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[linrad] Re: Abit IC7 allows 192kHz settings for CPU-load tests; Re: [linrad] CPU usage

> I have now understood that the CPU is not determining
> the delay in 'Linrad', but rather: the slower the CPU,
> the higher its relative usage, until it reaches 100%..
> Then the application probably "collapses"... a rare
> occurrence with today's fast processors.
No, it has a graceful degradation in that screen updates
deteriorate first:-)

> All of the integrated audio on several motherboards
> "propose" a sample rate choice up to 48000, except the
> Abit IC7 (proposing choices up to 192000). It does not
> mean that one can build an I/Q-receiver with successful
> utilization of the full bandwidth. Actually the IC7 is
> the worst quality board (compared to all others here)
> regarding the operation of the integrated audio circuit
> (at least as far as the microphone input is concerned).
The microphone input is always useless because of the high
noise level.............
Why do you not think one could build a 192 kHz I/Q receiver?

> For example under Skype VoIP-communications there is a
> strong tapping noise on the outgoing audio, and under
> 'Linrad' the input spectrum shows wide noise bands at
> 7 kHz and 41 kHz on the frequency spectrum when using
> 48000 sampling rate (when no input signals are fed).
Oooh! When there is no input at all, some A/D converters
show nasty things. You have to supply a noise floor from
your tower-mounted preamplifier (or simulate) to get an
idea about the real performance and it is always surprisingly
good in my experience:-)

I am pretty sure I wrote about this somewhere but I failed
to locate the page. The ugly-looking noise-bands disappear
if you send a signal above the highest frequency into the
soundcard for example. Sampling at 192 kHz you might send
150kHz into the card for example. That would remove the
noise bands without any adverse effects except for a small
loss in the largest tolerable signal level - the peak
amplitude has to stay within the the available bits for
the sampling at perhaps 10 MHz.
Adding anything is silly of course since preamplifier noise
will be sufficient to remove the noise bands:-)

> All of this may be better on the line input (however
> not tested). Also another main power supply should be
> tried to ensure that all MB-voltages have been clean.
> When feeding audio to the IC7 input, it was clearly seen
> that the circuitry was not designed for these high rates
> as the input spectrum showed multiple aliased responses,
> and the baseline level dropped off above 24 kHz separation
> form the center frequency. Also I noticed that the 'Linrad'
> frequency scale of the waterfall behaves like an old car's
> mileage meter, showing 100 000 as 00 000 (actually just 0).
> So the parameter needs an allocation for one more figure
> on the right end of the scale: e.g. 140000 Hz now 40000 Hz.
> Initially this was a little confusing, but the experiment
> yielded this improvement suggestion for the user-interface.
I am sorry to learn that the IC7 is properly designed for audio.
Sampling is done at perhaps 10 MHz but the digital filter
inside the chip limits the bandwidth to about 24 kHz even when
you ask to get the output bit stream at 192 kHz. This means that
some extra silicon is used to supply different anti-aliasing
filters for different output rates - a luxury that Delta44
and Lynx Two do not have:-)

The scale shows only tens of kHz. When the frequency you set
for the WSE unit is 144125000 Hz, you will see only the
last 5 digits. The full frequency information is shown in the
baseband graph.

> As such all these experiments have again derailed me from
> the goal to get some form of real world reception ability.
> Thus it is about time to link up some of the analog stuff
> in front of the sound inputs. That is actually the best
> incentive to learn about the use and optimization of the
> various parameters. The impulse generator for the quick
> calibration of the I/Q-channels (and the smart blanker)
> is actually not too complicated, but here its construction
> has been delayed already for years, due to all the other
> "interesting" 'Linrad' experiments and software challenges.
With the WSE converters you could calibrate with almost any
pulse generator. Feed it through a 6 dB pad into the RX2500
unit:-) Calibration will be slightly incorrect if the
signal source deviates much from 50 ohms.

As an alternative you might feed the pulse generator into
the RX10700 input.

> A key for quick successful results would be an "approach
> guidebook" that should be fairly strictly followed. The
> ongoing Live_CD development by Roger/W3SZ will certainly
> speed up the start-up time of new candidates (by 10 dB).
Or 15 dB;-)



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