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Old 25th February 2006, 02:28 PM   #11
PatPet is offline PatPet  Hong Kong
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Quote:
Originally posted by 4real
Again: you cannot FIR filter a non oversampled signal.

Well, you van of course, but not for this application. The whole point of this working is using oversampling.
I dont agree. What is the difference between summing or not summing in the digital circuitry? Assuming perfect DA conversion, the analog output should be the same as that of DA with summed input.
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Old 25th February 2006, 03:19 PM   #12
4real is offline 4real  Netherlands
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Quote:
Originally posted by rfbrw
Most definitely am. Pre and post edit. [/B]
Well, that's to bad then!

Let me explain why (I think) I am right:

What is the digital filter for: wel, exactly the same thing as the analog one would do, but this one works in the digital domain. So: we need to filter somewhere on the end of the digital bandwidth of the orriginal signal, lets say 20 Khz. Now we apply our 20 Khz FIR filter to the non oversampled signal and get a 20 Khz filterd non oversampled filter again. So now: what use is this? Answer: non what so ever! For the filter to work you would need a stopband bandwidth after the filter, and with non oversampled data, you don't have this.

If you thinkt is stupid or wrong, please elaborate, because with comments like this, you can better stay away for this forum completely.
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Old 25th February 2006, 04:38 PM   #13
4real is offline 4real  Netherlands
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Originally posted by rfbrw Wrong. It's function is to raise the sample rate thereby avoiding the need to use high order elliptic filters when removing the images.
Exactly! So it does exactly what a normal analog filter would do, only in the digital domain. Obviously you'll still need an analog filter, but it can be at a far higher frequency.

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I've around this place somewhat longer than you and I've seen wiiseacres like you come and go and short of a visit from the Grim Reaper I will not be going anywhere anytime soon.
So what! Since when does that mean anything? If you think I'm wrong, come with decent arguments. We're all here to learn, so and if I'm wrong, I want to be told, not called a stupid wiseacres (whatever that means...)!

It's to bad you didn't comment on the rest of my post, since that is the most important part!
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Old 25th February 2006, 04:51 PM   #14
rfbrw is offline rfbrw
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Quote:
Originally posted by 4real

It's to bad you didn't comment on the rest of my post, since that is the most important part!

I didn't comment on the rest of your post because it was irrelevant. A normal analogue filter cannot raise the sample rate. You need an oversampling digital filter to do that.
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Old 25th February 2006, 04:58 PM   #15
4real is offline 4real  Netherlands
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Quote:
Originally posted by rfbrw
I didn't comment on the rest of your post because it was irrelevant. A normal analogue filter cannot raise the sample rate. You need an oversampling digital filter to do that.
No it's not! The topic starter want to do FIR filtering in a non oversamping DAC. I say: you cannot do that for the reasons I already stated. Sure you can put an external ovesampling fiter in front of your (whatever) DAC, but that will do 4 to 8x at max. Imho, that is not really a lot, and I don't think It's of any use.
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Old 25th February 2006, 07:30 PM   #16
4real is offline 4real  Netherlands
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Are you really sure about that rfbrw? I never said it was mentioned in the opening post!

Quote:
Originally posted by PatPet
My intention of the approach to not summing the taps is to let the DACs run at 1fs which means the advantage of jitter immunity of non-os dacs is not taken away by oversampling.
So did I mis something here?
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Old 25th February 2006, 08:00 PM   #17
rfbrw is offline rfbrw
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Quote:
Originally posted by 4real


So did I mis something here?
Yes but where does one start? Seeking to match the alledged jitter immunity of nos-dacs does not make the thread about nos dacs.
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Old 25th February 2006, 09:27 PM   #18
4real is offline 4real  Netherlands
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Quote:
Originally posted by rfbrw
Yes but where does one start? Seeking to match the alledged jitter immunity of nos-dacs does not make the thread about nos dacs.
You still don't get to, do you! He want's to use a nos DAC, and want to use the FIR filtering (combining the (somtimes) present three stage filter).

So actually, it IS about nos DAC's, and this would mean that FIR filtering is of totally no use. That you can combine the three stage filter (if present) to just one is of course obvious, and is already said.

The actual problem is that the topic starter doesn't completely get that the FIR filter tabs are. They are no seperate signals, but just coefficients that are used with multiplication and delay (just to put it very simple just now), to create a filter.

The higher the frequency, the less tabs you need, meaning less delay. And also the higher sampling will help here, since it means you can process more tabs in the same time, resulting in less passband ripple. So it actually is a balace between speed, delay and quality.

The three filter in series will have not more or less delay than a single filter with about the same resulting response. It will however takes mess memory, since you can use three smaller tabs, and not one larger one I guess. But I also guess, that not all DAC's will use a three stage filter.

About the delay beeing audible: it's not of course, since there is no phase variation in the filter the delay is exactly the same at every frequency, and for both channels. There is no way you'll hear it. It's as if you would play the CD a few ms later that you inteded to do.

@Nemophyle: don't argue with rfbrw. You have been arround for even a shorter time than I did. you couldn't possibly know what you are talking about
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Old 25th February 2006, 09:52 PM   #19
Variac is offline Variac  Costa Rica
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FIR Digital Filter

unlike other forums, insults are not allowed here. Any posts with insults are cut and any wisdom therein is lost.

Next insult will result in a week in the bin.
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Old 26th February 2006, 03:45 AM   #20
wa2ise is offline wa2ise  United States
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If you don't use a tap's output, you effectively set that tap to zero.

Yes, one could just build a filter using many DACs being fed digital signals with appropriate delays applied to the inputs. And select difering sized summing resistors to create the tap weighting. Nobody does that, it's much cheaper and more accurate to upsample the digital audio and digitally filter that and then feed it to a higher percision 18 or 20 bit DAC. You'd still need the 8x or such system clock anyway, to provide the delays in my first circuit above. You just select a higher frequency crystal oscillator and use flip-flops to divide it down (very low jitter, not like PLLs).

I've forgotten why oversampling is considered bad anyway...
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