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20 authors
  1. Mar 11, 2011
    • Kevin P. Fleming's avatar
      Use "-march=native" when possible. · 7cf70df9
      Kevin P. Fleming authored 
      Recent versions of GCC have a tuning option value of 'native', which causes
the compiler to optimize the build for the CPU the compile is performed on.
Since most people are building Asterisk on the machine they plan to run it on,
the configure script and build system will now use this value unless a different
value is specified by the user in CFLAGS when the configure script is executed.
In addition, this value will be used for building the GSM and LPC10 codecs as
well, in preference to the logic that has been in their Makefiles forever to
optimize for certain types of CPUs.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7cf70df9
    • Alec L Davis's avatar
      Merged revisions 310287 via svnmerge from · c7c0664b
      Alec L Davis authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r310287 | alecdavis | 2011-03-11 19:47:44 +1300 (Fri, 11 Mar 2011) | 17 lines
  
  remote_bridge_loop: prevent segfault when after transfer of IAX2 of DAHDI call 
  
  If the channel condition is one of the following after breaking out of the loop, don't try to update_peer
  (where x = 0/1)
   1). ZOMBIE
   2). cx->tech_pvt != pvtx
   3). gluex != ast_rtp_instance_get_glue(cx->tech->type))
  
  (closes issue #18781)
  Reported by: alecdavis
  Patches: 
        bug18781.diff3.txt uploaded by alecdavis (license 585)
  Tested by: alecdavis, ZX81
  
  Review: https://reviewboard.asterisk.org/r/1128/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      c7c0664b
  3. Mar 10, 2011
  5. Mar 08, 2011
  7. Mar 07, 2011
  9. Mar 05, 2011
  11. Mar 04, 2011
  13. Mar 02, 2011
  15. Mar 01, 2011
  17. Feb 28, 2011
  19. Feb 25, 2011
  21. Feb 24, 2011
  23. Feb 23, 2011
  25. Feb 22, 2011
    • David Vossel's avatar
      Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd... · d760e81f
      David Vossel authored 
      Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff

-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
   using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
   and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate.  This allows
   for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
   updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.

-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c

Review: https://reviewboard.asterisk.org/r/1104/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      d760e81f
    • Andrew Latham's avatar
      Use ast_debug for console logging · 736133f8
      Andrew Latham authored 
      Guessed the log levels based on info that level 3
is the soft roof.  Can we create a page / document
to define the levels?



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      736133f8
  27. Feb 21, 2011