Digital Upconversion using the baseband data(IQ) files

Hi all,
     i want to upconvert the baseband data (complex data). i have code for upconverting but struggling to implement interpolater (resampler) before upconverter and i need to implement a filter after the upconverter. so can any one help me to implement both resampler and filter . Iam trying to upconvert a baseband to 70Mhz IF .



Thanks in advance.

On 25.5.15 10:54, ikram998563@gmail.com wrote:
> Hi all,
>       i want to upconvert the baseband data (complex data). i have code for upconverting but struggling to implement interpolater (resampler) before upconverter and i need to implement a filter after the upconverter. so can any one help me to implement both resampler and filter . Iam trying to upconvert a baseband to 70Mhz IF .
>
>
>
> Thanks in advance.
>

Homework?

-- 

-TV

On Mon, 25 May 2015 00:54:11 -0700 (PDT), ikram998563@gmail.com wrote:

>Hi all,
>     i want to upconvert the baseband data (complex data). i have code for =
>upconverting but struggling to implement interpolater (resampler) before up=
>converter and i need to implement a filter after the upconverter. so can an=
>y one help me to implement both resampler and filter . Iam trying to upconv=
>ert a baseband to 70Mhz IF .
>
>
>
>Thanks in advance.

If you LPF filter it at baseband and use a perfect (e.g., digital)
complex up-converter, you shouldn't need to filter it again after
upconversion.

Any number of interpolating algorithms can then be used before the
upconverter, including multi-rate filters if the upsample ratio isn't
integer.

You're asking about what may be a fairly complex operation without
providing any details of your system, so it's difficult to give a
detailed answer.

Plus maybe homework, so...   ;)


Eric Jacobsen
Anchor Hill Communications
http://www.anchorhill.com

Eric Jacobsen <eric.jacobsen@ieee.org> wrote:

>If you LPF filter it at baseband and use a perfect (e.g., digital)
>complex up-converter, you shouldn't need to filter it again after
>upconversion.

Assuming a finite sample rate in the upconverter, you still have
aliases at its output and your might want to explicitly filter
them out.

Steve

On Mon, 25 May 2015 18:27:25 +0000 (UTC), spope33@speedymail.org
(Steve Pope) wrote:

>Eric Jacobsen <eric.jacobsen@ieee.org> wrote:
>
>>If you LPF filter it at baseband and use a perfect (e.g., digital)
>>complex up-converter, you shouldn't need to filter it again after
>>upconversion.
>
>Assuming a finite sample rate in the upconverter, you still have
>aliases at its output and your might want to explicitly filter
>them out.
>
>Steve

Yeah, I was making some assumptions which I shouldn't have.


Eric Jacobsen
Anchor Hill Communications
http://www.anchorhill.com

if your baseband is of width 0~B MHz and you target to move it to +70Mhz
centre then make sure the sampling rate at mixer is > 2*(70 + B) Msps.
Your I/Q data must also have been shaped and then lifted up to same
sampling rate of mixer. As such you don't need any further filter after
mixer.

Kaz

---------------------------------------
Posted through http://www.DSPRelated.com

kaz <37480@DSPRelated> wrote:

>if your baseband is of width 0~B MHz and you target to move it to +70Mhz
>centre then make sure the sampling rate at mixer is > 2*(70 + B) Msps.

Actually it can be made to work with the mixer implemented at any
sample rate > 2*B Msps.  This is sometimes called subsampling.  But then 
you would definitely need a post-filter, and *its* sample rate (if it
is a digital filter) would have to be > 2*(70+B) Msps.

Steve

>kaz <37480@DSPRelated> wrote:
>
>>if your baseband is of width 0~B MHz and you target to move it to
>+70Mhz
>>centre then make sure the sampling rate at mixer is > 2*(70 + B) Msps.
>
>Actually it can be made to work with the mixer implemented at any
>sample rate > 2*B Msps.  This is sometimes called subsampling.  But then
>
>you would definitely need a post-filter, and *its* sample rate (if it
>is a digital filter) would have to be > 2*(70+B) Msps.
>
>Steve

But the target is 70Mhz absolute

Kaz
---------------------------------------
Posted through http://www.DSPRelated.com

kaz <37480@DSPRelated> wrote:

>>Actually it can be made to work with the mixer implemented at any
>>sample rate > 2*B Msps.  This is sometimes called subsampling.  But then

Oops.  4*B Msps, since the bandwidth of the signal being constructed
is 2*B MHz, centered at 70 MHz.

>>you would definitely need a post-filter, and *its* sample rate (if it
>>is a digital filter) would have to be > 2*(70+B) Msps.

>But the target is 70Mhz absolute

So long as the Nyquist criterion is met, the signal will contain
the desired component centered at 70 MHz, along with other aliases.  
Higher sample rates place those aliases further away from the component
of interest, and therefore easier to filter out.

Steve

>kaz <37480@DSPRelated> wrote:
>
>>>Actually it can be made to work with the mixer implemented at any
>>>sample rate > 2*B Msps.  This is sometimes called subsampling.  But
>then
>
>Oops.  4*B Msps, since the bandwidth of the signal being constructed
>is 2*B MHz, centered at 70 MHz.
>
>>>you would definitely need a post-filter, and *its* sample rate (if it
>>>is a digital filter) would have to be > 2*(70+B) Msps.
>
>>But the target is 70Mhz absolute
>
>So long as the Nyquist criterion is met, the signal will contain
>the desired component centered at 70 MHz, along with other aliases.  
>Higher sample rates place those aliases further away from the component
>of interest, and therefore easier to filter out.
>
>Steve

yes you can do it that way but if I say anything like that in my field
(FPGA DSP) I will lose my job instantly. We do undersampling at adc level
of IF signal but in the digits we don't. FPGA target minimum resource and
minimum modules. Soft DSP can play with equations as they wish. It looks
like there is a big big difference in the prespectives.

Kaz
---------------------------------------
Posted through http://www.DSPRelated.com