years ago
  52. 453f2ef Merge "client_id->clientId, bugfix for signaling of read abort on stop." by Nipun Kwatra · 14 years ago
  53. ea434da client_id->clientId, bugfix for signaling of read abort on stop. by Nipun Kwatra · 14 years ago
  54. 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
  55. c6ff7a9 am 7d3ff384: am b86365ad: Merge "Suppress the video recording start signal - bug 2950297" into gingerbread by James Dong · 14 years ago
  56. 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
  57. 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
  58. c4e0b70 ALoopers can now be named (useful to distinguish threads). by Andreas Huber · 14 years ago
  59. 90862e2 Workaround for a QCOM issue where the output buffer size advertised by the AVC encoder by James Dong · 14 years ago
  60. cf66e47 Merge "Added MediaSourceSplitter to split single source to multiple ones." by Nipun Kwatra · 14 years ago
  61. b86365a Merge "Suppress the video recording start signal - bug 2950297" into gingerbread by James Dong · 14 years ago
  62. f83cba7 Added MediaSourceSplitter to split single source to multiple ones. by Nipun Kwatra · 14 years ago
  63. 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
  64. d6a4004 We accidentally always aborted after 10 secs, even if the connection was fine. by Andreas Huber · 14 years ago
  65. d7f1c3d Suppress the video recording start signal - bug 2950297 by James Dong · 14 years ago
  66. 178e1d0 am 74ae6973: am 17a765a1: Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 14 years ago
  67. 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
  68. d222c84 Disable all the hardware decoders except for h.264 video decode, since the software decoders are faster. by Andreas Huber · 14 years ago
  69. 17a765a Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 14 years ago
  70. 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
  71. 0416da7 Support for RTP packets arriving interleaved with RTSP responses. by Andreas Huber · 14 years ago
  72. 318a759 Merge "Make sure that timestamp does not go backward in MP4 file writer" into gingerbread by James Dong · 14 years ago
  73. 05643cc am e58cd37d: am 8ac0983e: Merge "Fix support for per-frame unsynchronization in ID3V2.4 tags." into gingerbread by Andreas Huber · 14 years ago
  74. 8ac0983 Merge "Fix support for per-frame unsynchronization in ID3V2.4 tags." into gingerbread by Andreas Huber · 14 years ago
  75. bf4c0c8 am cec075cc: am 23584022: Merge "Ensure that buffering updates eventually hit 100% after we download everything." into gingerbread by Andreas Huber · 14 years ago
  76. 8735f89 Fix support for per-frame unsynchronization in ID3V2.4 tags. by Andreas Huber · 14 years ago
  77. 2358402 Merge "Ensure that buffering updates eventually hit 100% after we download everything." into gingerbread by Andreas Huber · 14 years ago
  78. a164410 am 96dc4559: am b8814dce: 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
  79. 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
  80. c23296e Ensure that buffering updates eventually hit 100% after we download everything. by Andreas Huber · 14 years ago
  81. 8d9d751 am 67ca90b3: am 6b6ae996: Merge "A first shot at proper support for seeking of rtsp streams." into gingerbread by Andreas Huber · 14 years ago
  82. e0dd7d3 A first shot at proper support for seeking of rtsp streams. by Andreas Huber · 14 years ago
  83. 05e80b4 Make sure that timestamp does not go backward in MP4 file writer by James Dong · 14 years ago
  84. 804539b am 31e71131: am 3e22ef1e: Merge "Better handling of rtsp connection and disconnection." into gingerbread by Andreas Huber · 14 years ago
  85. 7741ecc am 28a92120: am 3f51fa78: Runtime dump support for MediaWriter by James Dong · 14 years ago
  86. 7802b205 am 3fc01525: am b755e325: Merge "Only add 4 bytes offset for the output media buffer when SPS is not received for SW AVC encoder" into gingerbread by James Dong · 14 years ago
  87. 7fdaa23 Merge "Account for the _ADRENO constant being moved." by Andreas Huber · 14 years ago
  88. c6c9b49 Account for the _ADRENO constant being moved. by Jamie Gennis · 14 years ago
  89. 7f81d4c Merge changes Ic94c18a6,Iff770de1,Ifed6b4dc by Dima Zavin · 14 years ago
  90. d535076 Merge "Squashed commit of the following:" by Andreas Huber · 14 years ago
  91. e3c0183 Squashed commit of the following: by Andreas Huber · 14 years ago
  92. 3e22ef1 Merge "Better handling of rtsp connection and disconnection." into gingerbread by Andreas Huber · 14 years ago
  93. 37e2592 am 3540760d: am 0ea4ed3b: Don\'t drop a late frame which may lead to missing I frames in the MP4 file by James Dong · 14 years ago
  94. a75d87f am 177a7ad8: am 439fe407: Merge "Return error from MPEG4Writer stop() if the check on codec specific data failed" into gingerbread by James Dong · 14 years ago
  95. 8370be1 Better handling of rtsp connection and disconnection. by Andreas Huber · 14 years ago
  96. 30ba6cb libstagefright: enable tegra hw audio decoders by Dima Zavin · 14 years ago
  97. 6ad2c35 libstagefright: Enable tegra hw video decoders by pgudadhe · 14 years ago
  98. 3f51fa7 Runtime dump support for MediaWriter by James Dong · 14 years ago
  99. 349250f am c8d2fa70: am cbd038fe: Merge "Make MediaWriter stop and pause return errors if necessary" into gingerbread by James Dong · 14 years ago
  100. 83b3e35 am 873ebfb8: am 223e4f73: Merge "Support for MP4V-ES packetization format according to RFC3016." into gingerbread by Andreas Huber · 14 years ago