1. 8eb9141 More ticket #542: Updated default capture latency (tested OK on Vista 64bit for WMME & dsound) by Nanang Izzuddin · 16 years ago
  2. 32177c0 Large changeset to replace all occurences of year 2007 with 2008 in the copyright notice by Benny Prijono · 16 years ago
  3. 5b64b8d Updated doxygen documentations by Benny Prijono · 16 years ago
  4. df008d3 Make SDP parser more lenient by ignoring first newlines. by Benny Prijono · 16 years ago
  5. 224b4e2 Ticket #549: major modification in media transport API to support more offer/answer scenarios by Benny Prijono · 16 years ago
  6. 6472e1b Cleaned up unused field avg_jitter from rtcp session by Nanang Izzuddin · 16 years ago
  7. 495409d Added pjmedia_tonegen_rewind() by Benny Prijono · 16 years ago
  8. dd3d002 Implement ticket #546 and revisit ticket #439: by Benny Prijono · 16 years ago
  9. 4d49c78 More ticket #505: added constants to shrink the excess frames in jbuf even more slowly, default can only discard one excess frame per 200ms by Benny Prijono · 16 years ago
  10. 89ac2b4 Fixed various compilation warnings with gcc strict compilation by Benny Prijono · 16 years ago
  11. cbd38c6 Related to ticket #525: transport_srtp returns PJ_EINVALIDOP in UPDATE or re-INVITE when media is already active by Benny Prijono · 16 years ago
  12. b0dd211 More ticket #542: updated dsound.c to adapt with latency setting by Nanang Izzuddin · 16 years ago
  13. d7fefd7 Ticket #542: added new API and macro for sound device latency settings, also added new param for this in pjsua by Nanang Izzuddin · 16 years ago
  14. 9dbad15 Updated default speex quality settings and reenabled pjsua to set Speex codec quality based on media quality config by Nanang Izzuddin · 16 years ago
  15. fd461eb Added more validations & a new API (thanks Florian Bomers): by Nanang Izzuddin · 16 years ago
  16. 848be08 More ticket #485: major modification in transport_ice to support new ICE stream transport API by Benny Prijono · 16 years ago
  17. e4b4b7d Added field maximum bitrate to codec param, this is useful for providing safer frame size calculation, especially when peer's bitrate is unknown by Nanang Izzuddin · 16 years ago
  18. eb4c616 Cleaned up warnings of [u]char-[u]int conversions by Nanang Izzuddin · 16 years ago
  19. 3aef5e1 Ticket #473: by Nanang Izzuddin · 16 years ago
  20. 78e2591 Fixed bug in copying buffer and updated post process of buffer shrinking by Nanang Izzuddin · 16 years ago
  21. 7ca463c Fixed C & C++ cross linked problem and missing newly added libsrtp.lib in symbian_ua_gui.mmp by Nanang Izzuddin · 16 years ago
  22. d424297 Fixed uninitialized output size before calling G.722 encoder by Nanang Izzuddin · 16 years ago
  23. 700e838 Updated delaybuf to learn burst level in realtime instead of only in the beginning, this can optimize the latency and increase adaptivity by Nanang Izzuddin · 16 years ago
  24. 1ec45bf Added another WSOLA implementation, PJMEDIA_WSOLA_IMP_WSOLA_LITE, this is used by small devices by default (replacing PJMEDIA_WSOLA_IMP_NULL) by Nanang Izzuddin · 16 years ago
  25. b6242b7 Fixed bug in pjmedia_sdp_rtpmap_to_attr(): may overwrite past the buffer by Benny Prijono · 16 years ago
  26. 829ac02 Changed build optimizations settings for Speex, pjmedia, and symbian_sound for Symbian. Speex/8000 now also runs on Symbian! by Nanang Izzuddin · 16 years ago
  27. dfd3052 Changed build optimization settings for Speex and pjmedia for eVC4/Windows Mobile. Speex/8000 now runs on Windows Mobile! by Benny Prijono · 16 years ago
  28. 2d4ee7d More on ticket #535: updated files using and related to math.h by Nanang Izzuddin · 16 years ago
  29. 3fd3af9 Resample port get frame may cause buffer overflow when downport returns non-audio frame by Nanang Izzuddin · 16 years ago
  30. a5538ab Ticket #527: Commited ticket527.2.patch by Nanang Izzuddin · 16 years ago
  31. 5d070f8 More ticket #513: by Nanang Izzuddin · 16 years ago
  32. db9da77 Ticket #528: committed ticket528.patch by Nanang Izzuddin · 16 years ago
  33. 90f11cb More on ticket #513: by Nanang Izzuddin · 16 years ago
  34. e8b604d Ticket #513: Support for RTCP XR (initial patch) by Benny Prijono · 16 years ago
  35. 6780ae0 More ticket #497: added configuration to disable WSOLA and set default to disabled on WinCE and Symbian by Benny Prijono · 16 years ago
  36. 6b7834b More ticket #420: creation of IPv6 media streams (tested on SIPit22) by Benny Prijono · 16 years ago
  37. e7d5a10 More ticket #526: committed ticket526.2.patch by Benny Prijono · 16 years ago
  38. 24a2185 Fixed miscellaneous compile warnings/errors when built with C++ mode by Benny Prijono · 16 years ago
  39. 97a5759 Ticket #526: Pjsua crash after wav player destroyed inside the eof callback (thanks Tanguy Floc'h) by Benny Prijono · 16 years ago
  40. 427d145 More ticket #505: the jitter buffer only discard one packet at a time when optimizing the delay by Benny Prijono · 16 years ago
  41. aeb187d More ticket #523: renamed some variable names for clarity by Benny Prijono · 16 years ago
  42. 88efec5 Ticket #523: Handle incomplete audio frame from sound device (e.g. OSS) by Benny Prijono · 16 years ago
  43. f161655 More ticket #504: added missing new files! by Benny Prijono · 16 years ago
  44. 03c5c69 Ticket #517: Invalid argument error when binding media transport on MacOS X by Benny Prijono · 16 years ago
  45. 7d60d05 Ticket #504: final installment to support stereo audio all the way in PJMEDIA. Please see tickiet #504 for more info by Benny Prijono · 16 years ago
  46. a171e9e Fixed bug in automatic RTCP address calculation causing assertion failure in Symbian with ICE transport by Benny Prijono · 16 years ago
  47. cf5c06d Updated pjmedia_test with the latest jitter buffer API by Benny Prijono · 17 years ago
  48. 70119f6 Ticket #516: Assertion in sound device when headset is plugged/unplugged in MacOS X (thanks Alexei Kuznetsov) by Benny Prijono · 17 years ago
  49. 096fadb Ticket #505: optimizing the jitter buffer delay by Benny Prijono · 17 years ago
  50. 522e5e1 Fixed minor warning about unused variable in transport_srtp.c by Benny Prijono · 17 years ago
  51. 5e24839 More ticket #497: division by zero in wsola when min_extra is set to zero (the default value) by Benny Prijono · 17 years ago
  52. 734fc2d More ticket #479: bug in pjmedia_transport_get_info(), the info should be initialized by caller by Benny Prijono · 17 years ago
  53. f9f17b1 More ticket #507: fixed the bug introduced in r1871 about operator precedence by Benny Prijono · 17 years ago
  54. 71f657d More ticket #507: updated GNU build system with the G.722 codec and fixed minor warning about operator precedence by Benny Prijono · 17 years ago
  55. 7ffd775 Ticket #507: committed and tested g722 patch on Windows by Benny Prijono · 17 years ago
  56. 6e7c5ad More ticket #504: buffer overflow in splitcomb when handling stereo audio by Benny Prijono · 17 years ago
  57. e11c581 Ticket #504: fixed stream.c for stereo codecs by Benny Prijono · 17 years ago
  58. e1a5a85 Ticket #479: allow media transport framework to return transport specific info (for example, to know whether SRTP is enabled) by Benny Prijono · 17 years ago
  59. c12bc10 Ticket #494: Configuration option to use high quality tone generation by Benny Prijono · 17 years ago
  60. 8cc996a Ticket #504: assertion in sound device and recfile sample when using stereo by Benny Prijono · 17 years ago
  61. a99539c More ticket #438: wrong param passed to shrink_buffer() in set_max_cnt(), should be (buf_cnt - new_max_cnt), instead of (old_max_cnt - new_max_cnt) by Benny Prijono · 17 years ago
  62. 00d15a5 Ticket #502: New packet lost concealment (PLC) implementation and enable PLC on G711 and GSM codec by Benny Prijono · 17 years ago
  63. c97d686 More ticket #497: bug on pjmedia_wsola_save() when extra samples exceed samples_per_frame by Benny Prijono · 17 years ago
  64. 98b6df8 More ticket #438: improve docs, added channel_count in wsola, etc. by Benny Prijono · 17 years ago
  65. d34477c Ticket #501: Set master port worker thread priority to highest by default by Benny Prijono · 17 years ago
  66. f6654c5 Ticket #438: added delaybuf in splitcomb. Please see the checkin comment in ticket #438 for the details by Benny Prijono · 17 years ago
  67. 93be976 Ticket #499: NULL frame transmission in conference bridge is not clocked at the right interval by Benny Prijono · 17 years ago
  68. e8ec158 Ticket #497: changed clock rate variable in WSOLA from uint16 to uint32 to handle higher clock rate by Benny Prijono · 17 years ago
  69. 161ae3a Disable overflow/underflow test by Benny Prijono · 17 years ago
  70. 65afd88 More ticket #438: changed API call to delay_buf_create() by Benny Prijono · 17 years ago
  71. 07f6cc7 Added checking in delaybuf for buffer empty before calling shrink_buffer() by Benny Prijono · 17 years ago
  72. 5dbd4fc Ticket #438: Workaround for frame bursts from audio devices: added wsola in delaybuf, and put delaybuf in the bridge by Benny Prijono · 17 years ago
  73. 5887d02 Put wsola_test.c in pjmedia test by Benny Prijono · 17 years ago
  74. 031775c Added pjmedia_wsola_reset() by Benny Prijono · 17 years ago
  75. 4c8475d Fixed bug in wsola when discarding frame in non-contiguous buffer by Benny Prijono · 17 years ago
  76. c8f43b3 Modify WSOLA discard to support erasing frame from non-contiguous buffer by Benny Prijono · 17 years ago
  77. 713ccab A little bit of optimization in WSOLA by Benny Prijono · 17 years ago
  78. 800521c Ticket #496: Crash on sound port when only player is opened and delaybuf enabled by Benny Prijono · 17 years ago
  79. 4727a9a Ticket #497: WSOLA implementation by Benny Prijono · 17 years ago
  80. 7df2a15 Ticket #486: Handle G.722 wong clock rate bug and other codec with inconsistent clock rate by Benny Prijono · 17 years ago
  81. 2dbed82 Ticket #467: fixed issues with RTP/AVP vs RTP/SAVP negotiation by Benny Prijono · 17 years ago
  82. d72d686 Ticket #487: Crash occured when stream port has no transmitter on call using ILBC with different ptime by Benny Prijono · 17 years ago
  83. c5b6de1 Ticket #468: Added support for non looping playback in memory player by Benny Prijono · 17 years ago
  84. f430e41 Allow receiving SDP with lines terminated with LF instead of CRLF (thanks Juri Glass) by Benny Prijono · 17 years ago
  85. a1179ca Ticket #480: PJSIP rejects incoming call with m=image in the offer (thanks Thiago) by Benny Prijono · 17 years ago
  86. 45930a5 Ticket #478: Handle duplicated/misordered incoming DTMF packets by Benny Prijono · 17 years ago
  87. e3994fb More ticket #469: bail out from on_rx_rtp/rtcp loop if recvfrom returns PJ_ECANCELLED. This errno is returned when the key is mark as closing, which could happen when user closes the key inside the callback by Benny Prijono · 17 years ago
  88. 91476bf Handle short files in WAV player by Benny Prijono · 17 years ago
  89. 69036b7 Ticket #460: Concurrency problem when destroying stream (thanks Michael Broughton) by Benny Prijono · 17 years ago
  90. 378484d Removed const on on_rx_rtp() and on_rx_rtcp() functions arguments in stream.c and srtp.c (const was removed in r1763) by Benny Prijono · 17 years ago
  91. 527a236 More ticket #61: removed const from rtp and rtcp callback function declaration to allow in-place packet modification by Benny Prijono · 17 years ago
  92. 4085b1f Added media transport diagram for documentation by Benny Prijono · 17 years ago
  93. 6665bfc Ticket #464: jitter buffer should return frame length information by Benny Prijono · 17 years ago
  94. 4802871 Ticket #61: undo r1759 and replace it with other patches to fix unable to accept RTP/AVP offer with a=crypto attribute when use_sdp is set to zero. Also minor fix to streamutil by Benny Prijono · 17 years ago
  95. af1f56c Ticket #61: bug, unable to accept RTP/AVP with crypto attribute when use_srtp is set to zero by Benny Prijono · 17 years ago
  96. a7b376b Fixed doxygen comments everywhere by Benny Prijono · 17 years ago
  97. dc72f4c Ticket #459: pjmedia_clock_create() should not create thread when PJMEDIA_CLOCK_NO_ASYNC is given (thanks Alberto Takeshi Mayama) by Benny Prijono · 17 years ago
  98. 04218f3 More ticket #61: fix potential error when looking up SRTP error string, and improve logging information in SRTP transport by Benny Prijono · 17 years ago
  99. 389a769 More ticket #61: crashed on SRTP error string lookup by Benny Prijono · 17 years ago
  100. 2c42375 Ticket #61: added Windows Mobile/WinCE eVC project for SRTP and libsrtp by Benny Prijono · 17 years ago