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This patch fixes numerous doxygen warnings across Asterisk. It also updates
the makefile to regenerate the doxygen configuration on the local system
before running doxygen to help prevent warnings/errors on the local system.
Much thanks to Andrew for tackling one of the Asterisk janitor projects!
(issue ASTERISK-20259)
Reported by: Andrew Latham
Patches:
doxygen_partial.diff uploaded by Andrew Latham (license 5985)
make_progdocs.diff uploaded by Andrew Latham (license 5985)
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Review: https://reviewboard.asterisk.org/r/1970/
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Another very inappropriate placement of a ')' (again introduced in r362151)
caused the various truncate operations to attempt to truncate the sound file
at a position of '0'.
(issue ASTERISK-19655)
Reported by: Matt Jordan
(issue ASTERISK-19810)
Reported by: colbec
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Patch by: junky
Review: https://reviewboard.asterisk.org/r/1743/
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A very inappropriate placement of a ')' (introduced in r362151) caused the
maximum size of a file to be set as the result of a comparison operation, as
opposed to the result of the ftello operation. This resulted in seeking being
restricted to the beginning of the file, or 1 byte into the file. Thanks to
the Asterisk Test Suite for properly freaking out about this on at least one
test.
(issue ASTERISK-19655)
Reported by: Matt Jordan
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For the formats that support seek and/or truncate operations, many of
the C library calls used to determine or set the current position indicator
in the file stream were not being checked. In some situations, if an error
occurred, a negative value would be returned from the library call. This
could then be interpreted inappropriately as positional data.
This patch checks the return values from these library calls before
using them in subsequent operations.
(issue ASTERISK-19655)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1863/
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The principle difference between libvorbisfile v1.1.2 and newer (at least
v1.2.0) is the addition of the predefined callbacks OV_CALLBACKS_xxx in
vorbis/vorbisfile.h used for ov_open_callbacks().
* Updated the configure script to detect if libvorbisfile.h declares
OV_CALLBACKS_NOCLOSE.
* Copied the declaration of OV_CALLBACKS_NOCLOSE from v1.2.0 to allow
v1.1.2 to compile.
(closes issue ASTERISK-19370)
Reported by: Jonn Taylor
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Ogg/vorbis was fairly useless as a voicemail file format because it did
not implement the seek and tell format callbacks among other problems.
Since we were already using the libvorbis and libvorbisenc libraries we
can use libvorbisfile as it is also part of the vorbis library package.
* Made use the libvorbisfile to handle the ogg/vorbis file stream. The
format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.
(closes issue ASTERISK-16926)
Reported by: sque
Patches:
ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded by sque
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Patch by: Clod Patry
Review: https://reviewboard.asterisk.org/r/1651
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ASTERISK-18739
Patch by: pawel (modified)
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r330217 | seanbright | 2011-07-29 13:19:42 -0400 (Fri, 29 Jul 2011) | 9 lines
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r330213 | seanbright | 2011-07-29 13:18:56 -0400 (Fri, 29 Jul 2011) | 2 lines
Correct the check for O_RDONLY.
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r330204 | seanbright | 2011-07-29 12:58:40 -0400 (Fri, 29 Jul 2011) | 9 lines
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r330203 | seanbright | 2011-07-29 12:58:08 -0400 (Fri, 29 Jul 2011) | 2 lines
Only write to wav files that were opened to be written to.
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r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
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r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
Introduce <support_level> tags in MODULEINFO.
This change introduces MODULEINFO into many modules in Asterisk in order to show
the community support level for those modules. This is used by changes committed
to menuselect by Russell Bryant recently (r917 in menuselect). More information about
the support level types and what they mean is available on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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It was inconsistent to have the silk and celt format attribute
modules in the format file interpreter folder.
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through patch.
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r319083 | dvossel | 2011-05-16 09:26:33 -0500 (Mon, 16 May 2011) | 5 lines
Fixes Big Endian build issue.
(closes issue #19298)
Reported by: tzafrir
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r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines
Fix a bunch of compiler warnings generated by gcc 4.6.0.
Most of these are -Wunused-but-set-variable, but there were a few others
mixed in here, as well.
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r315259 | russell | 2011-04-25 14:37:32 -0500 (Mon, 25 Apr 2011) | 24 lines
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r315258 | russell | 2011-04-25 14:31:44 -0500 (Mon, 25 Apr 2011) | 17 lines
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r315257 | russell | 2011-04-25 14:28:41 -0500 (Mon, 25 Apr 2011) | 10 lines
Be more flexible with unknown chunks in wav files.
This patch makes format_wav ignore unknown chunks instead of erroring
out on them.
(closes issue #18306)
Reported by: jhirsch
Patches:
wav_skip_unknown_blocks.diff uploaded by jhirsch (license 1156)
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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/
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This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal. For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs. Functionally
no change in behavior should be present in this patch. Thanks to twilson
and russell for all the time they spent reviewing these changes.
Review: https://reviewboard.asterisk.org/r/1083/
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r284701 | qwell | 2010-09-02 11:43:09 -0500 (Thu, 02 Sep 2010) | 8 lines
Add slin16 support for format_wav (new wav16 file extension)
(closes issue #15029)
Reported by: andrew
Patches:
wav16.patch uploaded by andrew (license 240)
Tested by: qwell, andrew
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r279472 | tilghman | 2010-07-25 22:27:06 -0500 (Sun, 25 Jul 2010) | 2 lines
Formats need to load before apps, because some apps call ast_format_str_reduce() at load time.
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(closes issue #16293)
Reported by: malcolmd
Patches:
g719.passthrough.patch.7 uploaded by malcolmd (license 924)
format_g719.c uploaded by malcolmd (license 924)
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Updated the doxygen \arg line after looking at the file for some other Asterisk documentation
and noticing they weren't up to date. Thanks to seanbright for looking at the code for me :)
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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
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Reviewboard: https://reviewboard.asterisk.org/r/416/
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The 'pglobal' tool is quite handy indeed :-)
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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/
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r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr 2009) | 8 lines
Fix a few typos of the word "frequency."
(closes issue #14842)
Reported by: jvandal
Patches:
frequency-typo.diff uploaded by jvandal (license 413)
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r175825 | oej | 2009-02-15 21:33:17 +0100 (Sön, 15 Feb 2009) | 2 lines
format_ilbc does not depend on codec libraries and can therefore always be made. My mistake. Ursäkta!
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r175792 | oej | 2009-02-15 21:20:21 +0100 (Sön, 15 Feb 2009) | 2 lines
Disable format_ilbc.so by default, like codec_ilbc.so
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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/
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for ulaw.
(closes issue #14001)
Reported by: henrikw
Patches:
alw.diff uploaded by henrikw (license 627)
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branch, and add the ones needed for all the new code here too
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(Closes issue #13657)
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r143903 | seanbright | 2008-09-22 18:49:00 -0400 (Mon, 22 Sep 2008) | 8 lines
Use the advertised header size in .au files instead of just assuming they
are 24 bytes (the minimum).
(closes issue #13450)
Reported by: jamessan
Patches:
pcm-header.diff uploaded by jamessan (license 246)
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utils/
codecs/
and a change I missed from formats/
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this in pieces so the diffs are a little bit smaller and more reviewable.
pbx/ and formats/ first.
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(closes issue #13002)
Reported by: caio1982
Patches:
janitor_arraylen5.diff uploaded by caio1982 (license 22)
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- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
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recording.
(closes issue #11962)
Reported by: garlew
Patches:
recording.patch uploaded by garlew (license 376)
bug-11962.diff uploaded by snuffy (license 35)
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r111658 | qwell | 2008-03-28 11:19:56 -0500 (Fri, 28 Mar 2008) | 8 lines
The file size of WAV49 does not need to be an even number.
(closes issue #12128)
Reported by: mdu113
Patches:
12128-noevenlength.diff uploaded by qwell (license 4)
Tested by: qwell, mdu113
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Fix a number of other places where the number of samples in a G722 frame was
not properly handled because of various reasons.
main/rtp.c:
- When a G722 frame is read from the smoother, the number of samples in the
frame must be divided by 2 before being sent out over the network. Even
though G722 is 16 kHz, an error in some previous spec has made it so that
we have to list the number of samples such as if it was 8 kHz.
main/file.c:
- When scheduling the next time to expect a frame, take into account that the
format of the file we're reading from may not be 8 kHz.
codecs/codec_g722.c:
- When converting from G722 to slinear, g722_decode() expects its samples
parameter to be in the silly (real samples / 2) format. Make it so.
- When converting from slinear to G722, properly set the number of samples in
the frame to be the number of bytes of output * 2.
formats/format_pcm.c:
- This format module handles G722, among a number of other formats. However,
the read() and seek() functions did not account for the fact that G722 has
2 samples per byte.
(closes issue #12130, reported by rickross, patched by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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