On Wed, 17 Jun 2015 09:46:31 -0500, "alpha1" <103600@DSPRelated> wrote:

>Thank you Greg, Can u help me with the Robert orban's method????
>I am finding it difficult to understand.

Sorry. You'll have to go directly to Mr. Orban for that.

>On Wed, 17 Jun 2015 02:28:53 -0500, "alpha1" <103600@DSPRelated> wrote:
>
>>in 
>>http://www.dsprelated.com/showthread/comp.dsp/76256-1.php
>>you have designed an eighth order filter. What could be the error for
the
>>first order filter according to your FDLS method
>
>Off hand, I don't know, but it will be very large. If I used 8th-order
>in that previous case, it's probably because 8th-order was needed to
>satisfy some goodness-of-fit criterion. 
>
>I haven't received your direct email message, but once I do, I will send
>you the FDLS Matlab code. Then you can try it for yourself.

Thank you Greg, Can u help me with the Robert orban's method????
I am finding it difficult to understand.
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On Wed, 17 Jun 2015 02:28:53 -0500, "alpha1" <103600@DSPRelated> wrote:

>in 
>http://www.dsprelated.com/showthread/comp.dsp/76256-1.php
>you have designed an eighth order filter. What could be the error for the
>first order filter according to your FDLS method

Off hand, I don't know, but it will be very large. If I used 8th-order
in that previous case, it's probably because 8th-order was needed to
satisfy some goodness-of-fit criterion. 

I haven't received your direct email message, but once I do, I will send
you the FDLS Matlab code. Then you can try it for yourself.

>>On Tue, 16 Jun 2015 11:48:24 -0400, robert bristow-johnson
>><rbj@audioimagination.com> wrote:
>>
>>>dunno fer sure who "Robert" is.  i don't remember any solution i have 
>>>ever done for a 15/50 microsecond deemphasis filter.
>>
>>He's referring to Robert Orban.
>>
>>https://groups.google.com/d/msg/comp.dsp/9kuirTaABzg/CSg4hgh0saoJ
>
>yeah, Robert Orban.... I think Robert Orban's and Greg Berchin methods
>would realise an analog transfer function in digital domain with minimum
>error.... I am going through these methods....
>
>@ Greg I have mailed u...My name is sharvin
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Hi greg,

in 
http://www.dsprelated.com/showthread/comp.dsp/76256-1.php
you have designed an eighth order filter. What could be the error for the
first order filter according to your FDLS method

Thanks

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>On Tue, 16 Jun 2015 11:48:24 -0400, robert bristow-johnson
><rbj@audioimagination.com> wrote:
>
>>dunno fer sure who "Robert" is.  i don't remember any solution i have 
>>ever done for a 15/50 microsecond deemphasis filter.
>
>He's referring to Robert Orban.
>
>https://groups.google.com/d/msg/comp.dsp/9kuirTaABzg/CSg4hgh0saoJ

yeah, Robert Orban.... I think Robert Orban's and Greg Berchin methods
would realise an analog transfer function in digital domain with minimum
error.... I am going through these methods....

@ Greg I have mailed u...My name is sharvin
---------------------------------------
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On Tue, 16 Jun 2015 11:48:24 -0400, robert bristow-johnson
<rbj@audioimagination.com> wrote:

>dunno fer sure who "Robert" is.  i don't remember any solution i have 
>ever done for a 15/50 microsecond deemphasis filter.

He's referring to Robert Orban.

https://groups.google.com/d/msg/comp.dsp/9kuirTaABzg/CSg4hgh0saoJ

On 6/16/15 8:30 AM, alpha1 wrote:
> Hi,
>
> I went through
> https://groups.google.com/forum/#!original/comp.dsp/wvD_EPWgnek/NDywVIz6qlMJ
> I saw that Robert's poles and zeros at 48khz are really matching with the
> analog response. I used these poles and zeros and mapped them to poles and
> zeros at 44.1khz using BLT. I got error of 0.15dB only. But when I go for
> 32Khz or 24Khz the error is more than 0.5dB . I see Greg's and Robert's
> solutions are excellent. Can anyone help me with their design methods.

dunno fer sure who "Robert" is.  i don't remember any solution i have 
ever done for a 15/50 microsecond deemphasis filter.

-- 

r b-j                  rbj@audioimagination.com

"Imagination is more important than knowledge."

Hi,

I went through
https://groups.google.com/forum/#!original/comp.dsp/wvD_EPWgnek/NDywVIz6qlMJ
I saw that Robert's poles and zeros at 48khz are really matching with the
analog response. I used these poles and zeros and mapped them to poles and
zeros at 44.1khz using BLT. I got error of 0.15dB only. But when I go for
32Khz or 24Khz the error is more than 0.5dB . I see Greg's and Robert's
solutions are excellent. Can anyone help me with their design methods.

Thanks in advance
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On Tue, 09 Jun 2015 02:31:20 -0500, "alpha1" <103600@DSPRelated> wrote:

>Dear Greg Berchin,
>
>Could you please share your e-mail ID?
>
>Thanks in advance.

"I'm sorry, (alpha1). I'm afraid I can't do that."

Instead find the original comp.dsp thread at
https://groups.google.com/forum/#!original/comp.dsp/wvD_EPWgnek/NDywVIz6qlMJ
and locate my obfuscated address in the "From:" field there. Make the
changes that I detail in the message, and you'll have my email address.

Greg

>On Thu, 14 May 2015 12:59:43 -0500, "alpha1" <103600@DSPRelated> wrote:
>
>>I am designing a deemphasis filter of 15/50us specification. I am using
>>using bilinear transform method. But at frequencies near to nyquist
>>frequency I am getting an error of 1dB. I tried pre wrapping. But I
gives
>>error reduction only at the frequency where the wrapping is applied. 
>
>https://groups.google.com/d/topic/comp.dsp/WSMrkyEcBBo/discussion
>Eight years and twelve days. Time flies. 
>
>Send email directly to me if you want code for FDLS. Change "chatter" to
>"charter" and drop ".invalid".

Dear Greg Berchin,

Could you please share your e-mail ID?

Thanks in advance.
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