In radio amateur circles where exchanges of information are brief why make a
simple job difficult. Just use CW normal speed or QRSS if required.
g3kev
----- Original Message -----
From: "Andy Talbot" <[email protected]>
To: <[email protected]>
Sent: Sunday, May 06, 2012 4:14 PM
Subject: Re: LF: Generating 8970 Hz carrier with Spectrum Lab ?
The problem with USB slippage comes about because of the Isochronos
mode (yes, that is spelt correctly, it is not IsoSYNchronous) -
where it sends puts real time rapid data transfer as more important
than lost samples and sends samples in real time without checking
This is probably just what music and audio people want, but it screws
up using USB based sound cards as precision timed A/D converters.
USB does have other modes with full error checking, albeit with some
latency, but the latency wouldn't matter here.
While it would be a pretty simple task these days to build an external
A/D converter and interface to USB properly, the resulting data stream
would need a custom driver written, then all the decoding software we
know and love would have to be modified to work with this custom
interface. Then someone else would come along with another design
and we'd have to start again.
I suspect it is going to be absolutely impossible to use the soundcard
as a guaranteed glitch free A/D converter.
Joe Taylor K1JT in the WSJT suite has a dead simple pragmatic
solution. Instead of trying to read the data in real time, all the
WSJT modes read the entire transmisisons' worth of data to a .WAV
file, and the software works on that. While WSJT uses the soundcard
to do the job it will suffer from the same slippages (which don't
matter' of course' for those modes) the idea of an intermediate buffer
file would help.
A custom A/D could send its data in its own format via USB, or serial
COM port or whatever, to software that saves blocks in the format of a
.WAV file. Then the decoding software works on the resulting .WAV
files. It won't be real time any more, but none of these slow data
modes actually are that real time. The speed of reading and switching
(using a pair of ping-pong files if necessary) can make the whole
system pseudo real time - in the same way as WSJT appears to run
continuously.
Now, any A/D design can be used provided the results are written to a
.WAV file. Wav files can have any sample rate (so long as it is an
integer number of Hz) and do not have to be restricted to 48000, 11025
or whatever, so custom LF receivers using quite slow A/D converters
and low sample rates are now valid.
Just throwing that idea into the ring..
Andy
www.g4jnt.com
On 6 May 2012 16:14, James Moritz <[email protected]> wrote:
> Dear Andy, LF Group,
>
> A bit late, but never mind...
>
>> Has anyone tried using an external USB soundcard with a separate
>> locked clock? Most work from a 12MHz crystal which can be replaces
>> with a GPS locked source without too much effort. But I can't help
>> wondering if there will be subsequent USB synchronisation glitches
>> upsetting the input sampling.
>
>
>
> I can confirm that glitches do occur with USB sound cards. I have found
this
> to be a perennial problem trying to use such a sound card with the laptops
I
> have available. For 9kHz reception, the relatively rapid temperature
> fluctuations inside the laptop, and resulting cyclic drift of the internal
> soundcard sampling frequency interfere with the operation of DL4YHF's
> ingenious sample rate correction facility in SpecLab, making the internal
> sound card unusable for FFT resolution below a few millihertz. I found my
> USB soundcard solved this particular problem quite well, but introduced
> glitches that made achieving FFT resolution in the uHz range impractical.
>
> Watching Speclab's sample rate correction "history" window, the USB card
> sample rate typically starts off a few hundred ppm low (much larger than
the
> actual clock frequency error), but remaining stable to within a few ppm,
but
> then at unpredictable intervals abrupt jumps in sample rate of a similar
> order of magnitude occur, with corresponding "blips" on the spectrogram
> traces. The reported sample rate is always lower than the nominal value,
> suggesting that some samples are being periodically discarded somehow.
>
> The sound card uses a single-chip integrated audio codec and USB
> transceiver, using a single 12MHz crystal. I can't really believe in "USB
> slippage" in the hardware - surely losing some of the data would either be
> handled quietly by the USB error checking, or result in endless error
> messages. The same sound card seems to work in a glitch-free way when
> plugged into my desktop machine, where the reported sample rate error is
in
> line with the error in the crystal frequency.
>
> Cheers, Jim Moritz
> 73 de M0BMU
>
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