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In Reply to: I'm talking about SSRC not oversampling.... posted by Ted Smith on February 24, 2007 at 18:45:30:
As long as the output sample rate is higher than the input sample rate.You can call a conversion from 44.1 and 88.2 "sample rate conversion", "oversampling", or "upsampling". But if "zero-stuffing" is done in any such conversion instead of interpolation, it's just not going to sound right.
Follow Ups:
HowdyYou need to read and understand a basic digital signal processing book or take a class in the subject.
The interpolation you talk about is the digital filter I'm talking about.
The process of zero stuffing followed by (digital) filtering at 1/2 the input clock freq is exactly the correct way to raise sample rates by an integer.
The process of filtering at 1/2 the output clock rate and throwing out all but every Nth sample is exactly the correct way to lower sample rates by an integer N.
The process of zero stuffing followed by (digital) filtering at 1/2 the lowest of the input and output clock freqs and then throwing out all but every Nth sample is exactly the correct way to synchronously sample rate convert by an integer ratio.
If this isn't clear I'm sorry but I'm dumbfounded in that I don't know where to start to make it any clearer.
The problem is, Ted, is people mention the zero-stuffing part, and not the interpolation part. And then equate oversampling to zero-stuffing with no further process.Personally, what should be omitted is the "zero-stuffing". In the whole process, this part is trivial. (I personally would have called it "new placeholders" for the interpolated data.) The "interpolation" is the important part of oversampling. And it shouldn't be treated as separate from oversampling. Because it cannot even be executed without oversampling.
In fact, Ted, it's been used by some CD player and DAC manufacturers to dupe customers into thinking oversampling is "zero-stuffing" and interpolation is an "innovation" by the manufacturer!! Most people don't even realize it's deceptive advertising.
And I'm sure there have been quite a few products brought to market based on misconceptions about oversampling. I think the number of products that mangle the signal to be alarmingly high.
And then audiophiles complain about the sound, and engineers complain about audiophools criticizing what they think are products of superior fidelity.
And in practice, I'm not sure if the zero-stuffing part even takes place!! The oversample values may just as well be overwritten once the new data replaces the old. Had the concept been stated that way, there wouldn't have been so much confusion.
HowdyManufacturers aren't as technically illiterate as you make them out to be. Some (dCS for example) chose to use interpolation strategies that aren't based on filtering (and hence aren't mathematically optimal) they may be better by some other criteria (perhaps the designer's ear) and they are certainly less math. Most designers put their mark in by making different choices of how to implement the filter, e.g. FIR vs. IIR, minimum ripple or minimum phase...
Of course zero stuffing takes place in SSRC, you simply avoid multiplying by the samples you know are zero.
Like I said before, you should really implement these things for your self so you get a feel for them. If you can't implement a FIR filter with the desired impulse response or filter shape then you shouldn't be speculating about how these kind of things work.
I think misconceptions of classic oversampling played a major role in asynchronous conversion getting off the ground.ASRC, in my humble opinion, is a process that should have never found its way into consumer digital audio playback products. It was originally intended to convert data for transfer from one media format to another, not necessarily in real time.
But technically-corrupt products do make it to market when sound concepts are mis-characterized. And very few concepts have been a bigger victim of this than classic oversampling/digital filtering. The "old fashioned" 8x oversampling IMO is still the best method of digital filtering that we have. Where the innovation lies is in the FIR filter function, which is still the one part of CD playback design that I think has room for vast improvement.
HowdyYou continue to speculate about cause and effect of things you have no personal knowledge about.
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HowdyYou should have heard jj in person about some of the people here :)
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I found it hilarious last year when he got so incensed he started accusing me of things that Todd had said :-)I will admit I baited him a little, but I didn't expect him to swallow the bait, hook, line and sinker and then choke on it :-(
Check out Figure 4c and accompanying text on page 3 on the following link.
This is the first time I've seen "fuzzy" information on a chip data sheet, that could be interpreted totally wrong. Looking at the age of the data sheet, I think it was more an oversight than a deception.It deals with the paragraphs and Figure 4. Starting at "To the Rescue"....
Figure 4 shows how oversampling filters work in both the time and frequency domains. We start with a sampled signal at 44.1 kHz (Figure 4a), which has images in the frequency domain (Figure 4b). The next step in the process is to increase the sample rate of the digital signal by inserting zero-valued samples (Figure 4c), resulting in the spectrum shown in Figure 4d.
The root of this whole problem deals with the "analog traces" superimposed with the digitized signals in Figure 4. Projecting the impression that "zero stuffing" would have no impact on the signal, at that particular stage, if it were to be converted to analog. Technically Figure 4c does show the "spikes" with a "zero signal," as I stated earlier. But the "analog waveform" pictured in Figure 4c does *not* depict what the waveform would look like if it were to be converted to analog at that particular state. It's just a representation of the **input** analog signal, to provide reference.
But instead of providing reference, the analog traces in Figure 4 provided confusion.
The one picture that is factually incorrect is Figure 4d. It would *not* be identical to Figure 4b. (Along with the analog trace, this induces the misconception.) There should be "much taller" spikes present at the sample frequency and all harmonics. (44.1 kHz, 88.2 kHz, etc.) Because the signal has been converted to a pulse train whose fundamental frequency is the base sample rate. (If I were to re-write this white paper, I'd remove Figure 4d altogether.)
The low pass interpolation filter, mentioned afterwards, is where the digital filtering actually takes place. The "zero values" does not filter anything. It's just a pre-stage for the digital filter. The final reason for the misconception is this is never mentioned explicitly.
The sad part is this ambiguity has caused a big misconception. Zero-stuffing being what solely occurs with oversampling. Which has gone as far as being accepted fact across the audio engineering community. (I've even heard this misconception applied to digital radio applications at a place I used to work at.)
And it may have even compromised or corrupted some designs along the way.
But at the time this white paper came out, I don't know if anybody would have known enough about oversampling (aside from those within the chip manufacturers themselves) to have caught this.
And since this has evolved into accepted misconception, the only way to realize it's a misconception is to literally "field strip" the individual elements that make up the oversampling/digital filtering process.
*** Zero-stuffing being what solely occurs with oversampling. Which has gone as far as being accepted fact across the audio engineering community. ***I don't think anyone is saying that. Not Ted, not the paper, and certainly not myself.
You will note I said "zero-stuffing followed by a digital filter works", and NOT "zero-stuffing (by itself) works"
*** And it may have even compromised or corrupted some designs along the way. ***
"Can you give some actual examples?"The fact ASRC is even present in digital playback is the prime example.
And the sheer denial without explanation, going on all the time. It's maddening. Call me whacked all you want.
I'm putting stuff up on public record, and putting my real name on the line. I'd be thrilled to one day have all of you in the room, with nothing more than a huge white board. People are treating this like it's rocket science, and it need not be.
There was once a famous line- "If you cannot dazzle 'em with brilliance, baffle 'em with bull...." The statement holds so true, people can no longer distinguish one from the other. And that includes a lot of audio designers.
The bottom line is that for 20 years, the sound of digital mostly sucked. And it sucks now. And too many people would rather blame the end user for their delusions of "golden ears" than come to the reality that designs are not nearly as good as they've claimed.
Too much blind belief and consensus, not enough investigation. This is at the very core of why I think the audio industry is corrupt.
I'm not a big fan of ASRC myself.And I'm sympathetic to the view that potentially some designs are making it a lot more complicated than it needs to be.
My personal view is that digital "by the book" can sound pretty good (ie. an "application note" design using a flagship DAC without any fancy frills, or reclocking, or up/over-sampling, or weird output stages). Provided of course you can feed it a reasonably jitter free signal. Problem is achieving that "reasonably jitter free signal" :-)
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