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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I am also interested in capturing the live
audio stream from a call.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>The best place I’ve found to start
is to modify the mixmonitor application. This lets you easily grab frames
from a call. Also, you can grab both sides of the call muxed together or separately.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Asterisk has support for audio conversion
between several formats (check frame.h for supported formats). There are
two ways you could take advantage of this:<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:21.0pt;text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'><span style='mso-list:Ignore'>-<font size=1 face="Times New Roman"><span
style='font:7.0pt "Times New Roman"'>
</span></font></span></span></font><![endif]><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'>Request
a different format when setting up the channel spy (spy.read_queue.format).
This only works if you’re not bridging the two sides of the call.<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:21.0pt;text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'><span style='mso-list:Ignore'>-<font size=1 face="Times New Roman"><span
style='font:7.0pt "Times New Roman"'>
</span></font></span></span></font><![endif]><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'>Do your
own translation using the Asterisk translation stuff
(ast_translator_built_path, ast_translate…)<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:3.0pt'><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:3.0pt'><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'>However,
I’m not sure the Asterisk audio conversion stuff is going to be of help
to you. By default the audio frames you’d get in mixmonitor would
be 16-bit signed linear 8KHz and I’m guessing little endian. I
think you would need to perform the endianness conversion yourself.<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:3.0pt'><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:3.0pt'><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'>Hope
this helps<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> asterisk-dev-bounces@lists.digium.com
[mailto:asterisk-dev-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Jamel A<br>
<b><span style='font-weight:bold'>Sent:</span></b> Tuesday, July 04, 2006 5:44
AM<br>
<b><span style='font-weight:bold'>To:</span></b> asterisk-dev@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> [asterisk-dev] capture
and convert voice stream to a certain format</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Hello,<br>
<br>
I have a program that analyse (in live mode) voice captured by the microphone
(8KHz, 16 bit, mono, signed, big endian).<br>
I would like to analyse voice through Asterisk.<br>
To test, I've modified the Echo function (located in the apps directory) to
grab voice frame, but I thing that firstly I must convert the data right?<br>
<br>
How can I do that?<br>
<br>
Maybe, is there another solution? I know that we can record but I suppose that
we cannot read the file, wile being written...<o:p></o:p></span></font></p>
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