ASRC and Oversampling Filters

percy · 10th February 2009, 04:44 PM

Is there really any difference between ASRCs (like TI SRC4193/4) and Oversampling Digital Filters (like NPC SM5842/7) in the way they do integer (4x,8x) oversampling ?

I suppose it would be possible to set a integer Fsout/Fsin ratio in an ASRC ? And in that case, is there any difference in the way it accomplishes the oversampled output, as compared to a filter that is a "4x/8x oversampling digital filter" ?

One more question about OS DFs -
Typically they utilize a stepped approach to oversample, so if you wanted 8x oversampling they would first oversample to 2x, then they would oversample that result into 4x and then repeat the same exercise to get 8x. Basically use the result of the previous step as an input to the next step. Now I can understand why they might do that. They might want to make it flexible so the user can use any ratio they want, and it also seems like this method is less demanding in terms of processing power. BUT, my question is -

if there was ample processing power at your disposal, and 8x was the only ratio desired, then would the results(distortion, noise) of doing 8x oversampling at one shot be any different than doing the oversampling in three steps ?

percy · 11th February 2009, 04:47 PM

come on, I know there are DSP geeks and gurus around here who can easily answer my second question (about doing 8x v/s doing 2x + 2x + 2x).

darkfenriz · 11th February 2009, 04:57 PM

Quote:

Is there really any difference between ASRCs (like TI SRC4193/4) and Oversampling Digital Filters (like NPC SM5842/7) in the way they do integer (4x,8x) oversampling ?

To my understanding they both only increase the number of input samples by repeating or zero stuffing and apply a FIR filter.
And both are basically upsamplers, either synchronous or asynchronous. I am not a big expert on this though...

Wingfeather · 11th February 2009, 11:33 PM

Hi percy,

From the datasheet, the SRC4192/3 is a conventional 16x-oversampler, followed by a *resampler* and then a conventional (adjustable?) decimator. It doesn't say in the datasheet quite how the resampler works, but I would imagine it to be based on some sort of polynomial interpolation process.

As for the question of which method is best, it's really hard to say. I designed a few filters quickly in MATLAB (one 600-tap 8x oversampling filter and 3 polyphase 2x filters (150-tap, 20-tap and 16-tap) for cascading). I was pretty quick in the designs, but aimed for minimal passband ripple, minimal attenuation at 20kHz and 120-dB stopband attenuation. Overall, the responses of the two approaches in both frequency- and time-domain looked very similar. I would suggest that the 8x filter is somewhat less fiddly to design, but then you're looking at 600 MACs/sample as opposed to about 90 MACs/sample for the cascaded approach. Not really worth the simplicity on that front

To be practical, the cascaded approach would need a quantisation after each filter, as opposed to one single quantisation after the 8x filter, so there'd be 3 times as much quantisation noise in the cascaded version. If you use long enough word-lengths to begin with though, and dither at each truncation (see my many rants on dither on this forum if you don't know about it), then this is a non-issue.

If I were designing an oversampler, I'd not really be able to justify to myself the extra expense of the single-filter solution, even if I had the computing power. The gains just aren't there.

banana · 12th February 2009, 12:15 AM

The plain old AD1890 datasheet have some explanation on the principle of ASRC.

Link to AD1890 datasheet

ezkcdude · 12th February 2009, 12:15 AM

Quote:

Originally posted by percy
Is there really any difference between ASRCs (like TI SRC4193/4) and Oversampling Digital Filters (like NPC SM5842/7) in the way they do integer (4x,8x) oversampling ?

I suppose it would be possible to set a integer Fsout/Fsin ratio in an ASRC ? And in that case, is there any difference in the way it accomplishes the oversampled output, as compared to a filter that is a "4x/8x oversampling digital filter" ?

One more question about OS DFs -
Typically they utilize a stepped approach to oversample, so if you wanted 8x oversampling they would first oversample to 2x, then they would oversample that result into 4x and then repeat the same exercise to get 8x. Basically use the result of the previous step as an input to the next step. Now I can understand why they might do that. They might want to make it flexible so the user can use any ratio they want, and it also seems like this method is less demanding in terms of processing power. BUT, my question is -

if there was ample processing power at your disposal, and 8x was the only ratio desired, then would the results(distortion, noise) of doing 8x oversampling at one shot be any different than doing the oversampling in three steps ?

Classic ASRC thread.

percy · 12th February 2009, 03:19 PM

Wingfeather's reply is the closest to what I was looking for. I understand how sample rate conversion and oversampling works. I am trying to compare the working of an ASRC with an OSDF. Its intuitive to think they would be the same but apparently not.

Wingfeather, I saw the functional block diagram of the SRC4192/3. Would you agree that the SRC4192/3 does a lot more messing around with the data then a straightforward 8x oversampling filter ? Thanks for giving it a try in matlab though, appreciate that. However -

Quote:

so there'd be 3 times as much quantisation noise in the cascaded version...

you mean out of band noise ?

Quote:

The gains just aren't there.

but isn't that what they said for the "1-Bit" Dac too ?

Telstar · 12th February 2009, 03:25 PM

Quote:

Originally posted by Wingfeather

If I were designing an oversampler, I'd not really be able to justify to myself the extra expense of the single-filter solution, even if I had the computing power. The gains just aren't there.

I would.
Or I could scrap the filter altogether

Wingfeather · 13th February 2009, 12:41 AM

Quote:

Originally posted by percy
Would you agree that the SRC4192/3 does a lot more messing around with the data then a straightforward 8x oversampling filter ?

That's hard to say without knowing specifically how the resampler block works. If it's simply a zero-order hold (i.e. doesn't actually modify any values), and you're working with the SRC4193 (which has the "Direct Down-Sampler" option) then the ASRC appears to consist only of a single FIR oversampling filter. So the two should be the same. If the resampler is more complex than that (such as a linear/polynomial interpolation et al), then I would agree that it does mess more with the data. If your two clock domains are synchronous and you can thus get away with a pure FIR solution, then that's going to be the cleanest approach.

Quote:

Originally posted by percy
...you mean out of band noise ?

Well no, not really. A lone (dithered) quantiser will produce white quantisation noise. It could be shaped into the stop-band region if a noise-shaping quantiser is used, but on straight FIR filters such as used for oversampling that's not generally the case. There will thus be a white noise floor in the signal coming out of the filter.

With a word length of 32 bits, which is getting more and more common, the noise power from quantisation is of the order of -190dBFS. Having three quantisations (and so three times as much noise power) will degrade this by ~4.77dB - really not a huge problem at this sort of level. So while the quantisation noise is definitely in-band, it's not important provided that enough bits are used.

Wodgy · 13th February 2009, 01:11 AM

There are good details of the SRC4192 implementation in the associated patent. Go to:
http://patft.uspto.gov/netahtml/PTO/search-bool.html
and search for patent 7,262,716. The patent itself contains references to prior patents describing the techniques used in the AD1896, etc. and some discussion of differences.

I have a decent background in signal processing but I'd have to sit down for an afternoon to answer the original poster's question with any degree of certainty. On first impression, however, the results should be similar. Given that the SRC4192 has jitter rejection issues (whereas the SRC4392 doesn't), I don't see any point in using the SRC4192 just for its own sake, i.e. if you don't need ASRC. The SRC4392 is a good option, but you need a microcontroller.

10th February 2009, 04:44 PM	#1
percy diyAudio Member Join Date: Apr 2004 Location: MN	ASRC and Oversampling Filters Is there really any difference between ASRCs (like TI SRC4193/4) and Oversampling Digital Filters (like NPC SM5842/7) in the way they do integer* (4x,8x) oversampling* ? I suppose it would be possible to set a integer Fsout/Fsin ratio in an ASRC ? And in that case, is there any difference in the way it accomplishes the oversampled output, as compared to a filter that is a "4x/8x oversampling digital filter" ? One more question about OS DFs - Typically they utilize a stepped approach to oversample, so if you wanted 8x oversampling they would first oversample to 2x, then they would oversample that result into 4x and then repeat the same exercise to get 8x. Basically use the result of the previous step as an input to the next step. Now I can understand why they might do that. They might want to make it flexible so the user can use any ratio they want, and it also seems like this method is less demanding in terms of processing power. BUT, my question is - if there was ample processing power at your disposal, and 8x was the only ratio desired, then would the results(distortion, noise) of doing 8x oversampling at one shot be any different than doing the oversampling in three steps ?

11th February 2009, 11:33 PM	#4
Wingfeather diyAudio Member Join Date: Mar 2003 Location: Southwest, UK / York, UK / Edinburgh, UK	Hi percy, From the datasheet, the SRC4192/3 is a conventional 16x-oversampler, followed by a resampler and then a conventional (adjustable?) decimator. It doesn't say in the datasheet quite how the resampler works, but I would imagine it to be based on some sort of polynomial interpolation process. As for the question of which method is best, it's really hard to say. I designed a few filters quickly in MATLAB (one 600-tap 8x oversampling filter and 3 polyphase 2x filters (150-tap, 20-tap and 16-tap) for cascading). I was pretty quick in the designs, but aimed for minimal passband ripple, minimal attenuation at 20kHz and 120-dB stopband attenuation. Overall, the responses of the two approaches in both frequency- and time-domain looked very similar. I would suggest that the 8x filter is somewhat less fiddly to design, but then you're looking at 600 MACs/sample as opposed to about 90 MACs/sample for the cascaded approach. Not really worth the simplicity on that front To be practical, the cascaded approach would need a quantisation after each filter, as opposed to one single quantisation after the 8x filter, so there'd be 3 times as much quantisation noise in the cascaded version. If you use long enough word-lengths to begin with though, and dither at each truncation (see my many rants on dither on this forum if you don't know about it), then this is a non-issue. If I were designing an oversampler, I'd not really be able to justify to myself the extra expense of the single-filter solution, even if I had the computing power. The gains just aren't there. __________________ Wingfeather

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11th February 2009, 04:47 PM	#2
percy diyAudio Member Join Date: Apr 2004 Location: MN	come on, I know there are DSP geeks and gurus around here who can easily answer my second question (about doing 8x v/s doing 2x + 2x + 2x).

12th February 2009, 12:15 AM	#5
banana diyAudio Member Join Date: Feb 2003 Location: Hong Kong	The plain old AD1890 datasheet have some explanation on the principle of ASRC. Link to AD1890 datasheet

13th February 2009, 01:11 AM	#10
Wodgy diyAudio Member Join Date: Jul 2003	There are good details of the SRC4192 implementation in the associated patent. Go to: http://patft.uspto.gov/netahtml/PTO/search-bool.html and search for patent 7,262,716. The patent itself contains references to prior patents describing the techniques used in the AD1896, etc. and some discussion of differences. I have a decent background in signal processing but I'd have to sit down for an afternoon to answer the original poster's question with any degree of certainty. On first impression, however, the results should be similar. Given that the SRC4192 has jitter rejection issues (whereas the SRC4392 doesn't), I don't see any point in using the SRC4192 just for its own sake, i.e. if you don't need ASRC. The SRC4392 is a good option, but you need a microcontroller.