On Thu, 28 May 2015 12:36:32 -0500, "kaz" <37480@DSPRelated> wrote:

>>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
>>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.
>>
>>Hello ikram998563,
>>   After reading all the posts in this thread, 
>>do you now have the answer to your question?
>>
>>[-Rick-]
>
>Unfortunately this forum is unique in that every thread is hijacked to
>become battleground for all sorts of great ideas, possibilities,
>scenarios, clever workarounds ...etc away from understanding really what
>the post wanted in the first place and in what setup and background they
>are working. If you look at fpga fora you will see spot on short answers
>most of the time and contributors do not spoil a good answer just because
>they want to prove they know better answer.
>
>Kaz

That seems to be the normal evolution of casual group face-to-face
discussions, too, so I've never thought it was too remarkable.   I do
think it's worthwhile to do something like Rick did, though, and make
sure the OP got what they were after.   Sometimes when you do that the
OP has long since disappeared.   I don't know if that's because they
got what they wanted and left or they got annoyed at the digressions.

This place can be interesting sometimes.   ;)


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

On Thu, 28 May 2015 12:36:32 -0500, kaz wrote:

>>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
>>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.
>>
>>Hello ikram998563,
>>   After reading all the posts in this thread,
>>do you now have the answer to your question?
>>
>>[-Rick-]
> 
> Unfortunately this forum is unique in that every thread is hijacked to
> become battleground for all sorts of great ideas, possibilities,
> scenarios, clever workarounds ...etc away from understanding really what
> the post wanted in the first place and in what setup and background they
> are working. If you look at fpga fora you will see spot on short answers
> most of the time and contributors do not spoil a good answer just
> because they want to prove they know better answer.

Hardly unique, and I think we often get the job done.

Sometimes people ask simple questions to which there is no simple answer 
-- then, things get complex.

-- 

Tim Wescott
Wescott Design Services
http://www.wescottdesign.com

>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
>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.
>
>Hello ikram998563,
>   After reading all the posts in this thread, 
>do you now have the answer to your question?
>
>[-Rick-]

Unfortunately this forum is unique in that every thread is hijacked to
become battleground for all sorts of great ideas, possibilities,
scenarios, clever workarounds ...etc away from understanding really what
the post wanted in the first place and in what setup and background they
are working. If you look at fpga fora you will see spot on short answers
most of the time and contributors do not spoil a good answer just because
they want to prove they know better answer.

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

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 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.

Hello ikram998563,
   After reading all the posts in this thread, 
do you now have the answer to your question?

[-Rick-]

On Tue, 26 May 2015 02:05:12 +0000 (UTC), spope33@speedymail.org
(Steve Pope) wrote:

>kaz <37480@DSPRelated> replies to my post,
>
>>>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.
>
>>As my mind is limited to FPGA boundary I am a bit surprised at this
>>strategy. Is it really that common in "soft" dsp practice. You are going
>>the complicated way just to have lower sampling rate at mixing. 
>
>You said that the OP "needs to" sample at the higher rate in the
>mixer; I am simply stating this is not absolutely true.
>
>>I assume you will generate say 35Mhz and hence its multiples e.g. 70Mhz
>>then you will upsample baseband to that required by 35Mhz mixer (instead
>>of 70) then you create problem of aliasing then go to a higher sampling
>>rate to filter out the trouble created? or do you expect the analogue
>>engineer to sort it out. 
>
>I can think of reasons you'd want to subsample.  Say the signal bandwidth
>is low, say 50 KHz.  After the subsampled digital mixer, you could 
>implment a sigma-delta modulator, at a clock rate in the 2 to 4 MHz range, 
>that would emit a pulse-like signal that feeds into the analog domain, 
>and after that have a narrow, high-Q analog filter with a 70 MHz center 
>frequency from which the signal of interest could be recovered and 
>amplified.
>
>That may not sound very classy, but I can tell you it's extremely
>inexpensive -- all of your digital processing is at a low sample
>rate, and the analog circuits can be made to be cheap.
>
>You'd consider something like that for a high-volume, low-cost
>part (admittedly outside of the range of the OP's description of their
>problem).
>
>Steve

Those are all very good reasons to do that, and we've designed
products that way in the past.   If you can find an economical DAC
that has good enough output bandwidth to support subsampling, you can
potentially do away with a more complex (and expensive) AFE and just
digitize directly to IF.    Years ago we used that to get a 70MHz IF
with a 100MHz DAC and have also done 140 MHz IF the same way.   It
worked great with very low MER, but required carefully designed DAC
correction filters.   

These days DACs with higher sampling rates are not hard to come by,
but as you point out that means that the digital stuff has to run much
faster, which can also add significant cost depending on the
implementation.

Good tricks to know, in any case.


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

On Monday, May 25, 2015 at 3:54:15 AM UTC-4, ikram...@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.

What do you do for a living?

Dirk Bell

kaz <37480@DSPRelated> replies to my post,

>>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.

>As my mind is limited to FPGA boundary I am a bit surprised at this
>strategy. Is it really that common in "soft" dsp practice. You are going
>the complicated way just to have lower sampling rate at mixing. 

You said that the OP "needs to" sample at the higher rate in the
mixer; I am simply stating this is not absolutely true.

>I assume you will generate say 35Mhz and hence its multiples e.g. 70Mhz
>then you will upsample baseband to that required by 35Mhz mixer (instead
>of 70) then you create problem of aliasing then go to a higher sampling
>rate to filter out the trouble created? or do you expect the analogue
>engineer to sort it out. 

I can think of reasons you'd want to subsample.  Say the signal bandwidth
is low, say 50 KHz.  After the subsampled digital mixer, you could 
implment a sigma-delta modulator, at a clock rate in the 2 to 4 MHz range, 
that would emit a pulse-like signal that feeds into the analog domain, 
and after that have a narrow, high-Q analog filter with a 70 MHz center 
frequency from which the signal of interest could be recovered and 
amplified.

That may not sound very classy, but I can tell you it's extremely
inexpensive -- all of your digital processing is at a low sample
rate, and the analog circuits can be made to be cheap.

You'd consider something like that for a high-volume, low-cost
part (admittedly outside of the range of the OP's description of their
problem).

Steve

On Mon, 25 May 2015 21:08:56 +0000 (UTC), spope33@speedymail.org
(Steve Pope) wrote:

>Eric Jacobsen <eric.jacobsen@ieee.org> wrote:
>
>>On Mon, 25 May 2015 15:02:26 -0500, "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.
>
>>Yes, if you're reasonably careful you don't need any additional
>>filtering.   If you're not careful, and there's an analog IF filter,
>>you probably still don't need any additional digital filtering.
>
>I'm going to say that you always need a filter after this mixer.  It
>may be the filter is in the analog chain; and it might not even
>be an intentional filter, just a naturally-occuring filtering
>from parasitics, but it's stil here.
>
>>Since the sample rate at the output of the mixer is generally high,
>>I've always avoided digital filtering there to keep the complexity
>>low.   This has been for oversampled or subsampled applications.  So
>>far I've never had to do it, other than for something like a DAC droop
>>correction.
>
>So let's say you sample at 210 Ms/sec, a convenient number for this
>problem.  There will be aliases at 140 MHz and 280 MHz, and so forth.
>The only way these don't show up in your system output is they are
>being filtered out.  Maybe a sinx/x filter inherenet in your
>DAC is enough, maybe your next IF stage doesn't possibly have response at
>140 MHz; but it still constitutes a necessary filtering.
>
>Particularly if you are in a regulatory environment, you might need
>an explicit filter.
>
>Steve

Yup.   The images in the repeating Nyquist zones need to be considered
depending on the application.   That'll be true regardless of the
system and is usually in the domain of the reconstruction filter.

Usually if somebody is putting an IF at 70MHz there's a lot more
filtering before it gets where it's going.

But, as said before, there is no need to put a digital filter after
the mixer.   Even if you did, you'd still need to deal with the images
in the repeating Nyquist zones, so that's not part of any tradeoff in
the digital domain.



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

>Eric Jacobsen <eric.jacobsen@ieee.org> wrote:
>
>>On Mon, 25 May 2015 15:02:26 -0500, "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.
>
>>Yes, if you're reasonably careful you don't need any additional
>>filtering.   If you're not careful, and there's an analog IF filter,
>>you probably still don't need any additional digital filtering.
>
>I'm going to say that you always need a filter after this mixer.  It
>may be the filter is in the analog chain; and it might not even
>be an intentional filter, just a naturally-occuring filtering
>from parasitics, but it's stil here.

It looks like we discuss issues at system level rather than in the context
of a beginner's post asking about mixing/upsampling. This is fair enough
but in my work environment we have boards with almost fixed DAC/ADC
circuitry and most of changes are done at fpga level. In other words I as
digital designer will leave LPF of say DAC images to the existing analogue
design and only focus on FPGA domain. Hence I don't need any filter after
mixing. But the system needs it after DAC. As to sinx/x correction etc (I
can add equaliser, DPD...) are realy irrelevant to the post but do add
some practical stuff.

The positive thing I started to have better understanding of various posts
here after so many years.

Kaz



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

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

>On Mon, 25 May 2015 15:02:26 -0500, "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.

>Yes, if you're reasonably careful you don't need any additional
>filtering.   If you're not careful, and there's an analog IF filter,
>you probably still don't need any additional digital filtering.

I'm going to say that you always need a filter after this mixer.  It
may be the filter is in the analog chain; and it might not even
be an intentional filter, just a naturally-occuring filtering
from parasitics, but it's stil here.

>Since the sample rate at the output of the mixer is generally high,
>I've always avoided digital filtering there to keep the complexity
>low.   This has been for oversampled or subsampled applications.  So
>far I've never had to do it, other than for something like a DAC droop
>correction.

So let's say you sample at 210 Ms/sec, a convenient number for this
problem.  There will be aliases at 140 MHz and 280 MHz, and so forth.
The only way these don't show up in your system output is they are
being filtered out.  Maybe a sinx/x filter inherenet in your
DAC is enough, maybe your next IF stage doesn't possibly have response at
140 MHz; but it still constitutes a necessary filtering.

Particularly if you are in a regulatory environment, you might need
an explicit filter.

Steve