comp.dsp | 90 degree phase shift| page 2

Jerry Avins wrote:
> ingresman wrote:
>> Thanks for the link Andor, I found it in an earlier post but to be
>> honest the hilbert looks a bit odd. It just asks for sample rate and
>> the impulse length. The sample rate has no impact on the ouput
>> generated so I'm still a little confused. ...
> 
> The sample rate is a standard question that is needed for all the 
> filters _but_ the HT. It is there only as a vestige. Put in whatever 
> value will make the frequency axis of output plot "pretty".
> 
> Jerry

I guessed the sample rate would be required merely to format the graphs. 
However, it doesn't even do that. The graphs are labeled as a fraction 
of Fs. Its not clear why the sample rate is needed at all.

Steve

ingresman wrote:
> Thanks guys I'll track down that book.
>
> As for the encoding Wendy Carlos has a great deal of info on her site.
> You might remember switched on Back. The link is
> http://www.wendycarlos.com/surround/surround4.html#matrix

I remember back when "Switched on Bach" was released. It was by Walter
Carlos.

It seems that more than music may be synthesized ;-)

Clay

Hi All,

Sorry to be away for time. I've pulled together some code to do a
hilbert transform (from the university of york site) which i'll include
below in case anyone will find it usefull. If put a sample trought the
transform and then again will you get an invrerted signal i.e.
180degree shift. This code doesn't seem to do that si I'd be interested
in comments
cheers

#define NZEROS 500
#define GAIN   1.571116176e+00

static float xv_left[NZEROS+1];

static float xcoeffs[] =
  { +0.0000000000, +0.0003214310, +0.0000000000, +0.0003252099,
    +0.0000000000, +0.0003302355, +0.0000000000, +0.0003365372,
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    -0.0000000000, -0.0005757189, -0.0000000000, -0.0005483367,
    -0.0000000000, -0.0005226689, -0.0000000000, -0.0004986818,
    -0.0000000000, -0.0004763424, -0.0000000000, -0.0004556178,
    -0.0000000000, -0.0004364757, -0.0000000000, -0.0004188840,
    -0.0000000000, -0.0004028113, -0.0000000000, -0.0003882260,
    -0.0000000000, -0.0003750973, -0.0000000000, -0.0003633944,
    -0.0000000000, -0.0003530868, -0.0000000000, -0.0003441444,
    -0.0000000000, -0.0003365372, -0.0000000000, -0.0003302355,
    -0.0000000000, -0.0003252099, -0.0000000000, -0.0003214310,
    -0.0000000000,
  };
float shiftLeft(float isamp)
{
 float sum; int i;
 for (i = 0; i < NZEROS; i++)       xv_left[i] = xv_left[i+1];
 xv_left[NZEROS] = isamp / GAIN;
 sum = 0.0;
 for (i = 0; i <= NZEROS; i++)
           sum += (xcoeffs[i] * xv_left[i]);
       return sum;
}

Just put a sample in as a parameter and you shoud get a shifted value
out as the result. The code has compiled under Visual C on windows.
Clay wrote:

> ingresman wrote:
> > Thanks guys I'll track down that book.
> >
> > As for the encoding Wendy Carlos has a great deal of info on her site.
> > You might remember switched on Back. The link is
> > http://www.wendycarlos.com/surround/surround4.html#matrix
>
> I remember back when "Switched on Bach" was released. It was by Walter
> Carlos.
> 
> It seems that more than music may be synthesized ;-)
> 
> Clay

ingresman wrote:
> Hi All,
>
> Sorry to be away for time. I've pulled together some code to do a
> hilbert transform (from the university of york site) which i'll include
> below in case anyone will find it usefull.If put a sample trought the
> transform and then again will you get an invrerted signal i.e.
> 180degree shift. This code doesn't seem to do that si I'd be interested
> in comments

[code snipped]

The Hilbert transform is a bandpass operation. For a full band signal,
applying the HT twice will be different than multiplying by -1 [1]. Try
testing your implementation with a single sine wave and see if you get
the expected result. Be careful to align the input and output according
to the delay of the HT filter.

Regards,
Andor

[1] To find out whether multiplying by -1 is indeed a 180=B0 phase
shift, read this thread (beware: 2 hours spare time needed :-):
http://groups.google.com/group/comp.dsp/browse_frm/thread/af1b4caf51d6c239/=
6c8e066df0ec0daf?#6c8e066df0ec0daf

Andor wrote:

    ...

> The Hilbert transform is a bandpass operation. For a full band signal,
> applying the HT twice will be different than multiplying by -1 [1]

   ...

That's only because most of us aren't willing or able to wait for 90 
degrees of shift at 1E-21 Hz, let alone DC. As always, we have to 
distinguish between the theoretical and the practical when making 
pronouncements.

If Ingressman's filter had 1E-21 coefficients, it might come closer to 
meeting his expectations.

Jerry
-- 
Engineering is the art of making what you want from things you can get.
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90 degree phase shift

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