Hilbert filter bandwith

Hi Svilen:

Can you tell us where the 2.2kHz number you mention comes from? We believe this should be much Wider than that.

Regards

Miguel

Hi Svilen:

Please find attached the phase response of the Hilbert Transform provided in SigmaStudio.

Best regards

Miguel

I check again and find a error in my design. I was using copy of input signal instead second output of Hilbert transform. After making some changes I get the expected result.

Thank you.

Hi Svilen:

I'm glad to hear.

Regards

Where I find on SigmaStudio for 1701 Hilberts Transform

Thanks

Hi miguel,

thanks for the answer you are always kind.

I am working at night, because during the day I don't succeed I am learning to use Sigma,

I have succeeded in building a simple and good modulator and demodulator ssb.

Now I have the problem for the shift of 90 degrees the signal because I have to use the

upper and the lower sideband.

Considering that hilbert he cannot currently use, you can advise me to resolve the problem on 1701.

Thanks

Piero

Hi Piero:

We have received a good amount of requests for this. It should become available in the next SigmaStudio release early Spring 2013. The current implementation works only for the second generation SigmaDSP (1761, 1781, 144x)

Regards

Miguel

Hi,

You can use FIR Hilbert filter in ADAU 1701.

Correct, I would try an FIR approach. Thank you bluekf.

Like this. Fs=48 kHz

coefficients:

-0.00000000000001

-0.01130421534816

0.00000000000001

-0.00404243751853

-0.00000000000000

-0.00476230330122

0.00000000000000

-0.00556653243479

-0.00000000000000

-0.00646680643874

0.00000000000000

-0.00747026686182

-0.00000000000000

-0.00858797303542

-0.00000000000000

-0.00983355743268

-0.00000000000000

-0.01122821875967

0.00000000000000

-0.01280082930516

-0.00000000000000

-0.01457473353549

0.00000000000000

-0.01657721670001

-0.00000000000000

-0.01885375157264

0.00000000000000

-0.02151092509395

-0.00000000000000

-0.02458834062453

-0.00000000000000

-0.02825596308737

0.00000000000000

-0.03269067825482

-0.00000000000000

-0.03819723468368

0.00000000000000

-0.04525745806001

0.00000000000000

-0.05470854227249

0.00000000000000

-0.06813002647422

0.00000000000000

-0.08890832501744

0.00000000000000

-0.12586371909997

0.00000000000000

0.21132734326465

0.00000000000000

0.63632940840259

0.63632940840259

0.00000000000000

0.21132734326465

0.00000000000000

0.12586371909997

0.00000000000000

0.08890832501744

0.00000000000000

0.06813002647422

0.00000000000000

0.05470854227249

0.00000000000000

0.04525745806001

0.00000000000000

0.03819723468368

0.00000000000000

0.03269067825482

0.00000000000000

0.02825596308737

0.00000000000000

0.02458834062453

0.00000000000000

0.02151092509395

0.00000000000000

0.01885375157264

0.00000000000000

0.01657721670001

0.00000000000000

0.01457473353549

0.00000000000000

0.01280082930516

0.00000000000000

0.01122821875967

0.00000000000000

0.00983355743268

0.00000000000000

0.00858797303542

0.00000000000000

0.00747026686182

0.00000000000000

0.00646680643874

0.00000000000000

0.00556653243479

-0.00000000000000

0.00476230330122

0.00000000000000

0.00404243751853

-0.00000000000001

0.01130421534816

0.00000000000001

Hi Blueukf,

Your FIR filter makes an incredibly fine "spatializer" for headphones. Put it in one stereo channel with a compensating N/2 delay in the other, and have a listen. I find that it changes the usual "in your head" effect to feeling like the band is playing all around me! Thanks so much for your interesting post.

Best regards,

Bob

thanks blueukf,

You could explain me as you have done to calculate it and what program you have used, because my sample rate it is 96000.

Piero

Piero,

You can use interactive tool form MATLAB - FDA Tool.

MiKe

Thanks Mike

Thanks, blueukf

Piero,

This are coefficienst for your filter. Fs=96 kHz, magnitude ripple (300Hz-47000 Hz)=+/-0.3 dB (appr). Delay is (N-1)/2. N=251

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