On 1/9/2010 10:25 PM, rickman wrote:
> On Jan 6, 1:49 pm, Jorge Rivas <j.rivas.prietoER...@THIStue.nl> wrote:
>> On 1/6/2010 6:51 PM, ralphmalph wrote:
>>
>>
>>
>>> On Jan 6, 12:33 pm, Jorge Rivas <j.rivas.prietoER...@THIStue.nl>
>>> wrote:
>>>> Hi all,
>>
>>>> I have a communications system that is streaming analog data. On the
>>>> receiver side, this data is converted by an ADC with an oversampling of
>>>> 4 related to the conversion rate of the DAC on the transmitter. As it is
>>>> a system under testing (with variable speed), we don't have a proper
>>>> steep analog filter before the ADC, so the oversampling will help to
>>>> achieve a better performance. In my receiver (a custom FPGA board), the
>>>> ADC gives 4 different streams (synchronized) with the interpolated data.
>>
>>>> The problem with this interpolated data is that the symbol rate
>>>> increases, so there is too many samples for the rest of the receiver. I
>>>> know I could discard 3 of the 4 streams and I would have the sent data
>>>> rate (fixing so many other troubles :D), but then the oversampling made
>>>> would be useless.
>>
>>>> Another option would be to downsample the samples directly inside the
>>>> FPGA, but this would give the same result as in the previous paragraph.
>>>> Which is the proper way to handle that high number of samples?
>>
>>>> Thanks in advance :)
>>
>>> I am not clear about some of your terminology, but I think you are
>>> describing a receiver that is oversampling a bandlimited signal.  It
>>> does not matter what the rate of the DAC is at the transmitter, only
>>> the frequency and bandwidth of the signal being sent.
>>
>>> You don't say what your signal frequency is, but I assume it is
>>> baseband at this point.  So if you oversample, you can discard samples
>>> simply.  But that tosses out all the advantage of oversampling in the
>>> first place.  To retain the advantages, you need to use a digital low
>>> pass filter before decimating.  In fact, if you use a FIR filter, you
>>> can gain some computational efficiency by omitting the calculations of
>>> the the discarded samples.
>>
>>> If your signal is at a frequency higher than your nyquist rate, you
>>> need to consider the frequency downconversion that will happen as you
>>> decimate.  If this does not put your signal at baseband where you need
>>> it, then you must use a digital mixer to achieve the final result,
>>> either by mixing before the downconversion or before.  Mixing after
>>> the downconversion allows you to combine the input low pass filter
>>> with the down conversion as described above.
>>
>>> Do I understand what you are trying to do or have I muffed it?
>>
>>> Rick
>>
>> Thanks for your reply, Rick.
>>
>> The transmitted signal is indeed baseband. The spectrum of the signal is
>> pretty flat and has a bandwidth of around 300 MHz. The DAC is sampling
>> at 600 MSPS.
> 
> Is this a typo?  If your signal has a 300 MHz bandwidth, then you must
> sample it at a minimum of 600 MSPS to accurately represent it
> discretely.  I understand that the DAC is creating the signal, but are
> you saying that the ADC is sampling at 2400 MSPS???  That is pretty
> durn fast!

Yes, not a typo. The DAC is creating the signal at 600 MSPS to meet
Nyquist. The ADC is actually a dual ADC. It is working at 600 MHz but it
samples data in both edges. You have then 4 streams of 600 MSPS (one for
each edge on each ADC).

> 
> 
>> I cannot use a digital low pass filter directly on the input streaming,
>> because the clock frequency would be too high for my FPGA. There is
>> another problem, though. Each one of the streams of interpolated data
>> that go into the FPGA are already parallelized and do not correspond
>> directly with the sampled data. The ADC gives 8 bits of resolution per
>> sample, but on the FPGA I receive 4 streams of 32-bit wide symbols each.
> 
> Ok, if you don't want to filter the ADC data before you decimate it,
> you can toss samples without filtering, but you will not gain any
> advantage from the higher sample rate.  What you say about the data
> being "parallelized" and not corresponding to the sampled data makes
> no sense to me.  The way the data is bundled at the input to the FPGA
> is not material except that you must sort it out inside the FPGA so
> that you have a sequential data stream.  If you are interleaving
> multiple ADCs to increase the sample rate, you just have to interleave
> the input data to match.  That would use a mux.
> 

With the parallelization I tried to mean (as John described much better
than me) than I am receiving all my data in parallel, with a much lower
symbol rate on each stream. So, let's say that I am receiving 16 8-bit
symbols, at the same time. Sorry for the sloppy description before...

> 
>> I can understand what happens when a signal is interpolated or
>> decimated, but I ran out of ideas quite fast when there are Serial to
>> Parallel converters in the middle. The only thing I can think of is bit
>> manipulation... Any idea?
> 
> I think you are confusing yourself.  Whether the data is serial or
> parallel is irrelevant to this exercise.  Data is data.  The only
> concern is that you keep the order of the samples consistent.
> 
> Your initial post seems to accurately describe your situation.  You
> can decimate without filtering and get nothing from the oversampling
> or you can use a digital filter before decimating which effectively
> averages the multiple samples to improve the S/N ratio.
> 
> So what exactly are you not clear on?

I was highly confusing myself, you are right. That, together with some
problems in basic DSP (...). Now I think I have everything more clear,
but still don't know how to handle the 16 8-bit symbols. Problem is that
if that data were serial, I would know how to do the filtering. I cannot
serialize the data, as that would give me a stream of 2.4 GSPS, too much
for the FPGA to handle serially.

Imagine I have a bank of 16 filters, each one applied to each of the 16
streams. Does filtering each one of the streams separately have the same
effect as filtering the hypothetical serial 2.4 GSPS from the ADC with
the same kind of filter? That would be my main problem now. I think that
I tried to explain the setup and got messed with all the descriptions,
my fault :/

Jorge

On Jan 6, 1:49 pm, Jorge Rivas <j.rivas.prietoER...@THIStue.nl> wrote:
> On 1/6/2010 6:51 PM, ralphmalph wrote:
>
>
>
> > On Jan 6, 12:33 pm, Jorge Rivas <j.rivas.prietoER...@THIStue.nl>
> > wrote:
> >> Hi all,
>
> >> I have a communications system that is streaming analog data. On the
> >> receiver side, this data is converted by an ADC with an oversampling of
> >> 4 related to the conversion rate of the DAC on the transmitter. As it is
> >> a system under testing (with variable speed), we don't have a proper
> >> steep analog filter before the ADC, so the oversampling will help to
> >> achieve a better performance. In my receiver (a custom FPGA board), the
> >> ADC gives 4 different streams (synchronized) with the interpolated data.
>
> >> The problem with this interpolated data is that the symbol rate
> >> increases, so there is too many samples for the rest of the receiver. I
> >> know I could discard 3 of the 4 streams and I would have the sent data
> >> rate (fixing so many other troubles :D), but then the oversampling made
> >> would be useless.
>
> >> Another option would be to downsample the samples directly inside the
> >> FPGA, but this would give the same result as in the previous paragraph.
> >> Which is the proper way to handle that high number of samples?
>
> >> Thanks in advance :)
>
> > I am not clear about some of your terminology, but I think you are
> > describing a receiver that is oversampling a bandlimited signal.  It
> > does not matter what the rate of the DAC is at the transmitter, only
> > the frequency and bandwidth of the signal being sent.
>
> > You don't say what your signal frequency is, but I assume it is
> > baseband at this point.  So if you oversample, you can discard samples
> > simply.  But that tosses out all the advantage of oversampling in the
> > first place.  To retain the advantages, you need to use a digital low
> > pass filter before decimating.  In fact, if you use a FIR filter, you
> > can gain some computational efficiency by omitting the calculations of
> > the the discarded samples.
>
> > If your signal is at a frequency higher than your nyquist rate, you
> > need to consider the frequency downconversion that will happen as you
> > decimate.  If this does not put your signal at baseband where you need
> > it, then you must use a digital mixer to achieve the final result,
> > either by mixing before the downconversion or before.  Mixing after
> > the downconversion allows you to combine the input low pass filter
> > with the down conversion as described above.
>
> > Do I understand what you are trying to do or have I muffed it?
>
> > Rick
>
> Thanks for your reply, Rick.
>
> The transmitted signal is indeed baseband. The spectrum of the signal is
> pretty flat and has a bandwidth of around 300 MHz. The DAC is sampling
> at 600 MSPS.

Is this a typo?  If your signal has a 300 MHz bandwidth, then you must
sample it at a minimum of 600 MSPS to accurately represent it
discretely.  I understand that the DAC is creating the signal, but are
you saying that the ADC is sampling at 2400 MSPS???  That is pretty
durn fast!


> I cannot use a digital low pass filter directly on the input streaming,
> because the clock frequency would be too high for my FPGA. There is
> another problem, though. Each one of the streams of interpolated data
> that go into the FPGA are already parallelized and do not correspond
> directly with the sampled data. The ADC gives 8 bits of resolution per
> sample, but on the FPGA I receive 4 streams of 32-bit wide symbols each.

Ok, if you don't want to filter the ADC data before you decimate it,
you can toss samples without filtering, but you will not gain any
advantage from the higher sample rate.  What you say about the data
being "parallelized" and not corresponding to the sampled data makes
no sense to me.  The way the data is bundled at the input to the FPGA
is not material except that you must sort it out inside the FPGA so
that you have a sequential data stream.  If you are interleaving
multiple ADCs to increase the sample rate, you just have to interleave
the input data to match.  That would use a mux.


> I can understand what happens when a signal is interpolated or
> decimated, but I ran out of ideas quite fast when there are Serial to
> Parallel converters in the middle. The only thing I can think of is bit
> manipulation... Any idea?

I think you are confusing yourself.  Whether the data is serial or
parallel is irrelevant to this exercise.  Data is data.  The only
concern is that you keep the order of the samples consistent.

Your initial post seems to accurately describe your situation.  You
can decimate without filtering and get nothing from the oversampling
or you can use a digital filter before decimating which effectively
averages the multiple samples to improve the S/N ratio.

So what exactly are you not clear on?

Rick

Jorge Rivas wrote:
> On 1/7/2010 2:44 AM, Les Cargill wrote:
>> Jorge Rivas wrote:
>>> Hi all,
>>>
>>> I have a communications system that is streaming analog data. On the
>>> receiver side, this data is converted by an ADC with an oversampling of
>>> 4 related to the conversion rate of the DAC on the transmitter. As it is
>>> a system under testing (with variable speed), we don't have a proper
>>> steep analog filter before the ADC, so the oversampling will help to
>>> achieve a better performance. In my receiver (a custom FPGA board), the
>>> ADC gives 4 different streams (synchronized) with the interpolated data.
>>>
>>> The problem with this interpolated data is that the symbol rate
>>> increases, so there is too many samples for the rest of the receiver. I
>>> know I could discard 3 of the 4 streams and I would have the sent data
>>> rate (fixing so many other troubles :D), but then the oversampling made
>>> would be useless.
>>>
>>> Another option would be to downsample the samples directly inside the
>>> FPGA, but this would give the same result as in the previous paragraph.
>>> Which is the proper way to handle that high number of samples?
>>>
>>> Thanks in advance :)
>>>
>>>
>>
>> You need to do a proper link budget. I don't see one here. And I don't
>> see how you do 3 or 4 streams with "an" DAC.
>>
>> Seriously, hire a proper systems engineer and get all this
>> sorted out.
>>
>> -- 
>> Les Cargill
>>
>>
> 
> Hi Les,
> 
> The link budget is not an issue right now. The 4 streams correspond to
> the outputs of an *ADC* working at a higher frequency than the rest of
> the system.
> 
> Jorge

My bad - others mentioned decimation. I did not mean the RF link
budget, but the bitrates of the various blocks in the digital
domain. Poor choice of words on my part.

--
Les Cargill

On 1/7/2010 2:44 AM, Les Cargill wrote:
> Jorge Rivas wrote:
>> Hi all,
>>
>> I have a communications system that is streaming analog data. On the
>> receiver side, this data is converted by an ADC with an oversampling of
>> 4 related to the conversion rate of the DAC on the transmitter. As it is
>> a system under testing (with variable speed), we don't have a proper
>> steep analog filter before the ADC, so the oversampling will help to
>> achieve a better performance. In my receiver (a custom FPGA board), the
>> ADC gives 4 different streams (synchronized) with the interpolated data.
>>
>> The problem with this interpolated data is that the symbol rate
>> increases, so there is too many samples for the rest of the receiver. I
>> know I could discard 3 of the 4 streams and I would have the sent data
>> rate (fixing so many other troubles :D), but then the oversampling made
>> would be useless.
>>
>> Another option would be to downsample the samples directly inside the
>> FPGA, but this would give the same result as in the previous paragraph.
>> Which is the proper way to handle that high number of samples?
>>
>> Thanks in advance :)
>>
>>
> 
> 
> You need to do a proper link budget. I don't see one here. And I don't
> see how you do 3 or 4 streams with "an" DAC.
> 
> Seriously, hire a proper systems engineer and get all this
> sorted out.
> 
> -- 
> Les Cargill
> 
> 

Hi Les,

The link budget is not an issue right now. The 4 streams correspond to
the outputs of an *ADC* working at a higher frequency than the rest of
the system.

Jorge

On 1/7/2010 12:03 AM, John wrote:
> On Jan 6, 1:49 pm, Jorge Rivas <j.rivas.prietoER...@THIStue.nl> wrote:
>> On 1/6/2010 6:51 PM, ralphmalph wrote:
>>
>>
>>
>>> On Jan 6, 12:33 pm, Jorge Rivas <j.rivas.prietoER...@THIStue.nl>
>>> wrote:
>>>> Hi all,
>>
>>>> I have a communications system that is streaming analog data. On the
>>>> receiver side, this data is converted by an ADC with an oversampling of
>>>> 4 related to the conversion rate of the DAC on the transmitter. As it is
>>>> a system under testing (with variable speed), we don't have a proper
>>>> steep analog filter before the ADC, so the oversampling will help to
>>>> achieve a better performance. In my receiver (a custom FPGA board), the
>>>> ADC gives 4 different streams (synchronized) with the interpolated data.
>>
>>>> The problem with this interpolated data is that the symbol rate
>>>> increases, so there is too many samples for the rest of the receiver. I
>>>> know I could discard 3 of the 4 streams and I would have the sent data
>>>> rate (fixing so many other troubles :D), but then the oversampling made
>>>> would be useless.
>>
>>>> Another option would be to downsample the samples directly inside the
>>>> FPGA, but this would give the same result as in the previous paragraph.
>>>> Which is the proper way to handle that high number of samples?
>>
>>>> Thanks in advance :)
>>
>>> I am not clear about some of your terminology, but I think you are
>>> describing a receiver that is oversampling a bandlimited signal.  It
>>> does not matter what the rate of the DAC is at the transmitter, only
>>> the frequency and bandwidth of the signal being sent.
>>
>>> You don't say what your signal frequency is, but I assume it is
>>> baseband at this point.  So if you oversample, you can discard samples
>>> simply.  But that tosses out all the advantage of oversampling in the
>>> first place.  To retain the advantages, you need to use a digital low
>>> pass filter before decimating.  In fact, if you use a FIR filter, you
>>> can gain some computational efficiency by omitting the calculations of
>>> the the discarded samples.
>>
>>> If your signal is at a frequency higher than your nyquist rate, you
>>> need to consider the frequency downconversion that will happen as you
>>> decimate.  If this does not put your signal at baseband where you need
>>> it, then you must use a digital mixer to achieve the final result,
>>> either by mixing before the downconversion or before.  Mixing after
>>> the downconversion allows you to combine the input low pass filter
>>> with the down conversion as described above.
>>
>>> Do I understand what you are trying to do or have I muffed it?
>>
>>> Rick
>>
>> Thanks for your reply, Rick.
>>
>> The transmitted signal is indeed baseband. The spectrum of the signal is
>> pretty flat and has a bandwidth of around 300 MHz. The DAC is sampling
>> at 600 MSPS.
>>
>> I cannot use a digital low pass filter directly on the input streaming,
>> because the clock frequency would be too high for my FPGA. There is
>> another problem, though. Each one of the streams of interpolated data
>> that go into the FPGA are already parallelized and do not correspond
>> directly with the sampled data. The ADC gives 8 bits of resolution per
>> sample, but on the FPGA I receive 4 streams of 32-bit wide symbols each.
>>
>> I can understand what happens when a signal is interpolated or
>> decimated, but I ran out of ideas quite fast when there are Serial to
>> Parallel converters in the middle. The only thing I can think of is bit
>> manipulation... Any idea?
> 
> From your description, the FPGA receives rate 4X ADC data in blocks of
> 16 8-bit samples. You want to decimate the rate to X. A crude
> decimator with sinx/x response would add four sets of four time-
> sequential 8-bit samples in each block of 16 together: 16 samples in,
> 4 samples out.
> 
> John

Hi John,

That is exactly what I need, but I don't think I understand your answer
completely.

I think you mean that I should have a 4-input decimator and introduce on
each input the 1st, 2nd, 3rd and 4th "part" of my received block on the
FPGA, being each of those parts respectively the 1st, 2nd, 3rd and 4th
part of the received time-sequence time on the ADC before the parallel
conversion, am I right?

Thanks a lot for your answer,

Jorge

Jorge Rivas wrote:
> Hi all,
> 
> I have a communications system that is streaming analog data. On the
> receiver side, this data is converted by an ADC with an oversampling of
> 4 related to the conversion rate of the DAC on the transmitter. As it is
> a system under testing (with variable speed), we don't have a proper
> steep analog filter before the ADC, so the oversampling will help to
> achieve a better performance. In my receiver (a custom FPGA board), the
> ADC gives 4 different streams (synchronized) with the interpolated data.
> 
> The problem with this interpolated data is that the symbol rate
> increases, so there is too many samples for the rest of the receiver. I
> know I could discard 3 of the 4 streams and I would have the sent data
> rate (fixing so many other troubles :D), but then the oversampling made
> would be useless.
> 
> Another option would be to downsample the samples directly inside the
> FPGA, but this would give the same result as in the previous paragraph.
> Which is the proper way to handle that high number of samples?
> 
> Thanks in advance :)
> 
> 


You need to do a proper link budget. I don't see one here. And I don't
see how you do 3 or 4 streams with "an" DAC.

Seriously, hire a proper systems engineer and get all this
sorted out.

--
Les Cargill

On Jan 6, 1:49=A0pm, Jorge Rivas <j.rivas.prietoER...@THIStue.nl> wrote:
> On 1/6/2010 6:51 PM, ralphmalph wrote:
>
>
>
> > On Jan 6, 12:33 pm, Jorge Rivas <j.rivas.prietoER...@THIStue.nl>
> > wrote:
> >> Hi all,
>
> >> I have a communications system that is streaming analog data. On the
> >> receiver side, this data is converted by an ADC with an oversampling o=
f
> >> 4 related to the conversion rate of the DAC on the transmitter. As it =
is
> >> a system under testing (with variable speed), we don't have a proper
> >> steep analog filter before the ADC, so the oversampling will help to
> >> achieve a better performance. In my receiver (a custom FPGA board), th=
e
> >> ADC gives 4 different streams (synchronized) with the interpolated dat=
a.
>
> >> The problem with this interpolated data is that the symbol rate
> >> increases, so there is too many samples for the rest of the receiver. =
I
> >> know I could discard 3 of the 4 streams and I would have the sent data
> >> rate (fixing so many other troubles :D), but then the oversampling mad=
e
> >> would be useless.
>
> >> Another option would be to downsample the samples directly inside the
> >> FPGA, but this would give the same result as in the previous paragraph=
.
> >> Which is the proper way to handle that high number of samples?
>
> >> Thanks in advance :)
>
> > I am not clear about some of your terminology, but I think you are
> > describing a receiver that is oversampling a bandlimited signal. =A0It
> > does not matter what the rate of the DAC is at the transmitter, only
> > the frequency and bandwidth of the signal being sent.
>
> > You don't say what your signal frequency is, but I assume it is
> > baseband at this point. =A0So if you oversample, you can discard sample=
s
> > simply. =A0But that tosses out all the advantage of oversampling in the
> > first place. =A0To retain the advantages, you need to use a digital low
> > pass filter before decimating. =A0In fact, if you use a FIR filter, you
> > can gain some computational efficiency by omitting the calculations of
> > the the discarded samples.
>
> > If your signal is at a frequency higher than your nyquist rate, you
> > need to consider the frequency downconversion that will happen as you
> > decimate. =A0If this does not put your signal at baseband where you nee=
d
> > it, then you must use a digital mixer to achieve the final result,
> > either by mixing before the downconversion or before. =A0Mixing after
> > the downconversion allows you to combine the input low pass filter
> > with the down conversion as described above.
>
> > Do I understand what you are trying to do or have I muffed it?
>
> > Rick
>
> Thanks for your reply, Rick.
>
> The transmitted signal is indeed baseband. The spectrum of the signal is
> pretty flat and has a bandwidth of around 300 MHz. The DAC is sampling
> at 600 MSPS.
>
> I cannot use a digital low pass filter directly on the input streaming,
> because the clock frequency would be too high for my FPGA. There is
> another problem, though. Each one of the streams of interpolated data
> that go into the FPGA are already parallelized and do not correspond
> directly with the sampled data. The ADC gives 8 bits of resolution per
> sample, but on the FPGA I receive 4 streams of 32-bit wide symbols each.
>
> I can understand what happens when a signal is interpolated or
> decimated, but I ran out of ideas quite fast when there are Serial to
> Parallel converters in the middle. The only thing I can think of is bit
> manipulation... Any idea?

From your description, the FPGA receives rate 4X ADC data in blocks of
16 8-bit samples. You want to decimate the rate to X. A crude
decimator with sinx/x response would add four sets of four time-
sequential 8-bit samples in each block of 16 together: 16 samples in,
4 samples out.

John

On 1/6/2010 6:51 PM, ralphmalph wrote:
> On Jan 6, 12:33 pm, Jorge Rivas <j.rivas.prietoER...@THIStue.nl>
> wrote:
>> Hi all,
>>
>> I have a communications system that is streaming analog data. On the
>> receiver side, this data is converted by an ADC with an oversampling of
>> 4 related to the conversion rate of the DAC on the transmitter. As it is
>> a system under testing (with variable speed), we don't have a proper
>> steep analog filter before the ADC, so the oversampling will help to
>> achieve a better performance. In my receiver (a custom FPGA board), the
>> ADC gives 4 different streams (synchronized) with the interpolated data.
>>
>> The problem with this interpolated data is that the symbol rate
>> increases, so there is too many samples for the rest of the receiver. I
>> know I could discard 3 of the 4 streams and I would have the sent data
>> rate (fixing so many other troubles :D), but then the oversampling made
>> would be useless.
>>
>> Another option would be to downsample the samples directly inside the
>> FPGA, but this would give the same result as in the previous paragraph.
>> Which is the proper way to handle that high number of samples?
>>
>> Thanks in advance :)
> 
> I am not clear about some of your terminology, but I think you are
> describing a receiver that is oversampling a bandlimited signal.  It
> does not matter what the rate of the DAC is at the transmitter, only
> the frequency and bandwidth of the signal being sent.
> 
> You don't say what your signal frequency is, but I assume it is
> baseband at this point.  So if you oversample, you can discard samples
> simply.  But that tosses out all the advantage of oversampling in the
> first place.  To retain the advantages, you need to use a digital low
> pass filter before decimating.  In fact, if you use a FIR filter, you
> can gain some computational efficiency by omitting the calculations of
> the the discarded samples.
> 
> If your signal is at a frequency higher than your nyquist rate, you
> need to consider the frequency downconversion that will happen as you
> decimate.  If this does not put your signal at baseband where you need
> it, then you must use a digital mixer to achieve the final result,
> either by mixing before the downconversion or before.  Mixing after
> the downconversion allows you to combine the input low pass filter
> with the down conversion as described above.
> 
> Do I understand what you are trying to do or have I muffed it?
> 
> Rick

Thanks for your reply, Rick.

The transmitted signal is indeed baseband. The spectrum of the signal is
pretty flat and has a bandwidth of around 300 MHz. The DAC is sampling
at 600 MSPS.

I cannot use a digital low pass filter directly on the input streaming,
because the clock frequency would be too high for my FPGA. There is
another problem, though. Each one of the streams of interpolated data
that go into the FPGA are already parallelized and do not correspond
directly with the sampled data. The ADC gives 8 bits of resolution per
sample, but on the FPGA I receive 4 streams of 32-bit wide symbols each.

I can understand what happens when a signal is interpolated or
decimated, but I ran out of ideas quite fast when there are Serial to
Parallel converters in the middle. The only thing I can think of is bit
manipulation... Any idea?

On Jan 6, 12:33=A0pm, Jorge Rivas <j.rivas.prietoER...@THIStue.nl>
wrote:
> Hi all,
>
> I have a communications system that is streaming analog data. On the
> receiver side, this data is converted by an ADC with an oversampling of
> 4 related to the conversion rate of the DAC on the transmitter. As it is
> a system under testing (with variable speed), we don't have a proper
> steep analog filter before the ADC, so the oversampling will help to
> achieve a better performance. In my receiver (a custom FPGA board), the
> ADC gives 4 different streams (synchronized) with the interpolated data.
>
> The problem with this interpolated data is that the symbol rate
> increases, so there is too many samples for the rest of the receiver. I
> know I could discard 3 of the 4 streams and I would have the sent data
> rate (fixing so many other troubles :D), but then the oversampling made
> would be useless.
>
> Another option would be to downsample the samples directly inside the
> FPGA, but this would give the same result as in the previous paragraph.
> Which is the proper way to handle that high number of samples?
>
> Thanks in advance :)

I am not clear about some of your terminology, but I think you are
describing a receiver that is oversampling a bandlimited signal.  It
does not matter what the rate of the DAC is at the transmitter, only
the frequency and bandwidth of the signal being sent.

You don't say what your signal frequency is, but I assume it is
baseband at this point.  So if you oversample, you can discard samples
simply.  But that tosses out all the advantage of oversampling in the
first place.  To retain the advantages, you need to use a digital low
pass filter before decimating.  In fact, if you use a FIR filter, you
can gain some computational efficiency by omitting the calculations of
the the discarded samples.

If your signal is at a frequency higher than your nyquist rate, you
need to consider the frequency downconversion that will happen as you
decimate.  If this does not put your signal at baseband where you need
it, then you must use a digital mixer to achieve the final result,
either by mixing before the downconversion or before.  Mixing after
the downconversion allows you to combine the input low pass filter
with the down conversion as described above.

Do I understand what you are trying to do or have I muffed it?

Rick