I thought I would add a bit to the equation. I don't operate many SSB
contests (mainly CW) but I have pretty considerable experience with the N1MM
audio interface. I wrote the Flex KB article (needs updating) and was one of
the beta testers for the new N1MM audio interface.

The new N1MM audio interface allows two SEPARATE sound sources to be chosen.
I have it set up with my computer soundcard for microphone input and VAC for
output. All sample rates and delays are left at their PowerSDR/N1MM
defaults. This feature is new to N1MM in the past month or so. Therefore
when I want to record a voice macro I use the N1MM editor with CTRL-SHIFT-Fx
where Fx is the function key to record and CTRL-SHIFT-Fx to stop recording.
Then all sounds play back perfectly within N1MM without delay or truncation.
I could never get things to work correct when using Windows Audio or
Audacity to record the sounds.

Give this a try and see if it works for you.

I'll try to get the KB article updated in the next couple of weeks.

73,

Mack
W4AX
-------------------------------
On Tue, Jul 5, 2011 at 6:30 PM, Jim Jannuzzo <jsqu...@msn.com> wrote:

>
> I thought so- I believe you having the same sample rate up and down
> conversion issue that I was having on digital modes, and that 44.1 k is not
> where you want to be.  See if you can find out what sample rate N1MM uses
> for its audio files.  Record the wavs at a multiple of that sample rate.  I
> "forced" the VAC sample rates and the Windows sample rate to a multiple of
> the sample rate used in one of my digi programs.  Since many Flexers use
> N1MM, they can contribute the sample rates for their wav files and for VAC.
>  > Date: Tue, 5 Jul 2011 14:45:00 -0500
> > From: kd0...@gmail.com
> > To: flexradio@flex-radio.biz
> > Subject: [Flexradio] Fwd: Voice keyer through PSDR truncates end of audio
> on output, progressive worsening
> >
> > To: t...@flexradio.com
> >
> >
> > Thanks for the pointer.
> >
> > Since I was not sure which rate you were pointing to as being the
> > preferred, I tried both. I found that setting everything to 48khz
> > caused the voice to sound like I was talking through a fan on low
> > (buffering issue?), but setting the VAC lines in windows as you
> > suggest to single rather than double channel with all settings all
> > around at 44.1 seems to have reduced the speed with which it begins to
> > clip off the end, but it is still happening. I'd estimate the need to
> > reset every 5 minutes or so to avoid becoming unintelligible.
> >
> > The wavs I am using with these results were recorded at 48khz.
> >
> >
> >
> > On Tue, Jul 5, 2011 at 12:59 PM, Tim Ellison, W4TME <t...@flex-radio.com>
> wrote:
> > > If using Vista or Win7, make sure the default format sampling rate for
> the
> > > Windows VAC sound devices (playback and record)  matches what you use
> for
> > > the VAC.  The default is 44.1 KHz which forces Windows to do an
> > > up-conversion to 48KHz
> >
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-- 
73,

Mack de W4AX
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