1. d707fcb Calculate audio media drift time from AudioSource by James Dong · 16 years ago
  2. 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 · 16 years ago
  3. e91b462 Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread by James Dong · 16 years ago
  4. 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 · 16 years ago
  5. 6e20bdf Make sure that if initialization fails, AudioSource still behaves well. by James Dong · 16 years ago
  6. bcbe5af Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread by Andreas Huber · 16 years ago
  7. 82f7321 Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. by Andreas Huber · 16 years ago
  8. 389636c Keep gtalk video chat specific code consistent with rtsp changes. by Andreas Huber · 16 years ago
  9. 8f45bd7 Audio Effects: fix problems in volume control. by Eric Laurent · 16 years ago
  10. 0612475 Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread by Andreas Huber · 16 years ago
  11. 4dba3e9 Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. by Andreas Huber · 16 years ago
  12. 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 · 16 years ago
  13. 5edae61 fixedfft: Only includes cpu-features.h when __arm__ is defined. by Chia-chi Yeh · 16 years ago
  14. 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 · 16 years ago
  15. 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 · 16 years ago
  16. 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 · 16 years ago
  17. 1c842b2 Merge "Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore." into gingerbread by Andreas Huber · 16 years ago
  18. a1ffe49 Reverse the default setting of media.stagefright.enable-{rtsp,record} in preparation for building without opencore. by Andreas Huber · 16 years ago
  19. 3849699 Merge "LVM release 1.07 delivery." into gingerbread by Eric Laurent · 16 years ago
  20. e56121b Finetune some rtsp timeout constants. by Andreas Huber · 16 years ago
  21. c01ec02 Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread by Andreas Huber · 16 years ago
  22. a814c1f ALoopers can now be named (useful to distinguish threads). by Andreas Huber · 16 years ago
  23. 824c9ff Workaround for a QCOM issue where the output buffer size advertised by the AVC encoder by James Dong · 16 years ago
  24. 352c468 Merge "Suppress the video recording start signal - bug 2950297" into gingerbread by James Dong · 16 years ago
  25. 8d34297 Better support for rtsp (normal play-)time display. Better seek support, timeout if no packets arrive for too long. by Andreas Huber · 16 years ago
  26. d918324 LVM release 1.07 delivery. by Eric Laurent · 16 years ago
  27. cc6adf5 We accidentally always aborted after 10 secs, even if the connection was fine. by Andreas Huber · 16 years ago
  28. f1ae196 Suppress the video recording start signal - bug 2950297 by James Dong · 16 years ago
  29. 1b07372 Merge "Support for RTP packets arriving interleaved with RTSP responses." into gingerbread by Andreas Huber · 16 years ago
  30. 0792ce7 Support for RTP packets arriving interleaved with RTSP responses. by Andreas Huber · 16 years ago
  31. 640a72e Merge "Make sure that timestamp does not go backward in MP4 file writer" into gingerbread by James Dong · 16 years ago
  32. f302743 Merge "Fix support for per-frame unsynchronization in ID3V2.4 tags." into gingerbread by Andreas Huber · 16 years ago
  33. f0bfaa8 Merge "Added preset reverb." into gingerbread by Eric Laurent · 16 years ago
  34. ac994df Fix support for per-frame unsynchronization in ID3V2.4 tags. by Andreas Huber · 16 years ago
  35. 81ce489 Merge "Ensure that buffering updates eventually hit 100% after we download everything." into gingerbread by Andreas Huber · 16 years ago
  36. 3f71e8b 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 · 16 years ago
  37. 5a1c352 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 · 16 years ago
  38. 2a4b49b Ensure that buffering updates eventually hit 100% after we download everything. by Andreas Huber · 16 years ago
  39. 625981e Fix the simulator build. by Eric Laurent · 16 years ago
  40. 97344f1 Added preset reverb. by Eric Laurent · 16 years ago
  41. 094370f Merge "A first shot at proper support for seeking of rtsp streams." into gingerbread by Andreas Huber · 16 years ago
  42. cce326f A first shot at proper support for seeking of rtsp streams. by Andreas Huber · 16 years ago
  43. 40e9940 Make sure that timestamp does not go backward in MP4 file writer by James Dong · 16 years ago
  44. afb68fb Merge "LVM release 1.05 delivery" into gingerbread by Eric Laurent · 16 years ago
  45. 01f0908 Merge "Better handling of rtsp connection and disconnection." into gingerbread by Andreas Huber · 16 years ago
  46. 1b54324 Better handling of rtsp connection and disconnection. by Andreas Huber · 16 years ago
  47. dedf414 Runtime dump support for MediaWriter by James Dong · 16 years ago
  48. 4824824 Merge "Visualizer: replace the FFT implementation with a faster one." into gingerbread by Chia-chi Yeh · 16 years ago
  49. c59c6fd LVM release 1.05 delivery by Eric Laurent · 16 years ago
  50. be6fd20 setParamMaxFileDurationUs should allow zero time input as per API of setMaxDuration. by Nipun Kwatra · 16 years ago
  51. 76c6e8f Only add 4 bytes offset for the output media buffer when SPS is not received for SW AVC encoder by James Dong · 16 years ago
  52. 4f86a98 Don't drop a late frame which may lead to missing I frames in the MP4 file by James Dong · 16 years ago
  53. 690f546 Return error from MPEG4Writer stop() if the check on codec specific data failed by James Dong · 16 years ago
  54. e0b77ce Merge "Make MediaWriter stop and pause return errors if necessary" into gingerbread by James Dong · 16 years ago
  55. 3718791 Make MediaWriter stop and pause return errors if necessary by James Dong · 16 years ago
  56. 62cb04d Support for MP4V-ES packetization format according to RFC3016. by Andreas Huber · 16 years ago
  57. c0f7ec8 Merge "In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data." into gingerbread by Andreas Huber · 16 years ago
  58. 4478c01 Merge "Audio Effects: fixed "strength supported" parameter size." into gingerbread by Eric Laurent · 16 years ago
  59. 85f12e9 In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data. by Andreas Huber · 16 years ago
  60. 34c9ad8 Merge "Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description." into gingerbread by Andreas Huber · 16 years ago
  61. dbd2b7e Visualizer: replace the FFT implementation with a faster one. by Chia-chi Yeh · 16 years ago
  62. 7fa8a0e Audio Effects: fixed "strength supported" parameter size. by Eric Laurent · 16 years ago
  63. ef7af7f Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description. by Andreas Huber · 16 years ago
  64. d683105 Merge "Make the OggExtractor less verbose." into gingerbread by Andreas Huber · 16 years ago
  65. a65ccbf Be more lenient when validating ESDS information in mp4 audio tracks. Allow the absence of any codec specific data and assume that the mpeg4 headers are not lying to us. by Andreas Huber · 16 years ago
  66. 3e8843e Make the OggExtractor less verbose. by Andreas Huber · 16 years ago
  67. c059860 Change the default time scale for audio/video track during recording by James Dong · 16 years ago
  68. e259531 Use audio clock as the reference media clock by James Dong · 16 years ago
  69. abfedeb Mainly fix two mistakes that I made: by James Dong · 16 years ago
  70. b9299d1 Merge "Support getting codec, width, and height in URL for gtalk playback." into gingerbread by Mike Dodd · 16 years ago
  71. 60d77eb Fix software avc encoder crash at EOS. by Andreas Huber · 16 years ago
  72. 1aa4fed Support getting codec, width, and height in URL for gtalk playback. by Mike Dodd · 16 years ago
  73. d1753e3 Merge "Fix the h.263 assembler to properly subset a buffer's range if it already has a range applied." into gingerbread by Andreas Huber · 16 years ago
  74. 9089ba6 Merge "APacketSource is too verbose." into gingerbread by Andreas Huber · 16 years ago
  75. a6238a1 Fix the h.263 assembler to properly subset a buffer's range if it already has a range applied. by Andreas Huber · 16 years ago
  76. 5328abe Don't send late frames to software encoders for encoding by James Dong · 16 years ago
  77. bef4b56 Merge "Handle large audio lost" into gingerbread by James Dong · 16 years ago
  78. 7704377 Merge "Fix a crash due to unnecessary check on the codec config data for H263 video track" into gingerbread by James Dong · 16 years ago
  79. 910a17f Merge "Fix all fd leaks in authoring engine" into gingerbread by James Dong · 16 years ago
  80. 6dc387a APacketSource is too verbose. by Andreas Huber · 16 years ago
  81. 25d8312 Fix all fd leaks in authoring engine by James Dong · 16 years ago
  82. a37923e Fix a crash due to unnecessary check on the codec config data for H263 video track by James Dong · 16 years ago
  83. 4d074e8 Merge "This code in CameraSource really should hold the lock." into gingerbread by Andreas Huber · 16 years ago
  84. 56223b9 This code in CameraSource really should hold the lock. by Andreas Huber · 16 years ago
  85. 47204e1 Handle large audio lost by James Dong · 16 years ago
  86. eaf0e07 Remove some obsolete code. by Andreas Huber · 16 years ago
  87. 7010190 Merge "Add input buffer size check for software video encoders" into gingerbread by James Dong · 16 years ago
  88. 8aacc47 Merge "We're now going to ignore timestamps completely in gtalk video conferencing, playing video as soon as it comes in. We also make up fake timestamps in the rtp code, ignoring rtcp SR information to enable early startup." into gingerbread by Andreas Huber · 16 years ago
  89. ab68c7f Merge "Many, many developers misread or don't read the http specs and terminate lines with '\n' instead of CRLF '\r\n' as required. Enable the workaround for this by default. Also increase the socket read timeout to 30 secs." into gingerbread by Andreas Huber · 16 years ago
  90. f8ca904 We're now going to ignore timestamps completely in gtalk video conferencing, playing video as soon as it comes in. We also make up fake timestamps in the rtp code, ignoring rtcp SR information to enable early startup. by Andreas Huber · 16 years ago
  91. db004e1 Add input buffer size check for software video encoders by James Dong · 16 years ago
  92. 51dfe6d Only check the codec specific data when the output buffer contains kKeyIsCodecConfig in MP4 writer by James Dong · 16 years ago
  93. 2097f53 Many, many developers misread or don't read the http specs and terminate lines with '\n' instead of CRLF '\r\n' as required. Enable the workaround for this by default. Also increase the socket read timeout to 30 secs. by Andreas Huber · 16 years ago
  94. def871d Merge "Support multiple PPS and SPS in avcC box" into gingerbread by James Dong · 16 years ago
  95. 3266b2c Support multiple PPS and SPS in avcC box by James Dong · 16 years ago
  96. e758feb Merge "Support for extracting G.711 a-law and mu-law audio from WAV files and a corresponding software decoder." into gingerbread by Andreas Huber · 16 years ago
  97. 42d66f2 Support for extracting G.711 a-law and mu-law audio from WAV files and a corresponding software decoder. by Andreas Huber · 16 years ago
  98. 8569f0d Fixed several audio effects problems. by Eric Laurent · 16 years ago
  99. 8bf59e7 Support for communicating if a buffer read from the _extractor_ is a sync sample or not. by Andreas Huber · 16 years ago
  100. 2500c2d Properly skip the 'skip' chunk in mpeg4 files. by Andreas Huber · 16 years ago