The wiki page QMX QMX_plus Startup has been updated by @Ludwig_DH8WN <DH8WN@...>.

Compare Revisions

On 20/04/2025 18:54, Wiggle Pig via groups.io wrote:

Given that the usual failure in the finals is through overheating, as a number of examples have shown,

A number of examples have shown BS170s blown apart.
But is that due to overheating in service or a result of a high voltage causing one to short THEN blow?

73 Alan G4ZFQ

开云体育

Hi,

After the installation of the latest Beta, the LCD shows only processing. Is this normal?

Few minutes later i un/replugged the powersupply and it works as expected, ok, a little bit better than expected.

A big Thanks to Hans for this awesome peace of hamgadget.

73, Jens

Hello Hans,

I would like to respond to your concerns one at a time

On Fri, 2025-04-11 at 18:00:46 +0300, Hans Summers via groups.io wrote:

I don't think your proposal would work well, though I find myself
lacking a deep enough understanding to explain why clearly. Perhaps
someone else could contribute here also. But I'll try. ... Weaver
demodulation is used in some SDRs. A disadvantage is that there is a
hole in the response at the sub-carrier frequency (the CW sidetone
frequency, in your nomenclature). In the analog world it is hard to
make the hole small but in DSP it could be made a lot smaller, such
that it is unobtrusive on an SSB receiver. However for CW, I wonder
how that would work... the "hole" would be exactly at the center of
the passband, and the CW transmission could well have energy either
side of it, and given that the actual bandwidth of a CW transmission
could be only 10 or 20 Hz, even a small "hole" would have a marked
effect.

There is no "hole" in the Weaver modulation if we make it digital. All
the information of the received signal is present in the I and Q
channels.

Suppose we want to listen to a CW signal transmitted at 7020 kHz. We set
the VFO to 7020-12 = 7008 kHz. The mixed signals contain the sum and
difference of these frequencies: 14028 and 12 kHz. Only the 12 kHz
components pass through the lowpass filters in front of the ADC, and
appear digitally as cosine and sine of a 12 kHz signal.

We are now in the digital world. We mix the 12 kHz IQ signals with the
-12 kHz IF frequency. The mixer outputs two DC components, which are the
cosine and sine of the phase difference between the mixed signals.

These DC components pass almost unaffectected through the lowpass
filters, say ±50 Hz. The outputs are then mixed with the sidetone
frequency (700 Hz) before being fed to the speaker's DAC. We will here a
700 Hz sine tone.

Suppose we want to move the passband 20 Hz up. Then we adjust the IF
mixer to 12020 Hz and the sidetone frequency to 720 Hz. The received
signal will now appear as -20 Hz signals in the LP filter (the frequency
needs a sign in the IQ domain). We will still hear a 700 Hz tone in the
speaker, while the passband will be from -20 to +80 Hz around 7020 kHz.
We shouldn't adjust the VFO because that would change the USB output.

An additional problem is that you lose the advantage of doing the SDR
at a 12 kHz IF. The 12 kHz IF gives immunity to hum and noise which
can often be found on a direct conversion receiver near to (or within
a few kHz of) 0 Hz. Operating at a 12 kHz IF produces significant
advantages. The conversion to 12 kHz IF is quite trivial digitally and
computationally inexpensive because the sampling is at 4x this (48
ksps).

I still use the 12 kHz IF, as described above. The bias, low-frequency
noise, and hum are avoided.

Finally another problem. To obtain a 50 Hz bandwidth CW filter, you
would need to have a 25 Hz LPF. To do such a low pass filter in DSP
will require a very long (large number of taps) DSP filter. To avoid
this, it will be necessary to decimate down to a much lower sample
rate. Which is fine. But you cannot avoid the fact that there will be
a large delay through the filter. Intuitively and/or approximately
speaking the filter must be 40 ms long for a 25 Hz LPF.

There is essentially no (extra) delay caused by the implementation of
the filters. They run as a series of first or second order filters, not
as finite-impulse filters. The realizations of the filters are on
so-called balanced form, which makes them numerically robust. It is
possible to run them using 16-bit arithmetics. (I have been running it
on a 48MHz SAMD21 M0+ at 16-kHz.). The response of the filter will have
a delay, but this is determined by the bandwidth and type of filter.

The proposed filter works both for CW and SSB (Rx and Tx), without using
Hilbert transforms.


Best regards

Anders

I post a message like this every 6months or so in hope...

I'm looking for someone who has an original QCX Classic case they no longer want. This is the BaMa Tech case that was sold by QRP Labs for sometime. The case is not available new any more.

If you have such a case or maybe a cased QCX Classic that is fried and dead and you want to sell, please contact me andy.mm0fmf AT gmail.com.

I'm in the UK and Brexit etc. has made shipping to and from the UK much more "fun and challenging" now. But if you are in the EU or in the US I have friends/relatives in either that will handle shipping to me making it simple for you :-)

Here's hoping.

Thanks
Andy, MM0FMF