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20 authors
  1. Mar 28, 2015
    • Matthew Jordan's avatar
      clang compiler warnings: Fix -Wself-assign · baa010e8
      Matthew Jordan authored 
      Assigning a variable to itself isn't super useful. However, the WAV format
modules make use of this in order to perform byte endian checks. This patch
works around the warning by only performing the self assignment if we are
going to do more than just assign it to ourselves. Which is odd, but true.

ASTERISK-24917
Reported by: dkdegroot
patches:
  rb4544.patch submitted by dkdegroot (License 6600)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@433690 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      baa010e8
  3. May 09, 2014
  5. Feb 05, 2014
  7. Jun 07, 2012
  9. Apr 29, 2012
  11. Apr 17, 2012
  13. Apr 16, 2012
  15. Apr 06, 2012
  17. Feb 16, 2012
  19. Feb 14, 2012
  21. Feb 08, 2012
  23. Nov 09, 2011
  25. Jul 29, 2011
  27. Jul 14, 2011
  29. Jul 08, 2011
  31. Jul 07, 2011
  33. May 16, 2011
  35. May 03, 2011
  37. Apr 25, 2011
  39. 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
  41. Feb 03, 2011
  43. Sep 02, 2010
  45. Jul 26, 2010
  47. Jul 20, 2010
  49. Jun 16, 2010
  51. Apr 20, 2010
  53. Dec 08, 2009
    • Russell Bryant's avatar
      Set a module load priority for format modules. · 8ab22f5d
      Russell Bryant authored 
      A recent change to app_voicemail made it such that the module now assumes that
all format modules are available while processing voicemail configuration.
However, when autoloading modules, it was possible that app_voicemail was
loaded before the format modules.  Since format modules don't depend on
anything, set a module load priority on them to ensure that they get loaded
first when autoloading.

This fix applies to trunk, 1.6.1, and 1.6.2.  The fix for 1.4 and 1.6.0 will
require a different approach since the module load priority functionality is
not present in the module API.

(issue #16412)
Reported by: jiddings


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      8ab22f5d
  55. Nov 04, 2009
  57. Oct 19, 2009
  59. Jun 15, 2009
  61. May 21, 2009
    • Kevin P. Fleming's avatar
      Const-ify the world (or at least a good part of it) · e6b2e9a7
      Kevin P. Fleming authored 
      This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:

- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments

In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.

Review: https://reviewboard.asterisk.org/r/251/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e6b2e9a7
  63. Apr 08, 2009
  65. Feb 15, 2009
  67. Feb 13, 2009
    • Kevin P. Fleming's avatar
      Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C... · 2a53f2ec
      Kevin P. Fleming authored 
      Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)

This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.

Along the way, some related work was done:

1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.

2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.

3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).

Review: http://reviewboard.digium.com/r/158/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      2a53f2ec
  69. Dec 08, 2008
  71. Nov 02, 2008