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<DIV dir=ltr align=left><SPAN class=921495717-18062009><FONT face=Arial
color=#0000ff size=2>Thanks Brian, </FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=921495717-18062009><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=921495717-18062009><FONT face=Arial
color=#0000ff size=2>So, just to calrify will the base call always be 8kHz?
</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=921495717-18062009><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=921495717-18062009><FONT face=Arial
color=#0000ff size=2>On a related note, do you happen to know the bitrate of
each open channel/live call? Is it 16 kilobits per second like the recorded
audio? I need to do some calculations on the badwidth required to handle a
certain number of concurrent calls.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=921495717-18062009><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=921495717-18062009><FONT face=Arial
color=#0000ff size=2>Many thanks</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=921495717-18062009><FONT face=Arial
color=#0000ff size=2>Andy</FONT></SPAN></DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
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<FONT face=Tahoma size=2><B>From:</B>
freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] <B>On Behalf Of </B>Brian
West<BR><B>Sent:</B> 18 June 2009 18:11<BR><B>To:</B>
freeswitch-users@lists.freeswitch.org<BR><B>Subject:</B> Re: [Freeswitch-users]
Sample rate and recordFile<BR></FONT><BR></DIV>
<DIV></DIV><BR>
<DIV>
<DIV>On Jun 18, 2009, at 11:54 AM, Andy wrote:</DIV><BR
class=Apple-interchange-newline>
<BLOCKQUOTE type="cite"><SPAN class=Apple-style-span
style="WORD-SPACING: 0px; FONT: 12px Helvetica; TEXT-TRANSFORM: none; COLOR: rgb(0,0,0); TEXT-INDENT: 0px; WHITE-SPACE: normal; LETTER-SPACING: normal; BORDER-COLLAPSE: separate; orphans: 2; widows: 2; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0">
<DIV dir=ltr align=left><SPAN class=359422916-18062009><FONT face=Arial
color=#0000ff size=2>1) I notice that when I change the sample rate it
automatically changes the bit rate too. I understand why this is the case but
wondered if it was just as easy to be able to control the bitrate as well as
the sample rate.</FONT></SPAN></DIV></SPAN></BLOCKQUOTE>
<DIV><BR></DIV>
<DIV>If you're talking about mod_shout, NO. You'll end up picking an
invalid bitrate and asking why it doesn't work... been there done that... I
changed it a few months back to pick the optimal bitrate for the sample
rate.</DIV><BR>
<BLOCKQUOTE type="cite"><SPAN class=Apple-style-span
style="WORD-SPACING: 0px; FONT: 12px Helvetica; TEXT-TRANSFORM: none; COLOR: rgb(0,0,0); TEXT-INDENT: 0px; WHITE-SPACE: normal; LETTER-SPACING: normal; BORDER-COLLAPSE: separate; orphans: 2; widows: 2; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0">
<DIV dir=ltr align=left><SPAN class=359422916-18062009><FONT face=Arial
color=#0000ff size=2>2) When I use a sample rate other than 8000 I get a
warning 'Sample rate doesn't match'. I guess this puts some extra load on the
server. If all my calls are being recorded and all at 11025 can/should I alter
the sample rate of the base call to
11025?</FONT></SPAN></DIV></SPAN></BLOCKQUOTE></DIV><BR>
<DIV>NO. Your phone call is running at 8kHz, Your sound file is 11025 and
they don't match, If you were to play this file into an 8k channel without a
resample it would sound a little like satan. or a dragging tape deck.
The file has to be resampled to match the current session
rate. </DIV>
<DIV><BR></DIV>
<DIV>/b</DIV>
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