Upsampling. Truth vs Marketing


Has anyone done a blind AB test of the up sampling capabilities of a player? If so what was the result?

The reason why I ask because all the players and converters that do support up sampling are going to 192 from 44.1. And that is just plane wrong.

This would add huge amount of interpolation errors to the conversion. And should sound like crap, compared.
I understand why MFG don't go the logical 176.4khz, because once again they would have to write more software.

All and all I would like to hear from users who think their player sounds better playing Redbook (44.1) up sampled to 192. I have never come across a sample rate converter chip that does this well sonically and if one exist, then it is truly a silver bullet, then again....44.1 should only be up sample to 88.2 or 176.4 unless you can first go to many GHz and then down sample it 192, even then you will have interpolation errors.
izsakmixer
Hmmm... I'm surprised that nobody jumped all over me for stating the obvious. That is, digital is a poor replication of what is originally an analogue source.

I'm also glad to see that nobody contradicts the fact that having more sampling points can only improve the linearity of a system which is less than linear to begin with. After all, if digital was linear, we could linearly reproduce standardized test tones. The fact that we can't do that, at least not as of yet with current standards, would only lead one to believe that analogue is still a more accurate means of reproducing even more complex waveforms.

Converting analogue to digital back to analogue again only lends itself to potential signal degradation and a loss of information. One would think that by sampling as much of the data as possible ( via upsampling above the normal sampling rate ), that one would have the greatest chances for better performance with a reduction the amount of non-linearities that already exist in the format. Evidently, there are those that see things differently. Sean
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Sean, thank you for all the diagrams, and patient tutoring. It all makes logical sense, sure.

If I had never heard the Audio Note, I would be looking at one of the top up sampling players on the market. Oh well.
More corrections! They don't affect the basic idea, but could easily confuse people. Sorry about that. Hopefully this is it.

If the waveform is sampled at a frequency four times that which corresponds to the uniform time spacing of the example, (44.1 KHz perhaps) the data will look like the following:

Note also that we are now quantized at 1/4, (0+0+0+1)/4 ,instead of 1, which is the quantization of the raw data stream obtained from the disc. A factor of 4. That’s like 2 bits of additional resolution. That’s how Phillips got 16 bit performance from a 14 bit D/A.

OK Sean...Sorry you felt left out because no one jumped all over you. The following is my modification of your statement.

Some digital representations of analog (analogue in England) waveforms are a poor replication of the analog source because they lack the resolution (bits) and sampling rate appropriate for the bandwidth of the signal. Inaccuracy is not inherent to the digital format, but represents a design decision regarding what level of error is acceptable.
As long as human beings are analog, the initial & final music will always be analog. What's in between can be digital. Digital is a compromise for an analog signal - no doubt. How good or bad it is depends on how well the digital system is engineered for the 20Hz-20KHz bandwidth. Digital is chosen mostly for its cost effectiveness (scalability of the DSP engines with shrinking CMOS technology) & what Carver Mead once pointed out - its tremendous noise immunity. Corrupting a stream of digital data to the point of making it useless is very difficult as it requires a lot energy to flip a bit. Some bits do get flipped but the overall context of the message is very much retrievable by using various error correction algorithms. This is hardly the case with a purely analog music signal.

Having more sampling points with an estimation filter allows the digital to better track the analog waveform. Whatever benefits one accured with over/upsampling could be lost by distortions in the analog reconstruction filter. Hence, the above mention of implementation. Having somebody engineer a good re-produced sound CDP solution is priceless (for everything else there's MasterCard!).

Yes, if one converts from analog->digital->analog, one does degrade the original sound. That is to be expected as we take only a finite # of samples (hence the term "quantization"). BTW, if we had infinite # of samples, it would analog! In the redbook CD format, the powers-that-were decided in all their infinite wisdom to Nyquist sample the data onto the CD disc . Thus, no matter how much one oversamples, one can never undo this. Hence the rise of "hi-res" music formats. In fact, if the over/upsampling was A1 perfect, you'd get *exactly* what was on the CD, which is Nyquist sampled!! How good is taking just 2 samples of a dynamically waveform music signal? Not very good I'm afraid!
Eldartford cited his experience: 4 samples was worth every effort. I've found 5-8 samples is worth the effort. The difference is that my work is voice-related. Not hi-res by any standards but when people hear another voice at the other end, they do want to recognize it. Need more samples for this.
FWIW. IMHO.
El: Thanks for correcting my previous errors, your previous errors and then confirming my last statements.

The bottom line is that, as good as digital is and can be, it is still trailing behind analogue as we know both formats today. It is too bad that the decisions foisted upon the audio industry when selecting these design parameters were made by those that don't really listen to the products that they produce. Otherwise, we would have started off with wider bandwidth designs and higher sampling rates to begin with, making conversations like this moot. Then again, hind-sight is almost always 20/20 : ) Sean
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