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Old 10th January 2006, 05:41 PM   #1
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Default dsp upsample question

Hello,
So we know that upsampling/interpolation is the process of padding zeros to the samples that are coming in, but what happens if instead of padding zeros, we hold the sample at that value?
Specifically, a forum member was doubling the data sample rate, i.e. same data is sampled 2 times before being processed by an interpolation filter. So assume the original data was a sampled sine wave with points spaced 1/fs apart. Now when you double the sample rate while holding the same data, joining the dots at every 1/2fs will not give you the same original sine wave. So does that mean that after the interpolation filter, the processed waveform will be different from the original waveform?

Thanks,
Richard
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Old 10th January 2006, 06:22 PM   #2
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Don't you mean creating zero valued samples ?
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Old 10th January 2006, 07:18 PM   #3
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Default I think I was confused

Yeah, doesn't quite make sense if I were to fill in zero-value samples, does it?

So then if I were to double the sample rate and now sample the same bit twice, would this be equivalent to upsampling by 2?

If this were true, then technically, I should be able to run the sampling clock at a much higher rate, thereby sampling the same data many times and thus push the images way out there where a 2-3 pole low-pass filter should take care of the out-of-band noise.

Where did I go wrong here? It can't be that simple.
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Old 10th January 2006, 09:23 PM   #4
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Haven't given it too much thought but doubling the sample rate and repeating samples would seem to get you back to where you started.
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Old 11th January 2006, 03:59 AM   #5
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But the images will be 2x or 4x higher, wouldn't it?
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Old 11th January 2006, 05:50 AM   #6
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Default Re: dsp upsample question

Quote:
Originally posted by rlim
But the images will be 2x or 4x higher, wouldn't it?
Given it more thought and I asssume you are referring to linear interpolation, oversampling at its simplest. Can be done with simple hardware with no need to dither,truncate or round-off. Philips use it in the Bistream process and Wadia and a few others also use it. Used on its own it sounds awful and it doesn't get any better as you increase the oversampling ratio. Unless you actually need the higher sample rate, kicking the images into touch is probably the only reason to go beyond 2x but then again this is not my area of expertise.
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Old 11th January 2006, 06:33 AM   #7
Werner is offline Werner  Europe
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Default Re: dsp upsample question

Quote:
Originally posted by rlim
[B]Hello,
So we know that upsampling/interpolation is the process of padding zeros
That's not what it is.


Quote:
Originally posted by rlim
But the images will be 2x or 4x higher, wouldn't it?
Not at all.

In the context of digital audio replay oversampling comprises of incr4easing the data rate (either by inserting 0 samples or by repeating the previous one(s), the difference is not very relevant here) following by steep digital low-pass filtering at half the original sample rate. It's that last part that makes the output fluid and pushes the images out to above half the target sample rate.
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Old 11th January 2006, 07:09 AM   #8
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Default Re: Re: dsp upsample question

Quote:
Originally posted by Werner

In the context of digital audio replay oversampling comprises of incr4easing the data rate (either by inserting 0 samples or by repeating the previous one(s), the difference is not very relevant here) following by steep digital low-pass filtering at half the original sample rate. It's that last part that makes the output fluid and pushes the images out to above half the target sample rate.
If the OP is referring to the thread I think he is then there is definitely no low digital pass filter involved as there is no oversampling filter involved.
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Old 11th January 2006, 06:28 PM   #9
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Default sorry I wasn't clear.

Probably because it's not even clear in my head either.
So let's recap. Let's assume we have data, bit clock and fs clock going into a 2X oversampled digital low-pass filter before hiting a dac. I have attached a drawing where in the left column, you see interpolation at work, where data samples are padded with zeros for every adjacent sample(middle plot). After going through the filter, the data is reconstructed but now at a rate of 1/2fs.

Now on the right column, instead of sending the data into the filter directly, we double the sample rate, sampling the same data twice, which after the 2X oversampling in the chip, will result in the middle plot - overall 4X oversampling.

So my question is whether doubling the sample rate before sending data into the 2x oversampling digital filter give us the same original waveform as just sending data straight to the digital filter? Seems like that's a way of increasing the overall oversampling ratio.

The thing that's confusing me is that the middle plot in the right column isn't quite 4X upsampling since the 'filled in' points are not zeros.
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Old 11th January 2006, 06:29 PM   #10
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Default oops. Here's the image

image attached.
Attached Images
File Type: gif upsample.gif (8.3 KB, 133 views)
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