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
I agree with Megasam, specs dont tell you anything about the sound. I cant really comment on the upsampling question, but my experience with amplifiers tells me you dont always get what you would be led to expect. That goes for power ratings, type of technology, price or age. My Totem Ones taught me that numbers can decieve. When a 25 watt ss amp drives a speaker better than a 100 watt ss amp of fairly good quality, you know the only way to tell is by listening. Buy the stuff that sounds good.
Check out what Audio Research has to say about it with their CD3 III machine.
Get an 8.5 x 11 piece of paper and a pencil.

Now draw a straight line horizontally across the page about 1/4 of the way down. Do the same think half-way down the page and then another horizontal line at 3/4 of the way down the page. The top line will be called "A" and the bottom line will be called "B". The center line is simply a dividing line between A & B.

The A & B lines represent zero in these two different lay-outs. The space above the zero line in each case is positive and the space below each zero line is negative. In effect, the upper half of the page has a positive section, the zero line for section A and a negative section. Then you have the dividing line in the middle of the page separating A from B. Below that, we have the positive section of B, the zero line and the negative section of B.

Now start at the left side of the page and draw a series of 20 random "dots" above and below the "zero line" in section A. For demonstration purposes, place 10 dots above and 10 dots below the zero line. Put your first dot in the positive section, move down in into the negative section and place your second dot there. When doing this, start moving gradually from the left side of the page over towards the right. Vary the distance of each dot above and below the zero line and spread them across the entire width of the page.

In other words, some dots should be closer to the zero line, others should be spaced further away from it, etc... Some dots will be closer together and some will be spaced further apart. We are looking for strictly random spacing here, both in terms of above and below the zero line and from left to right.

When you've got all 20 dots filled in on the positive and negative sections of part A, duplicate the placement of those dots down in section B. In other words, the dots in section B should be identical to those in section A, both in terms of the heights above and below the zero line and their left to right spacing.

Now that you have that done, we are going to stay in section B. What you need to do now is to add another 20 dots to what you have in section B. Same rules apply i.e. random vertical and horizontal spacing with ten more above zero in the positive range and ten more below zero in the negative range.

When that is done, set your pencil down and look at the A dots and then look at the B dots. Each dot represents a sampling point of a bunch of musical notes. In case you haven't figured it out, A represents "standard" CD playback and B represents "upsampling".

Pick up the pencil and start in section A. We are going to play "connect the dots" here. Working from left to right, draw a line from zero up to the first positive dot. From that first positive dot, draw a line down to the first negative dot. From that first negative dot, draw a line to the second positive dot. From there, down to the second negative dot. Alternate between positive and negative going from left to right until you've got all twenty dots connected with one jagged line.

Now go down and do section B the same way, but this time, start at zero and draw a single line up to TWO positive dots, then down to the two negative dots, back up to two positive dots and then down to two negative dots, etc... Do this just like you did with section A, working left to right. When you're done, you'll have all fourty dots connected with one jagged line.

Now compare the two "waveforms" that you've just drawn. Section A is a standard CD waveform and section B is an upsampled waveform. As you can see, even though you started off with the same basic amount of data in both sections A & B ( 20 sampling points ), adding the additional data via upsampling ( twice the sampling rate hence twice the data ) drastically alters the waveform and response. Not only are some of the changes in polarity from positive to negative not as abrupt, there's a lot more contouring taking place to fill in the gaps between sample points in section B.

While upsampling hasn't "created more notes" to fill in those gaps, it simply allows us to follow what was already there more accurately. Not only are some of the amplitudes different, but the contours ( rise and fall times ) of each note can be seen more accurately.

By following those contours more precisely, we limit overshoot and ringing. The reduction of overshoot results in a more natural sound albeit less "artificially hyped" in the dynamics department. At the same time, we can also hear how notes decay a little longer, rather than just hitting and fading to the next note so rapidly. This adds a more "lush" albeit "slower" presentation to what we are hearing, much like the natural decay of a plucked string on an upgright bass or cello.

These "extra" samples as seen in the upsampling section of B should not be confused with "oversampling". Oversampling simply looks at the same points on section A and confirms that each reading is in line with the last reading taken. There's no more data recovered, it just keeps verifying that the limited data that it has retrieved in those samples is correct and consistent. It does this repeated sampling of the same points 4x, 8x, 16x, etc... If the samples don't jive, error correction can kick in as needed. The more error correction that kicks in, the more "self interpretation" of the data that the machine itself has to do.

Other than that, the majority of what makes the most audible differences in players is the type and quantity of filtering used and how it is implimented into the circuit. Audio Note eschews much of the filtering and gets rid of the oversampling, which reduces a LOT of the in-band noise and distortion that lesser designs introduce. At the same time, it can introduce out of band noise and distortion into the equation, which isn't good either.

Obviously, the key is to find a way to increase the sampling rates to recover more of the data AND do so in conjunction with well thought out filtering and error correction circuitry. In doing so, we should end up with more of the benefits with less of the drawbacks. Sean
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PS.... This is kinda - sorta the "quick and dirty" explanation of upsampling. It may not be perfect on all counts in terms of technical accuracy, but you should get the basic idea of what's going on and why it has the potential to be a superior performer.
Sean,
basically your "kinda - sorta the "quick and dirty" explanation" is pretty good for the layman.
I have objections on some of the text you posted:
* Audio Note eschews much of the filtering and gets rid of the oversampling, which reduces a LOT of the in-band noise and distortion
How does getting rid of oversampling & eschewing much of the filtering reduce IN-BAND noise & distortion???
AFAIK, anything in-band cannot be touched. It's sacred as it's THE signal we are looking for. If noise exists in-band or if distortion exists in-band, you basically have to live w/ it OR design better electronics. What you wrote will not do the trick.

* At the same time, it can introduce out of band noise and distortion into the equation
what are you referring to here? i.e. when you write "it can introduce....", what is "it"??

* Obviously, the key is to find a way to increase the sampling rates to recover more of the data
Increasing the sampling rate does NOT recover more data. It, however, allows the discrete-time system to follow the original analog data more truthfully. This is evident from your section A, section B example.

On a historical note, Philips is the co. that is to be credited or discredited with the concept of upsampling. The original idea at Philips Reasearch Labs was to somehow get that analog filter order lower & that transition band less steep. In the original redbook spec, the transition band is 20KHz-22.05KHz. Upsampling was the answer from an engineering perspective & from a cost prespective. They really didn't care about the sonic effects back then.
FWIW. IMHO.

Bombay: Your own description answers the problems that you questioned i.e. "On a historical note, Philips is the co. that is to be credited or discredited with the concept of upsampling. The original idea at Philips Reasearch Labs was to somehow get that analog filter order lower & that transition band less steep. In the original redbook spec, the transition band is 20KHz-22.05KHz. Upsampling was the answer from an engineering perspective & from a cost prespective. They really didn't care about the sonic effects back then."

By playing games with the actual cut-off frequency and Q of the filtering OR by removing the majority of filtering, you reduce the amount of roll-off, phase shift and distortion in the treble region. As far as oversampling and error correction goes, that simply equates to more tampering that the machine itself is doing with the signal and / or noise that it is generating within the power supply and support circuitry.

In effect, error correction is "somewhat" like negative feedback. As such, Audio Note feels that small errors aren't as much of a negative as the problems that result from trying to correct them. Between the lack of oversampling and their approach to filtering, many people seem to agree with the sonic results that they've achieved. As a side note, Moncrieff covered error correction in IAR many years ago. Sean
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