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

Re: Quad 2805

Expand Messages
  • tonycdk
    Laurie ... I made the measurements so long ago that I can not be absolutely certain of the configuration details. The measurement at the higher of the two
    Message 1 of 40 , Feb 20, 2013
    • 0 Attachment
      Laurie

      > Tony, were those observations taken with Acourate switched in or without.

      I made the measurements so long ago that I can not be absolutely certain of the
      configuration details. The measurement at the higher of the two frequencies is
      above my normal crossover frequency. The signal will have been processed
      through my convolution computer so I probably had the Acourate
      convolution filters set to flat over the whole of the frequency range. There is
      the possibility that I had somewhat shaped the amplifier response, to
      approximately linearize the subwoofers, but I can not say for sure. I have
      checked my archives and do not appear to have retained the measurement
      results.

      > and so wouldn't be able to provide a compensatory boost just at the start of a
      > signal, should it even be thought to be desirable

      I don't think that any attempt to correct for this would be desirable.

      > From what I vaguely understand of digital EQ the equalisation in all the available
      > systems only provides steady state filters and boosts, albeit sophisticated ones.

      This is not the case with Acourate. You can window the impulse to eliminate the
      most important room sound if you want to. The program has a "frequency
      dependent windowing (FDW)" that allows this. The filters are computed for a time
      that is determined by the length of the window. You can also process as more or
      less "steady state" if that is what you wish. The general consensus is that one
      should correct to get the direct sound flat and that is when you would use the
      FDW capability.

      I think that the Holm software follows a similar approach because, when I looked at it, it seemed to be following Acourate - but with less capability.

      Tony



      --- In regsaudioforum@yahoogroups.com, "laurie483000" <laurie483000@...> wrote:
      >
      > Tony, were those observations taken with Acourate switched in or without. However, apart from the business of creating delays for integration of subs / drive units as part of the x-over facilities, I believe that neither Acourate nor any other DSP EQ systems allow for small timing changes and so wouldn't be able to provide a compensatory boost just at the start of a signal, should it even be thought to be desirable.
      >
      > From what I vaguely understand of digital EQ the equalisation in all the available systems only provides steady state filters and boosts, albeit sophisticated ones. I've read the relevant articles on REGONAUDIO a few times, but am still unsure about this, so stand to be corrected. (Some of my fuzzy comprehension comes from the Room Impulse graphs in the middle of the Sigtech review - it shows a room reflection being dealt with after a small time interval, but maybe this is a phasing matter.)
      >
      >
      > Laurie
      >
      >
      > --- In regsaudioforum@yahoogroups.com, "tonycdk" <tcdk@> wrote:
      > >
      > > Robert
      > >
      > > No Fourier transforms here. Look at the file 'subwoofer results (or is it measurements)' in my folder.
      > >
      > > Tony
      > >
      > > PS. When I first posted this material I seem to remember you telling me that you had observed the same type of thing.
      > >
      > >
      > >
      > > --- In regsaudioforum@yahoogroups.com, "Robert" <regtas43@> wrote:
      > > >
      > > > WHen you turn on a sine or cosine wave(from nothing)
      > > > the Fourier transform has got a bunch of stuff in it
      > > > in addition to the energy in the frequency of the
      > > > sine wave. So when you filter-split this signal and send the low pass
      > > > filtered part to the sub, then what happens
      > > > has some stuff in it that one might not expect.
      > > > Have a look at this
      > > > http://www.thefouriertransform.com/pairs/rightSidedSinusoids.php
      > > > and this
      > > > http://www.thefouriertransform.com/pairs/rightSidedSinusoids.php#sine.
      > > > REG
      > > >
      > > > PS Or is this what is being discussed? I missed the beginning
      > > > of this somehow...
      > > >
      > > > --- In regsaudioforum@yahoogroups.com, "tonycdk" <tcdk@> wrote:
      > > > >
      > > > > This effect showed that, starting from rest, my sub took a few cycles to stabilize at the input frequency. This is more an item of interest than a concern. The ear takes a few cycles before it identifies the frequency of a note and, in any case, even the first half cycle amplitude is down by less than 2dB.
      > > > >
      > > > > The measurement sets were taken at my listening position in my room and should not necessarily be taken as typical. I have tried a similar thing with a servo driven speaker with awful results - which may have been caused by the effects of the room (not mine).
      >
    • tonycdk
      Robert ... I have been pondering your question (which is one reason for the delay in answering - the other being an extensive search through my archives). I do
      Message 40 of 40 , Feb 22, 2013
      • 0 Attachment
        Robert

        >
        > It would be interesting to know whether
        > this is a nonlinearity or whether it is
        > an effect of the band limiting of the woofer.

        I have been pondering your question (which is one reason for the delay in
        answering - the other being an extensive search through my archives).

        I do have some uncorrected subwoofer measurements out to about 3 kHz
        but I don't think I have software that would do what you suggest.

        Just to provide some additional information I have posted in my photos
        folder some more plots from the measurement file for the subwoofer
        waterfall that I posted a few days ago. The plots are: Frequency
        response, group delays, and impulse response. The 'overshoot' on
        the impulse response is interesting, as it results from the low-frequency
        roll-off of my measurement system. The microphone rolls-off at about
        6 Hz and I do not know much about the actual response of the speaker
        below this frequency. If I ignore the response below 6 Hz (i.e. make the
        response amplitude equal to the 6 Hz value at frequencies below 6 Hz)
        the impulse cleans-up considerably. I have posted the comparisons in
        my photos folder as well.

        One of the interesting aspects of the measurements was to demonstrate
        how the speaker/enclosure centering forces quickly dominate the motion.
        This is very clear from the results for the offset drive case where the
        motion very quickly stabilizes about the zero displacement position.
        The restoring forces are probably quite large as the enclosure is closed
        with a very small volume.

        For the measurement starting at zero displacement, the motion starts
        from rest. This is where, in the full cycle, the speaker moving parts
        have a maximum velocity. Consequently one would expect the first
        peak to be a little low. I would think that this is a repeatable but
        'non-linear' effect. One can get a feel for this effect as a part of the
        overshoot at the end of the pulse train (drive ends with the speaker
        at maximum velocity) results from moving part momentum - and
        there is obviously some room effects in the decay.

        However, there is clearly something else affecting the motion as
        one would not necessarily expect the overshoot that follows on
        the negative displacement. There are also possibly the effects
        that you mention playing a part. If you can suggest what I can
        use to perform what you asked please let me know the details.

        >
        > But my experience with the Helsinkis made
        > me wonder if my faith in the insensitivity
        > of the ear to timing effects in the bass
        > might not be a little greater than was
        > justified! Still this looks really good to me.
        >

        What is the order of magnitude of the time in the timing effects?


        Tony




        --- In regsaudioforum@yahoogroups.com, "Robert" <regtas43@...> wrote:
        >
        > It would be interesting to know whether
        > this is a nonlinearity or whether it is
        > an effect of the band limiting of the woofer.
        > (the meaning of the Fourier transform
        > remark includes, though I did not say
        > so explictly. that the bandlimiting
        > of the device itself is going to alter
        > the waveform since it too acts as a filter
        > in addition to the crossover filtering).
        > There can be nonlinear
        > effects, too, of course especially
        > of the stick/slip sort.
        > So it would be interesting to figure out
        > which aspect was linear and which nonlinear.
        >
        > I agree that the ear will not be very sensitive
        > to this sort of thing--or so one would guess.
        >
        > But my experience with the Helsinkis made
        > me wonder if my faith in the insensitivity
        > of the ear to timing effects in the bass
        > might not be a little greater than was
        > justified! Still this looks really good to me.
        >
        > One can pretty much figure out which part of the effect
        > is linear and which part not, I think, by just running
        > a program to see what the output would be
        > of the linear system with the frequency and
        > phase response measured by Acourate via slow log
        > sweep at a moderate level, where nonlinearities
        > would one would hope be minimal and in particular
        > where stick/slip effects would be minimized
        > by the fact that the driver would be vibrating throughout
        > (but one needs to measure all the way down and all
        > the way up pretty far anyway, as you know).
        > Don't you think this would be reasonably accurate
        > (no pun intended)?
        >
        > Actually in audio terms(by the standards
        > of most systems) the whole thing
        > looks really dandy. But it would still be
        > interesting to see what is going on in
        > linear versus nonlinear terms
        >
        > REG
        >
        > --- In regsaudioforum@yahoogroups.com, "tonycdk" <tcdk@> wrote:
        > >
        > > Robert
        > >
        > > No Fourier transforms here. Look at the file 'subwoofer results (or is it measurements)' in my folder.
        > >
        > > Tony
        > >
        > > PS. When I first posted this material I seem to remember you telling me that you had observed the same type of thing.
        > >
        > >
        > >
        > > --- In regsaudioforum@yahoogroups.com, "Robert" <regtas43@> wrote:
        > > >
        > > > WHen you turn on a sine or cosine wave(from nothing)
        > > > the Fourier transform has got a bunch of stuff in it
        > > > in addition to the energy in the frequency of the
        > > > sine wave. So when you filter-split this signal and send the low pass
        > > > filtered part to the sub, then what happens
        > > > has some stuff in it that one might not expect.
        > > > Have a look at this
        > > > http://www.thefouriertransform.com/pairs/rightSidedSinusoids.php
        > > > and this
        > > > http://www.thefouriertransform.com/pairs/rightSidedSinusoids.php#sine.
        > > > REG
        > > >
        > > > PS Or is this what is being discussed? I missed the beginning
        > > > of this somehow...
        > > >
        > > > --- In regsaudioforum@yahoogroups.com, "tonycdk" <tcdk@> wrote:
        > > > >
        > > > > This effect showed that, starting from rest, my sub took a few cycles to stabilize at the input frequency. This is more an item of interest than a concern. The ear takes a few cycles before it identifies the frequency of a note and, in any case, even the first half cycle amplitude is down by less than 2dB.
        > > > >
        > > > > The measurement sets were taken at my listening position in my room and should not necessarily be taken as typical. I have tried a similar thing with a servo driven speaker with awful results - which may have been caused by the effects of the room (not mine).
        > > > >
        > > > >
        > > > > Tony
        > > > >
        > > > >
        > > > >
        > > > > --- In regsaudioforum@yahoogroups.com, "laurie483000" <laurie483000@> wrote:
        > > > >
        > > > > >
        > > > > > Whilst quickly perusing Tony's files something about subs also caught my eye. If I picked things up correctly, in doing some testing of subs talking, there is talk of delays in I.B. designs building up the sound level in reacting to the input signal, but I'll read a bit more before asking any more questions or creating anger and alarm.
        > > > > >
        > > > >
        > > >
        > >
        >
      Your message has been successfully submitted and would be delivered to recipients shortly.