Skip to content
Snippets Groups Projects
  1. May 09, 2012
  2. Apr 17, 2012
  3. Apr 12, 2012
  4. Mar 07, 2012
  5. Feb 14, 2012
  6. Jan 20, 2012
  7. Jan 18, 2012
  8. Nov 29, 2011
  9. Sep 19, 2011
  10. Aug 05, 2011
  11. Jul 14, 2011
  12. Jul 05, 2011
    • Tilghman Lesher's avatar
      Merged revisions 326411 via svnmerge from · 7d179abf
      Tilghman Lesher authored
      https://origsvn.digium.com/svn/asterisk/branches/1.8
      
      ........
        r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines
        
        Add the attribute "type" to each "<use>" for menuselect.
        
        This matters only when autoconf fails to detect that weak linking is supported.
        External optional dependencies will become optional in both cases, as they are
        removed at compile time when not detected.  However, runtime-optional modules
        are made mandatory when weak linking is not found.  This change affects only
        the external optional dependencies; previously, they were incorrectly required
        when weak linking support was not detected.
        
        Patches:
        	20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003)
        
        Tested by: iasgoscouk
      ........
      
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7d179abf
  13. May 03, 2011
  14. Apr 20, 2011
  15. Mar 18, 2011
  16. 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
  17. 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
  18. Feb 04, 2011
  19. Feb 03, 2011
  20. Nov 05, 2010
    • Shaun Ruffell's avatar
      Merged revisions 293970 via svnmerge from · 178f3f18
      Shaun Ruffell authored
      https://origsvn.digium.com/svn/asterisk/branches/1.8
      
      ................
        r293970 | sruffell | 2010-11-04 19:07:11 -0500 (Thu, 04 Nov 2010) | 32 lines
        
        Merged revisions 293969 via svnmerge from 
        https://origsvn.digium.com/svn/asterisk/branches/1.6.2
        
        ................
          r293969 | sruffell | 2010-11-04 19:06:02 -0500 (Thu, 04 Nov 2010) | 25 lines
          
          Merged revisions 293968 via svnmerge from 
          https://origsvn.digium.com/svn/asterisk/branches/1.4
          
          ........
            r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010) | 17 lines
            
            codecs/codec_dahdi: Prevent "choppy" audio when receiving unexpected frame sizes.
            
            dahdi-linux 2.4.0 (specifically commit 9034) added the capability for
            the wctc4xxp to return more than a single packet of data in response to
            a read.  However, when decoding packets, codec_dahdi was still assuming
            that the default number of samples was in each read.
            
            In other words, each packet your provider sent you, regardless of size,
            would result in 20 ms of decoded data (30 ms if decoding G723). If your
            provider was sending 60 ms packets then codec_dahdi would end up
            stripping 40 ms of data from each transcoded frame resulting in "choppy"
            audio.
            
            This would only affect systems where G729 packets are arriving in sizes
            greater than 20ms or G723 packets arriving in sizes greater than 30ms.
            
            DAHDI-744.
          ........
        ................
      ................
      
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293971 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      178f3f18
  21. Sep 09, 2010
  22. Jun 21, 2010
  23. Jun 17, 2010
  24. Jun 03, 2010
  25. Mar 23, 2010
    • Kevin P. Fleming's avatar
      Change per-file debug and verbose levels to be per-module, the way · ae6008ef
      Kevin P. Fleming authored
      users expect them to work.
      
      'core set debug' and 'core set verbose' can optionally change the
      level for a specific filename; however, this is actually for a
      specific source file name, not the module that source file is included
      in. With examples like chan_sip, chan_iax2, chan_misdn and others
      consisting of multiple source files, this will not lead to the
      behavior that users expect. If they want to set the debug level for
      chan_sip, they want it set for all of chan_sip, and not to have to
      also set it for reqresp_parser and other files that comprise the
      chan_sip module.
      
      This patch changes this functionality to be module-name based instead
      of file-name based.
      
      To make this work, some Makefile modifications were required to ensure
      that the AST_MODULE definition is present in each object file produced
      for each module as well.
      
      Review: https://reviewboard.asterisk.org/r/574/
      
      
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      ae6008ef
  26. Mar 16, 2010
  27. Mar 15, 2010
  28. Mar 09, 2010
  29. Nov 10, 2009
  30. Nov 06, 2009
  31. Nov 04, 2009
  32. Oct 21, 2009
  33. Aug 10, 2009
Loading