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  1. May 03, 2011
  2. Apr 20, 2011
  3. Mar 18, 2011
  4. 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
  5. 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
  6. Feb 04, 2011
  7. Feb 03, 2011
  8. 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
  9. Sep 09, 2010
  10. Jun 21, 2010
  11. Jun 17, 2010
  12. Jun 03, 2010
  13. 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
  14. Mar 16, 2010
  15. Mar 15, 2010
  16. Mar 09, 2010
  17. Nov 10, 2009
  18. Nov 06, 2009
  19. Nov 04, 2009
  20. Oct 21, 2009
  21. Aug 10, 2009
  22. Jul 30, 2009
  23. Jul 21, 2009
    • Kevin P. Fleming's avatar
      Merged revisions 207647 via svnmerge from · 96e4e31e
      Kevin P. Fleming authored
      https://origsvn.digium.com/svn/asterisk/branches/1.4
      
      ........
        r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines
        
        Ensure that user-provided CFLAGS and LDFLAGS are honored.
        
        This commit changes the build system so that user-provided flags (in ASTCFLAGS
        and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
        by the build system itself, so that the user can effectively override the
        build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
        be provided *either* in the environment before running 'make', or as variable
        assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
        is no longer necessary, so they are no longer documented, but are still supported
        so as not to break existing build systems that supply them when building Asterisk.
      ........
      
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      96e4e31e
  24. Jul 15, 2009
  25. Jun 18, 2009
  26. May 15, 2009
  27. Feb 17, 2009
    • Shaun Ruffell's avatar
      Several changes to codec_dahdi to play nice with G723. · c65029b7
      Shaun Ruffell authored
      This commit brings in the changes that were living out on the
      svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch.  codec_dahdi.c now
      always uses signed linear as the simple codec so that a soft g729 codec will
      not end up being preferred to the hardware codec.  There are also changes to
      allow codec_dahdi.c to feed packets to the hardware in the native sample size of
      the codec.  This solves problems with choppy audio when using G723. 
      
      
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      c65029b7
  28. Nov 20, 2008
    • Kevin P. Fleming's avatar
      Merged revisions 157859 via svnmerge from · 8d5deb31
      Kevin P. Fleming authored
      https://origsvn.digium.com/svn/asterisk/branches/1.4
      
      ........
        r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines
        
        the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems.
        
        with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course).
        
        while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain
      ........
      
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      8d5deb31
  29. Oct 30, 2008
  30. Oct 17, 2008
  31. Oct 15, 2008
  32. Sep 12, 2008
    • Tilghman Lesher's avatar
      Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiating · 08af5bb3
      Tilghman Lesher authored
      when a file is invalid from when a file is missing.  This is most important when
      we have two configuration files.  Consider the following example:
      
      Old system:
      sip.conf     users.conf     Old result               New result
      ========     ==========     ==========               ==========
      Missing      Missing        SIP doesn't load         SIP doesn't load
      Missing      OK             SIP doesn't load         SIP doesn't load
      Missing      Invalid        SIP doesn't load         SIP doesn't load
      OK           Missing        SIP loads                SIP loads
      OK           OK             SIP loads                SIP loads
      OK           Invalid        SIP loads incompletely   SIP doesn't load
      Invalid      Missing        SIP doesn't load         SIP doesn't load
      Invalid      OK             SIP doesn't load         SIP doesn't load
      Invalid      Invalid        SIP doesn't load         SIP doesn't load
      
      So in the case when users.conf doesn't load because there's a typo that
      disrupts the syntax, we may only partially load users, instead of failing with
      an error, which may cause some calls not to get processed.  Worse yet, the old
      system would do this with no indication that anything was even wrong.
      
      (closes issue #10690)
       Reported by: dtyoo
       Patches: 
             20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)
      
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      08af5bb3
  33. Sep 02, 2008
  34. Aug 20, 2008
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