Loading ...
Sorry, an error occurred while loading the content.

Re: Fw: Perseus remote Sound from Moscow rz3dvp missed ?

Expand Messages
  • iv3nwv
    ... Hi Paul, there must be a bug somewhere for sure. Interestingly the server continues to pass audio data packet to the client but they simply contain
    Message 1 of 48 , Jul 7, 2013
      --- In perseus_SDR@yahoogroups.com, pd0psb@... wrote:

      > I experienced the same behaviour with my server on some occasions.
      > 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.

      Hi Paul,
      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
    • Leif Asbrink
      Hello Nico, ... I am afraid you apply a conventional model which is not applicable in the QRN-fighting context. Consider a sampling rate of 4 MHz. Apply a
      Message 48 of 48 , Jul 18, 2013
        Hello Nico,

        > If one computes the number of taps of a FIR decimation
        > 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)
        > where:
        > 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.

        I am afraid you apply a "conventional" model which is
        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
        sources better.

        > In Perseus the decimation filter has been designed so that
        > 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 if
        > the output sampling frequency were 40 MS/s. It will
        > always be 20 samples long.
        In Linrad, the PC software will take the fourier transform of the
        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,
        > 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 output
        > 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...
        Yes:-) This is what I advocate. 4 MHz sampling and
        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)
        > 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?
        As far as I undersdtand it is impossible to clean up a 200 kHz
        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 but
        > 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).
        Five years later it is very likely that a factor of two is OK:-)


      Your message has been successfully submitted and would be delivered to recipients shortly.