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20 authors
  1. Mar 28, 2015
    • Matthew Jordan's avatar
      clang compiler warnings: Fix a variety of "unused" warnings · 29e6597f
      Matthew Jordan authored 
      This patch fixes the -Wunused-value -Wunused-variable -Wunused-const-variable
errors caught by clang. Specifically:

* apps/app_queue.c: removed unused qpm_cmd_usage[], qum_cmd_usage[],
                    qsmp_cmd_usage[]
* cel/cel_sqlite3_custom.c: removed unused name[] = "cel_sqlite3_custom"
* codecs/gsm/src/gsm_create.c: removed unused ident[] = "$Header$"
* funcs/func_env.c:729: Fixed ast_str_append_substr.
* main/editline/np/strlcat.c: removed unused rcsid variable
* main/editline/np/strlcpy.c: removed unused rcsid variable
* utils/conf2ael.c: removed unused cfextension_states
* utils/extconf.c: removed unused cfextension_states

Review: https://reviewboard.asterisk.org/r/4526

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4526.patch submitted by dkdegroot (License 6600)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@433693 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      29e6597f
  3. Oct 24, 2013
  5. Apr 12, 2013
  7. Mar 19, 2013
  9. Jan 21, 2013
  11. Dec 07, 2012
    • Kinsey Moore's avatar
      codec_dahdi: Fix output of "transcoder show" CLI command. · 7ce2ba52
      Kinsey Moore authored 
      In r306010 "Asterisk media architecture conversion - no more format
bitfields", the logic for incrementing encoders and decoders when
opening transcoder channels was changed without making the corresponding
change when decrementing encoder / decoder channels.  The result being
that when a channel was destroyed, codec_dahdi couldn't properly tell if
it was an encoder or decoder, and the default case is to assume it was a
decoder.

This could result in negative numbers for decoders in use like in:
  VOIP6*CLI> transcoder show
  2/-2 encoders/decoders of 92 channels are in use.

(closes issue ASTERISK-19921)
Patch-by: Shaun Ruffell
........

Merged revisions 377382 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377383 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7ce2ba52
  13. Oct 22, 2012
  15. Sep 25, 2012
  17. Jul 23, 2012
  19. Jun 15, 2012
    • Kevin P. Fleming's avatar
      Multiple revisions 369001-369002 · 166b4e2b
      Kevin P. Fleming authored 
      ........
  r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines
  
  Add support-level indications to many more source files.
  
  Since we now have tools that scan through the source tree looking for files
  with specific support levels, we need to ensure that every file that is
  a component of a 'core' or 'extended' module (or the main Asterisk binary)
  is explicitly marked with its support level. This patch adds support-level
  indications to many more source files in tree, but avoids adding them to
  third-party libraries that are included in the tree and to source files
  that don't end up involved in Asterisk itself.
........
  r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines
  
  Add a script to enable finding source files without support-levels defined.
........

Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 369005 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      166b4e2b
  21. May 09, 2012
  23. Apr 17, 2012
  25. Apr 12, 2012
  27. Mar 07, 2012
  29. Feb 14, 2012
  31. Jan 20, 2012
  33. Jan 18, 2012
  35. Nov 29, 2011
  37. Sep 19, 2011
  39. Aug 05, 2011
  41. Jul 14, 2011
  43. 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
  45. May 03, 2011
  47. Apr 20, 2011
  49. Mar 18, 2011
  51. 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
  53. 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
  55. Feb 04, 2011
  57. Feb 03, 2011
  59. 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
  61. Sep 09, 2010
  63. Jun 21, 2010
  65. Jun 17, 2010