Re: Fw: Perseus remote Sound from Moscow rz3dvp missed ?
- --- In perseus_SDR@yahoogroups.com, pd0psb@... wrote:
> I experienced the same behaviour with my server on some occasions.Hi Paul,
> It seemed like the digital audio stream was in constant full saturation preventing the RX audio to pass while the server itself was running correctly.
there must be a bug somewhere for sure.
Interestingly the server continues to pass audio data packet to the client but they simply contain meaningless data (btw, the S-Meter indication is passed along with such packets, that's why it goes up to a meaningless value it too).
Using an old server versions does not help just because the bug comes from them and has not been added in the latest one (server version v11a is just a recompile of the v10c in which the USB data transfer size has been modified to supports the USB3.0, no other mods in it).
Fake servers and other oddities occur simply because both the server and the client do some minimal tests to see who they are speaking with (and sometimes they fail to discover, hi). Server version X is meant to work properly only with client version up to Y and viceversa.
Sooner or later I'll list on the Peresus directory only the latest version servers as I do not mean to support what has been already superseeded. Different versions create confusion and, as already demonstrated, they do not serve to solve any problem, those who were believed being added in the last server version included.
> Please forgive me but I was so naughty to behave very badly on your server so you can take a look in the crash log when it tips over.Thank you. As in these gorgeus, sunny Sundays I've usually nothing to do and I get rapidly annoyed tanning im my garden, I'm going in my office right now to examine the crash you have kindly created into our server to help all of us :-D
> May be it would be possible to create an error-detect-restart routine for this if 100% crazy-user-proofing is wanted.Uhmmm... There's a more simple solution:
I'll blacklist your IP :-D
Nico / IV3NWV
- Hello Nico,
> If one computes the number of taps of a FIR decimationI am afraid you apply a "conventional" model which is
> filter with a decent performance (say 0.1 dB in-band ripple
> and 100 dB alias image rejection) he discover a simple
> rule of thumb:
> N =(about) 4*D/(1-B/Fco)
> N is the required decimation filter number of taps
> D is the decimation factor
> B/Fco is ratio between the desired output alias free bandwidth and the output sampling frequency.
> Since after filtering the decimator takes one output every D
> input samples, the output impulse response is no more
> than N/D samples long, that's to say:
> N/D =(about) 4/(1-B/Fco)
> Note that the length of the output impulse response
> *does not* depend on the output sampling frequency, but just on the B/Fco ratio.
> If such a ratio is high the output pulse can be quite long.
not applicable in the QRN-fighting context.
Consider a sampling rate of 4 MHz.
Apply a FIR filter that has say 0.1 dB in-band ripple
and a -1 dB point at say 0.8 MHz. The -20 dB point should
be at 2 MHz and the -100 dB point at 3.2 MHz. The alias-free
range (-100 dB) would be +/- 0.8 MHz but a clever DSP software
could compensate for the fall-off between say 0.8 and 1.6 MHz
to provide a perfectly flat passband of 3.2 MHz or so. The alias
suppression at the corner frequencies would be poor. Maybe 20 dB,
but I do not think that would impair the noise-fighting.
The useful bandwidth for receiving would be 1.6 MHz only and
not any improvement over the 2 MHz sampling. The purpose of the
faster sampling would only be to eliminate certain interference
> In Perseus the decimation filter has been designed so thatYes.
> the alias-free bandwidth is 80% the output sampling frequency
> (1.6 MHz when the sampling rate is 2 MS/s) which is a good
> compromise between the decimation filters complexity and
> the efficiency of the digital signal processing made on the PC.
> At such a B/Fco ratio you can expect that each output pulse
> due to an istantaneous glitch at the receiver input is
> approximately 4/(1-0.8) = 20 samples long whatever the
> output sampling frequency is.
> You can't really resolve it into a single pulse even ifIn Linrad, the PC software will take the fourier transform of the
> the output sampling frequency were 40 MS/s. It will
> always be 20 samples long.
input data stream, divide it by the fourier transform of the
impulse response of the hardware and multiply it by a "desired
pulse response" This way the pulse length is made shorter than 20
samples and at the same time the ~0.1 dB ripple is removed.
The length of the pulse is determined by the "desired pulse response"
which depends on the skirt steepness that the user has decided.
The smart blanker knows the exact shape of the pulse and its length
so it does not matter that the pulse is long in terms of samples.
I am aware that very few operators use Linrad and that only
a very small fraction of the users care to calibrate their
systems properly. I have tried to explain the theory, but I
do not think I have been sucessful at all. I am interested
in static rain at high bandwidth because I have a feeling
recordings would show a dramatic difference between the
Linrad blanker and other blankers.
> Of course 20 samples at 40 MS/s are a 0.5us interval,Yes:-)
> which is a much shorter time interval than that obtained
> if the sample rate were 2 MS/s but instead of increasing
> the output sample rate one can obtain the same result
> simply relaxing the B/Fco requirement.
> If the B/Fco ratio were 60% instead of 80% the outputYes:-) This is what I advocate. 4 MHz sampling and
> pulse lenght would be the half the original, if it were
> 40% one third and if it were 20% one fourth of it, a
> mere 5 samples interval (2.5us @ 2MS/s), which is even
> the half of what one could obtain attempting to double
> the output sampling frequency (and mantaining the
> original 80% B/Fco ratio).
> The penalty is that the the alias free bandwidth
> is much less than the output sample rate...
40% alias-free bandwidth. I also want the -10 dB point
to be fairly high, maybe 80% of Nyquist.
> but who cares if we would just be satisfied to (carefully)As far as I undersdtand it is impossible to clean up a 200 kHz
> clean-up a not-so-wide 200 kHz bandwidth out of a 2 MS/s
> IQ stream?
> And if it works, wouldn't it be better than obtaining the
> same result using 4 MS/s maybe overloading a poor man CPU?
wide segment of a 2MS/s IQ stream if the (random) secondary
pulses can not be resolved. From old experience as well as from
the one and only wideband recording at my disposal a bandwidth
of 1.6 MHz is marginal. It may or it may not work.
> BTW, making a new 4MS/s DDC would not be impossible butFive years later it is very likely that a factor of two is OK:-)
> as I haven't implemented it yet I can't say that what
> was initially conceived for a much smaller output sample
> rate could sustain it (in 2008 I was even not sure that
> the 2 MS/s rate could really work).