Secrets Q & A

Vinyl vs. CD - A Running Commentary - Parts 6 - 9

ARTICLE INDEX

When attending CES and Rocky Mountain Audio Fest over the past couple of years, I noticed that turntables are starting to be the majority of sources for exhibit rooms. We all have heard about the vinyl renaissance, but I just did not realize how strong it is. I mean, LPs are being played everywhere at hi-fi shows. So, we decided to compare Vinyl LPs to CDs in terms of the recording process, manufacturing, and the sound. Parts 1-5 (Part I: Introduction, Part 2: The Technology, Part 3: Turntables, Tonearms, and Cartridges, Part 4: The RIAA Curve, and Part 5: Setting up the Turntable) are published in a previous article. Here are Parts 6-9.

Part 6: In the Groove

The frequency response of an LP is about 10 Hz to 25 kHz, and it has a dynamic range of 75 dB. All the information is contained within a groove about 0.002" in width.

This groove is originally cut in a lacquer-coated aluminum disc using a cutting head with a sapphire stylus about 0.0002" in diameter (10 times smaller than the groove that is cut). Sapphire (a.k.a, corundum, which is the crystalline form of aluminum oxide) is actually better than diamond for cutting the original master, although diamond is better at playing the LPs at home. The cutting head is driven by stereo power amplifiers of about 400 watts each. So, cutting the master is the opposite of playing the recording once it is on the platter that you purchase. The diamond stylus that you play the records with is about 0.001" across.

A negative replica is made from the master, and then vinyl biscuits are melted on the negative, cooled, trimmed, and packaged for the consumer.

Why 33-1/3 RPM?

When the technology moved forward from cylinders to discs, it was calculated by Emil Berliner (inventor of the Gramophone) that 78 RPM would reproduce the frequency response using a steel needle. The 33-1/3 RPM speed was developed because the disc used for sound in early movies would hold 11 minutes of the soundtrack, which was the length of one movie reel. Then CBS applied that rotational speed for commercial music recordings and we had the LP with 30 minutes of music on each side of the disc.

The 45 RPM disc may seem to have been just for the popular music crowd, with two hit songs on each disc, but it was actually a technological advancement in sound quality as well, and in fact, current re-releases of LPs are sometimes in the 45 RPM format, with the music that was originally on two sides of one disc, now on two sides of two discs. I obtained a few of these and will discuss them later in comparison to the more conventional 33-1/3 RPM discs.

The LP was introduced in 1948, and stereo LPs soon followed. They were manufactured from polyvinyl chloride, which was sure a big improvement over the older 78's made from shellac (much lower surface noise). You could drop them without breaking them, but they were (are) just as sensitive to scratching, and of course, they didn't (don't) do well if left in the sun (they warp).

So, let's analyze the groove's capability.

The stylus tip follows the groove at 33-1/3 RPM, which means a linear velocity. The problem (Problem 1) is that the linear velocity changes, because the diameter of the spiral is smaller at the center of the disc than at the outer edge. This is one reason some high-end LP re-releases are at 45 RPM. With the smaller spiral diameter, the high frequencies are crowded into a smaller linear space, and they suffer.

The groove itself is like a "V", with the right channel being represented by sharp peaks and valleys on the outer edge of the V, and the left channel on the inner edge. So, the stylus, following the groove, is raised on the peaks, and lowered in the valleys. The stylus is connected to the cantilever, and the cantilever has either the coils (MC) or magnets (MM) attached to it, which moves, and induces electrical current in the coils, which is translated into the music through the amplifiers and speakers.

Stereo grooves could have used a lateral motion for one channel and vertical motion for the other, but it would have resulted in less compatibility with monophonic cartridges, and also, the lateral motion channel would have had lower fidelity than the vertical channel.

Problem 2: The stylus is attempting to translate both channels at the same time, and because the movement of the stylus shares some common direction (the stylus moves downward for both left and right channels), there is some bleeding of each channel into the other, called "crosstalk". In fact, channel separation is "only" about 30 dB. In other words, if you had a groove where there was a 1 kHz sine wave recorded only in the left channel, and you played it back at 100 dB from the left speaker, you would also hear that sine wave in the right channel at 70 dB.

That number doesn't sound very good at first glance, but with each 10 dB increase, it sounds (subjectively) twice as loud, and with each 10 dB decrease, it sounds 1/2 as loud. So, going from 100 dB to 90 dB cuts the loudness by 1/2, then to 80 dB is another 50% reduction, then to 70 dB is an additional 50% softer, for a total of 8 times softer at 70 dB. Nevertheless, there is substantial crosstalk with the stereo phonograph recording. CDs, by comparison, have no crosstalk generated by the medium itself. The two tracks are discrete. Any crosstalk is generated by the downstream electronics after the signal is converted from digital to analog. However, that crosstalk is still much less than the crosstalk from tracking an LP groove.

How do you like the problems with LPs so far? Here's another one. Problem 3: Because the translation of the sound from the groove to the stylus requires that the stylus be falling into the valleys or pushed up with the peaks in the groove, having a out of phase information in the left vs. right channel causes some issues with the laws of physics. In other words, if there is a valley in the left channel at the exact same time there is a peak in the right channel, the stylus finds it a little difficult to be going in opposite directions at the same time. The result? Inaccurate signal reproduction at that instant in time. So, LPs cannot handle a lot of material that is out of phase between the left and right channels.

There were some technology improvements made during the 1980's, such as the method of disc cutting (making the master disc that would be used to produce the vinyl copies that we purchased). Although the CD began to eclipse the LP in that same decade, we can certainly benefit from the new LP cutting technology that was developed at that time, and which has continued to improve, by going back and remastering (recutting) early classic recordings. Such remasterings and re-issues are now popping up all over the place. Some rock groups, whose musicians were only babies when the CD was introduced, insist in their contracts that their albums be released both as CDs and LPs.

Why, then, if there are these obviously serious problems, do so many people feel the LP sounds "better" than the CD?

Dynamics may be one reason. Along with digital music reproduction, came the ability to mix and edit the music digitally, and this offered incredible flexibility. Boom boxes proliferated, and these boxes didn't have much in the way of dynamic capability, so CD producers compressed the dynamic range, boosting the quiet parts of the music and attenuating the level of the loud passages. That way, consumers could hear every part of the music, with it all having about the same level of loudness. Although this worked for boom boxes, the music sounded terrible on good audio systems.

Not all companies compressed the dynamics on their CD releases. Telarc is notable here, producing early CDs that, even today, are just marvelous.

In my tests, comparing an LP version of a recording to its CD or SACD counterpart, I have not found dynamics to be an issue at all. In fact, the CD and SACD have the potential to record a higher dynamic range than the LP (LP has a dynamic range of 75 dB, CD has 96 dB, SACD has 120 dB, DVD-A has 144 dB). The problem is that CDs and SACDs have not truly exploited that potential.

Some have said that it is the extended frequency response that LPs have, but SACD has a very high frequency response, at least as good as LP. However, because of filters that are used in playing back digital recordings, there is phase shift that causes problems in the audible high frequency range, and my initial impression is that it is the high frequency area of an LP that has better definition than its digital counterpart.

Well, if the idea is that analog all the way gives the best sound, then why not just go back to playing analog recordings on cassette tapes? The problem there is that the tape speed is so low, the dynamic range is only about 60 dB. Plus, you have tape wear, tape hiss, and the inconvenience of having to rewind or fast foward to get to a particular song.

 


Part 7: THD+N Test Results

OK, so now that we have all the preliminary discussion in place, let's do some measuring.

For these initial tests, I used a McIntosh MCD201 SACD/CD player, McIntosh MT10 Turntable with factory cartridge (made by Clearaudio), and a Manley Steelhead phono stage. I connected the analog output of the MCD201 and the Manley Steelhead directly to our Audio Precision SYS-2122 analytical instrument. I used a CD with test tones recorded at - 5dB and a test LP called the Ultimate Analogue Test LP.

vinyl-vs-cd-ultimate-analogue-test-lp-cover-art.jpg

To start, I used 1 kHz sine waves.

Here is a graph of the spectrum generated from the test CD. THD+N was 0.005%. That is very low and very good. Notice that the one distortion peak visible is third order.

At 10 kHz, distortion rose to 0.01%. Again, the one visible peak is third order.

And . . . (drum roll) . . . here is what you have been waiting for: The Vinyl Results.

At 1 kHz, 0 dB, distortion was 7%. "Wow, that's a lot of distortion," you say. You bet it is, but notice that it is nearly all second order. Also, the noise level contributes to this high number. At best, the noise is 70 dB below the signal, whereas with the CD test, it is more than 100 dB lower. We could measure just the THD, but much of the harmonics are buried in the noise, so this would not be a fair estimation of the resulting sound.

For this particular graph, I set the "load" for the cartridge at 25 ohms.

Here is the same test, but with the cartridge load set to 400 ohms. THD+N went up a bit, but in this case, I think the distortion peaks were a bit higher, while the noise floor improved. This is why seeing the spectrum is so important. The numbers indicate the overall sum of everything, while the graph shows you where the numbers are coming from. Notice that the primary distortion peak is second order.

For 10 kHz, I kept the 400 ohm cartridge load. THD+N was a very high 20%, but again, the noise level contributes significantly to the number. Notice that the only visible distortion peak is second order.

So, what can we say at this point? It seems to me - and this is with some of the very best hi-fi components in the world - that one of the most significant reasons vinyl aficionados love the LP sound is that the distortion is very, very much like that in Pure Class A triode single-ended tube amplifiers. There is a lot of distortion, but it is virtually all second order (even-ordered), which is euphonic, meaning that it is pleasing to the ears. Now, keep in mind that the distortion seen here for the CD is from a superb CD player. If you look at various lower cost CD player reviews out there, you will see more higher order distortion peaks, with odd-ordered peaks closer in level to the even-ordered peaks. Bottom line is that CD has much less distortion, but more of it is odd-ordered, while vinyl has more distortion, but it is euphonic second order.

 


Part 8: Phono Preamplifiers

Now it is time to talk about a very important part of the signal path in playing LPs: The Phono Preamplifier (also called Phono Stage).

For our first look into the phono preamp, I chose three: Musical Surroundings Phonomena, Bryston BP-1.5, and Manley Steelhead, all of which are reported on here. Future cartridge, phono preamp, and turntable reviews will be published in our regular product review section.

Let's start with the Phonomena (the lowest priced of the three) which is manufactured and marketed by Musical Surroundings, located in Northern California.

At $999, this appears to be a bargain, because it has a lot of features including:

♦ Built-in automatically charging NiMH battery pack that powers the preamplifier and which disconnects from the AC supply whenever it is in operation.

♦ Adjustable gain - in 13 steps - from 40 dB to 60 dB.

♦ Switch for MM vs. MC cartridges (MM cartridges are voltage sources while MC cartridges are current sources).

♦ Adjustable resistive loading (resistance) - in 17 steps - from 30 Ohms to 100 kOhms.

♦ 300 pF (pico-Farads) capacitive loading switch, either on (300 pico-Farads) or off (200 pico-Farads).

Here is a photo of the Phonomena's front panel.

vinyl-vs-cd-phenomena-phono-preamp-front-main.jpg

There is no on/off switch, as the Phonomena is meant to stay connected to the wall-wart power supply which keeps the battery pack charging at all times except when you are using it. The status window glows flashing red when it is charging, steady red when it is connected to the wall-wart while you are using it (a situation where the battery pack was not fully charged when you need to use it), and green when it is in use (internal relays disconnect the preamp from the wall-wart charger, you don't have to physically unplug it).

The rear panel has all the necessary jacks and switches (click on the photo to see a larger version). To turn it on, you simply touch the red status window, and it will change to green. You will hear the relays inside disconnecting from the wall-wart power.

Besides the RCA input and output jacks, each channel also has its own set of dip switches to select the gain, resistance, and capacitive load. The jack for the wall-wart cable is on the right.

 

So, when you turn the Phonomena on, there is only low voltage DC anywhere in the circuit. Because phono preamps have to deal with very, very low level signals (0.3 mV up to 5 mV), this is important.

I tested the Phonomena with a VPI HR-X turntable (we recently added this turntable so we can compare cartridges in future reviews, using the McIntosh MT10 with its superb MC cartridge as a reference), Sumiko Blackbird MC cartridge ($799, 2.5 mV output, 2 gram recommended tracking force, 135 ohm internal impedance, i.e., DCR or DC Resistance of the coils that will drive a 47 kOhm input on an MM phono stage if you didn't have a preamp with an MC input), BAT VK-5i preamplifier, McIntosh MC1201 power amplifiers, and Carver Amazing speakers. Cables were Nordost, Legenburg, and various others.

vinyl-vs-cd-vpi-hrx-turntable-front-main-2.jpg

The Phonomena delivered a detailed, full bodied sound with low noise. The idea of having the phono preamp powered by rechargeable batteries is an excellent one, because there will be no AC interference from components that are connected directly to the AC. This reduces 60 Hz hum. The problems with commenting on phono preamplifiers are that (1) the noise floor from the LP is very high relative to whatever noise the preamp is generating, and (2) there are lots of combination of dip switch settings, and although the cartridge manufacturer lists specifications, the bottom line really depends on your listening preferences.

I changed some of the dip switches and got different sound characteristics, all appealing in one way or another, such as tight or open. So, I started marking each LP with the settings that I liked the most on that particular LP. Accuracy? Not necessarily. Pleasurable? Most definitely.

To explain this a bit further, with a large group of instruments, the high total amount of second order harmonic distortion produced by so many sources could obscure the detail of individual instruments. With a jazz quartet, on the other hand, the harmonics are only coming from a few sources, and this adds a richness to the music, making it sound more "alive"

As you can see below, I have started to accumulate re-issues of classic jazz albums. In fact, what I am finding is that I prefer listening to small groups of instruments, such as jazz trios, quartets, and quintets, on LP, while I prefer CD/SACD with orchestral music.

vinyl-vs-cd-music-lee-morgan-lee-way.jpg

So, I tried different settings, listened, and gathered spectra with our Audio Precision. Here are the results.

Because I had set the Phonomena to a 475 Ohm resistance and capacitance set to "Off" when I used the MT10 with its Clearaudio cartridge, I left the settings that way to begin with the VPI and Sumiko Blackbird cartridge.

Below is shown a spectrum with a 1 kHz sine wave input (using a test LP). The second harmonic is 47 dB below the fundamental. Also note the third and fourth harmonics. The upward slope in bass response is due to the application of the reverse RIAA curve ("De-emphasis") in the phono preamp. This accentuates the low frequency surface noise, unfortunately, as well as delivers the proper curve to the LP recording.

For the distortion measurements, I used the input from a cartridge (Sumiko Blackbird) rather than generating test signals by the 2722 because I wanted an actual cartridge impedance to be placed on the phono preamp. The 2722 was used only for analyzing distortion from the phono preamp output. In the case of frequency response measurements, I used the 2722 to generate the signal (see below) as well as to analyze the output.

I then switched on the capacitive load, and gathered another spectrum, shown below.

What changed? Well, the second harmonic is now a little lower (49 dB below the fundamental), but the third harmonic is now a bit higher, the fourth is lower, and there are visible fifth and sixth harmonics. So, theoretically, the capacitive switch should be left in the off position.

Setting the resistance to 50 kOhms (the impedance of the cartridge is 135 Ohms) yielded the following spectrum. It is more like the 475 Ohm capacitive "On" graph than the "Off" graph. Therefore, I would say that the 475 Ohm load, capacitive "Off" setting would be "technically" the best. Although audible differences were not apparent, I suspect that a consumer spending hours and hours on end listening to one vs. the other settings would eventually decide on one particular setting. Remember that the Off position for capacitive loading is 100 pF, while the On position is 200pF.

Here is the Phonomena frequency response. For this test (on all three phono preamps), I used our Audio Precision to generate the sweep instead of using a sweep signal from an LP and cartridge. I set the input and gain to yield as near to 1 volt output at 1 kHz as possible, then ran the sweep, using a script that compared the preamp's output with the standard RIAA de-emphasis curve. If the preamp's RIAA algorithm corresponded perfectly with the standard curve, the result would be a flat line.

You can see that the Phonomena's response is flat from 10 Hz all the way out to 30 kHz with only a tiny deviation here and there. Overall response is ± 0.1 dB. Superb.

Shown below is the standard RIAA de-emphasis curve, to which the tested preamplifiers were compared. Basically what this means is that when a 10 Hz signal comes into the preamp, it (should) boost it by 20 dB, no boost at 1 kHz, and attenuates it by 20 dB at 20 kHz.

vinyl-vs-cd-riaa-curve-de-emphasis.gif

The Phonomena's RIAA de-emphasis EQ was the flattest response measured in all three phono preamps tested, another reason the Phonomena is a bargain. As you will see though, the others did very well also, and the deviation from the standard curve was miniscule.

In any case, I really liked the Musical Surroundings Phonomena and would recommend it to anyone on a medium budget. We will get some of the truly inexpensive phono preamps ($150 - $300) as soon as possible to make comparisons.

Next on tap is the Bryston BP-1.5.

The Bryston BP-1.5 phono preamp needs an outboard power supply. The good news is that if you have a Bryston BP-26 preamplifier, you already have the power supply that will drive the BP-1.5 phono preamp. In the photo below are the BP-1.5, then underneath that is the BP-26, and on the bottom is the power supply that drives them both. The rear panel actually has several outputs so you can power all sorts of Bryston products. The BP-1.5 is $1,800 USA not including the power supply, so it's more expensive than the Phononema and less expensive than the Manley Steelhead. The photo below shows the BP-1.5 on top, the BP-26 in the middle, and the power supply on the bottom.

vinyl-vs-cd-bryston-bp-1.5-phono-stage.jpg

Here is a photo of the rear panel (click to enlarge it). You can see the dip switches on the right side.

 

The BP-1.5 has variable gain settings from 35 dB up to 63.5 dB, depending on whether you have set the front panel to MM or MC. For MM, capacitive loading ranges from Off to 188 pF. Resistance is not variable.

I listened to LPs (McIntosh MT10, VPI HR-X, Sumiko Blackbird) using the Bryston BP-26 and BP-1.5 combo (the BP-26 is an excellent preamp, as reported previously), but I compared the sound of the BP-1.5 using the same preamplifier (BAT VK-5i) that I used for the other two preamplifiers for consistency in this review.

vinyl-vs-cd-music-art-pepper-rhythm.jpg

The sound was very tight and had very low noise, with plenty of gain when it was needed (the MT10 cartridge only outputs 0.7 mV). By the way, the Art Pepper album shown in the photo is one of the best re-releases on LP that I have yet heard, out of about a dozen albums that I have purchased. This particular one is the 33-1/3 re-issue at $25. If you want it in the 45 RPM version (two LPs), the price is $100. That is a bit steep for me, but apparently there are some LP aficionados out there who are willing and able to shell it out. No question but that it is a terrific album though.

The 1 kHz sine wave test is shown below. Notice that the second and fourth harmonics predominate, with third and fifth almost non-existent. This is excellent performance

Here is the frequency response. The horizontal line is the comparison of the standard RIAA curve with the response of the BP-1.5. It is within ± 0.25 dB, 20 Hz - 20 kHz.

Now to the Manley Steelhead.

I have known EveAnna Manley, chief honcha at Manley Labs for years, and she is really a special person. Let's just say that you will never get bored when she is in the conversation. But more to the point, she manufactures terrific products: preamplifiers, amplifiers, microphones, and other things, including a Phono preamplifier. Her hi-fi components are named after fish, and this one is the Steelhead. In fact the air vents on the top of the chassis are shaped like a fish.

The Steelhead consists of two chassis, one of which is the power supply, and other being the control unit. Here is a photo of the front of the control unit with the power supply on top.

 

On the left are the Input Selector (two MC and one MM), the Gain (50 dB through 65 dB, then capacitive load selector for each channel and resistance selector in the middle. To be more precise, there is an autoformer for the MC inputs that provide impedance load matching for the MC cartridge, while the MM load dial provides purely resistive loading for MM cartridges.

For resistance, you can select up to 400 Ohms for MC (moving coil) inputs and up to 47 kOhms for the MM (moving magnet cartridge) input. Capacitance can be loaded up to 1,100 pF (or none at all). Then, Mute, Dim (turns down the volume for when you change the side of the LP or put on a different LP), Sum (mono), Line (connects an external line-level source), and Sleep (standby - which keeps the "logic" alive) buttons. The Mute light comes on when you first turn it on, and it has to be turned off manually to start playing music. The turn-on mute delay keeps power-on thumps from being amplified downstream.

The rear panel is shown below.

 

The Steelhead has two variable out jacks which are connected to the volume control on the front panel. This is for using the Steelhead to drive power amplifiers rather than going through another preamplifier. If you only are listening to LPs, this would work fine. However, if you want to go through your receiver or two-channel preamp where you also have a CD player, then you should use the fixed level output. Of course, you could do both, with the variable output going to a pair of power amplifiers, and the fixed output going to a separate preamplifier. One last possibility is to use one of the variable output sets to pad the volume of another preamplifier downstream so that the input signal for that second preamp is not too high.

On the other hand, there is also a Line In, so you could connect your CD player to that input, and use the Steelhead as your preamplifier. But, if you have a third source . . . .

OK, I think you get the picture. The Steelhead is probably the most flexible phono stage in the world. It's $7,300 and that is no chump change, but in my opinion, it is worth the hefty price (the Streelhead with its two chassis is a hefty product). It has up to 72 dB gain for the MC inputs, and up to 65 dB for the MM input, which is the highest gain spec of all the phono stages we have tested so far (granted, not very many, but 72 dB of gain is high in any comparison).

vinyl-vs-cd-music-john-coltrane-blue-train.jpg

 

 

 

 

 

 

 

 

 

 

The multi-pin jack in the center of the rear panel is for connection to the separate power supply. You were also probably wondering about the RFI Shunt dip switches. Those are for a situation where you hear a radio station in the background for whatever reason (you might live near a radio station transmitter antenna for example). When you are trying to deal with a signal of less than a millivolt, this can happen.

The audio chassis contains two 6922 tubes and four 7044 tubes. The whole thing is quite heavy, more so than some receivers. Tubes need a lot of power, and analog power supplies weigh a lot more than switching power supplies.

For the tests, I used the VPI HR-X turntable and Sumiko cartridge, as I did with the Phonomena.

First, the Steelhead has more available gain, and therefore, can easily work with low output cartridges, such as the Clearaudio MC in the McIntosh MT10. For high output cartridges like the Sumiko Blackbird, the Steelhead just coasts along, using one of its lower gain stage settings. In this case, I used a mid-level gain setting to bring the output voltage as close to the 2 volt output I was getting with the Phonomena. It ended up being 1.8 volts. Here is the spectrum.

The second-order harmonic is a bit lower, and the third is a bit higher than with the Phonomena.

I did not test the Steelhead with any capacitive loading selected (this is primarily for MM cartridges).

The frequency response is shown below. The RIAA comparison with the standard curve indicates that the overall response is ± 0.25 dB throughout the audible band.

The Steelhead delivered plenty of volume, with clear highs and deep bass. I had no complaints at all. Because the Steelhead lets you add or subtract resistance and capacitive loading (I did not test the Steelhead with capacitive loading because this is primarily for MM cartridges), I adjusted the resistance while the music was playing and found that different settings gave tighter vs. more open sound. You really just have to try the settings out for yourself and decide what you like. In fact, you very well might like different settings for different albums or various types of music. I don't know of any other phono preamp that offers this kind of flexibility.

I noticed that when playing jazz albums cut at 45 RPM, the sound from the Sumiko Blackbird was like the Clearaudio cartridge sounded at 33-1/3 RPM. The disadvantage of having albums at 45 RPM is that it takes two LPs instead of one, and this raises the price substantially ($50 instead of $30).

Frankly, I did not hear definitive differences between the Phonomena, BP-1.5, and Steelhead, but there are so many permutations that are possible with the settings, it is very difficult to compare them. All three sounded so good, I would be happy with any of them, but the output impedance and current drive available in the output stage of the Steelhead (the Steelhead has 350 Volt rails compared to low voltage rails in the Phonomena and BP-1.5, which means higher headroom and output capability) could be useful when you have to drive a difficult load. The large amount of gain capability in the Steelhead is essential if you need to amplify a low output cartridge, e.g., 0.7 mV. So, the Steelhead has the edge. Of course, the Steelhead is more expensive than the other two combined. Another problem in comparing the phono preamps is that the inherent high surface noise obscures the higher order harmonics that might distinguish them.

You can also change the resistance and capacitive loading while you are listening with the Steelhead, while with most other phono preamps, including the Phonomena and BP-1.5, you have to turn the preamp around and flip dip switches on the rear panel. Perhaps when we get some entry level phono preamplifiers, more clear differences will emerge. For now, all three of these products get my official approval.

I could hear more difference between 33-1/3 RPM and 45 RPM than I could between the three phono preamps listed here. It was mainly in the treble. The ride cymbals sounded more natural. Cleaner. More life. More of all the good adjectives. This is because the hills and valleys in the grooves of the LP are farther apart, so the stylus has an easier time tracking the fine detail of the high frequencies. However, 33-1/3 does still sound terrific. No two ways about it. I just wish the 45 RPM LPs were not so expensive. You might complain about $20 CDs. Well, how about $100 LPs ! ! ! ? ? ?

 


Part 9: A Few Bits About DACs

Following the completion of Part 8 of this treatise, the comment section became the largest set of remarks, questions, and hotly contested arguments we have yet seen here at Secrets. It kind of all boils down to how well a needle sliding through a vinyl valley can reproduce the original musical signal compared to a digital bitstream that represents samples taken during the music, and are used to reconstruct the musical signal.

I tested some 10 kHz and 20 kHz sine waves that were recorded at several word lengths (16 bit or 24 bit) and sampling frequencies (44.1 kHz, 96 kHz, 192 kHz), analyzing them in a software sequencer.

Here are the results.

Below is a graph, showing the fine detail of digital recordings made at 16/44, 16/96, 16/192, 24/44, 24/96, and 24/192 using a 10 kHz sine wave. I wanted to vary only the word length for the test, to show that any differences would be a result of the sampling frequency; otherwise there would be a compounding of variables. So, the sampling frequency graphs are shown both for 16 bit and 24 bit word lengths. From left to right, the plot spans 0.001 second, that is 1/1000th of a second. The dots on the graphs represent individual samples, so we are talking about really fine detail analysis here.

The dots represent finite voltage values that are fed in sequence as a stream to the DAC, which then produces a stair-stepped output, after which a low-pass reconstruction filter smooths out the signal. What I want you to notice is how jagged the lines are at standard Redbook CD 16/44. The DAC and reconstruction filter's job is to make these jagged lines more sinusoidal, so that it will be like the music that was recorded, which is also sinusoidal. There are various ways of applying filters, and we won't get into that here, but I was really surprised at how poorly the 16/44 digital stream is representing the original 10 kHz sine wave. Notice that even at 16/96, the lines are not all that smooth. But, at 16/192, the sine wave looks very good. If we were observing a 10 kHz sine wave coming off an LP and displayed on an oscilloscope, it would be essentially sinusoidal, not jagged (it would not be perfectly smooth of course, as there would be at least a little distortion in the signal being played).

Click on the photos to make them larger, then mouse over the enlarged photo and if there is a small box with arrows in the right bottom corner, click it to make the image full size.

Here is a 20 kHz sine wave recorded at 16/44, 24/96, and 24/192. You can see that, even at 24/192, the wave form is not smooth, so a reconstruction filter still has to be applied. Is it any wonder that the arguments about Vinyl vs. CD go on and on?

 

FEATURED COMMENTS

LP sound doesn't always stay great
Written by Pb , August 12, 2008

One factor that I think should be taken into account in comparing CD versus LP is that an LP will be at its best when you first buy it, but will inevitably suffer the effects of exposure to dust/dirt, and some wear to the groove walls. In contrast, unless you are careless about handling CD's and SACD's, they will last indefinitely and will always sound the same as they did when you got them. Thus, LP's may have some nicely "euphonic" distortion when new, but how "euphonic" is that distortion after they have been used for a while?

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Written by JEJ , August 12, 2008

It is important to be careful when handling CDs, DVDs, or LPs, but digital media are certainly more forgiving of scratches than LPs. However, if you handle them only by the edge and use a good cartridge that is properly aligned, and low tracking force, LPs can weather the use over many years.

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Written by JEJ , August 12, 2008

By the way, I had to split this article into two parts because it got so large, the CMS database couldn't handle it. So, we lost all those great comments. If you want to repost what you said, feel free to do so.

  • CLICK TO VIEW MORE COMMENTS

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    Written by MattP , August 28, 2008

    One unfortunate end of this article is that I think many of your readers will walk away telling all those vinyl lovers that they love a flawed format because you "proved" that it has much greater distortion, inferior channel separation, and inferior dynamic range to a good digital format. I wish that reviewers in general would emphasize more strongly what research actually says about the audibility of distortion with regard to order, frequency, and level. Research has not just shown even order distortion to be more pleasing (euphonic), its less audible as well. That makes a metric such as percentage of total a completely useless metric for measuring the audibility of distortion. Distortion, regardless of order, is much less audible in the lower frequencies as well, so one might argue that measuring distortion with the audio band is less important than measuring the distortion within its own audible band.

    The next limitation I see with this article is the way it approaches digital technology. I feel that the article was approached with a bias from the start, a preference toward digital as a technologically superior system, and thus was not as objectively gone through. I would have liked to see more discussion of digitals limitation, especially with regard to format limitations that exist today. Many have argued, even in our precious scientific journals, that the CD and DVD format are not just limited by the use of 44.1 or 48khz sampling rates, and 16 bit or 24 bit data rates, but also by the pcm encoding in the first place. Further arguments talk about the inevitable limitations digital will impose when representing the real world, i.e. an analogue signal. Digital can not, by its own design really, represent an analogue signal with no loss, mathematically or acoustically, it must always vary from the original. The real argument here comes from if the digitally recreated analogue waveform can be audibly indistinguishable from the original. Since all of this is based on listening tests which then use statistical probabilities to see if people can distinguish differences, it has long been argued that this research is flawed. If nothing else, its proved very little, and confused many. Statistics do not allow us to do anything more than reject or fail to reject the null hypothesis, therefor if someone can not hear the difference it does not mean a difference does not exist, or even that a difference is inaudible. It simply means that the difference could not be differentiated under the specific test conditions used.

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    Baseline
    Written by Joe Mudry , August 29, 2008

    Perhaps, it would help to compare apples to apples somewhat. I would suggest the Rolling Stones latest dsd releases. Using these, you could compare the sacd, cd, and vinyl editions, which were all made from the same dsd master.

    We have done it. I won't comment, but I think that you may be surprised by the results. Regards.

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    ...
    Written by JEJ , August 29, 2008

    I really didn't go into the review with a bias about vinyl and digital one way or the other. I wasn't trying to prove that one is better than the other. I just want to find out what the differences are in ways that can be measured with test instruments. Everything that we hear from the speaker can be measured. There is no part that is in the ether somewhere floating around and which cannot be measured. It's all there. So far, what I have found is that I like small ensembles on vinyl, and classical orchestra on CD. I understand that vinyl produces more distortion. But, the bottom line is what one likes, what one prefers, what gives the most enjoyment. Vinyl, in certain cases, gives me more enjoyment than the same music on CD, even though there is more distortion.

    Getting some vinyl that originated from a digital recording is a good idea. It would show what vinyl adds to the recording.

    The current study does not answer all the questions about vinyl vs CD. There are lots of other articles out there on this subject. This is just Secrets' addition to the story.

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    misinformation rules
    Written by Nick P. , August 29, 2008

    PCM *is* continuous.

    At least 20 years ago we saw the graphs in the audio mage explaining PCM and the dots or bars representing samples and today apparently some readers are stuck the implication that digital snapshots means info between them goes missing. That's not how it works. The dots do not get connected by straight lines, nor by stair steps, nor by "French curve" best-fit approximations. The dots get connected by curves that are EXACTLY like the input.

    And while we may know that the audio bandwidth is half the sampling rate, and we probably read something about anti-alias filters on the A/D side and reconstruction filters on the D/A side, do we think we understand how PCM works?

    Allow me to explain the part that's too-often missing and/or misunderstood: By restricting the signal bandwidth on both ends to half the sampling rate, there can be only one curve connecting the dots, and it's the same as the input signal. The curves cannot make sharp turns from one dot to another, nor can they be arbitrary. Low pass filters can be thought of as speed limiters on the signal's frequency and for that matter the shape of the curve. For mathematical proof, search "Shannon-Nyquist".

    As for how well the filters do their job, that's another story and it's one reason early CD didn't live up to expectations. This was overcome a long time ago with better filter design and oversampling. Dither helped along the way. If you've been listening to the same 12 recordings the last 20 years, it might be time for something different.

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    still apples to oranges, part 1
    Written by Nick P. , August 29, 2008

    Regarding the Rolling Stones hybrid SACDs, both layers may sound a lot better than previous releases but they're not a valid choice for DSD/PCM listening comparisons. The difference in sound alone doesn't prove anything except that the sound is different. Assume nothing just because both layers were made from a remastered tape.

    Rather, rip the CD layer to PC like you would any CD and with a wave editor/viewer. You will see clipping on the peaks, like the "volume wars" I talked about earlier but mild by today's standards but still enough to change the sound (the original masters probably went no higher than 15 kHz on a good day, and three clipped samples on the CD can cut through that).

    Now record the SACD-layer analog output to PC (since you can't easily rip SACD but if you can, do that instead) and look at the wave. See any clipping? Nope. So why was the CD layer was transferred a hair too loudly? Don't know but makes me say hmmm... producers know who their audience is.

    (For new readers, "volume wars" clipping is not caused by a limitation of technology, it's the producer's hand.)

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    still apples to oranges, part 2
    Written by Nick P. , August 29, 2008

    Even if you were to use a hybrid SACD with identically-treated layers from a common master for a listening comparison, what's to say that the player is treating them equally? Nothing, unless the player is tested and results are known. The first production Sony SACD player ($5K) sweetened the high end when playing SACD according to Stereophile's published graphs - the upper audio frequencies had a rising response. Aside from that, later SACD players had to use a 6th order 50 kHz low-pass filter on the output because the high amount of noise inherent in DSD (pushed out of the audio range through noise shaping, which only increased its magnitude above audio frequencies) has to be blocked. Without the 50 kHz output filter, high frequency garbage was knocking some power amps into instability. You may remember some players (Denon?) with a switch for selecting between 50 kHz, 100 kHz, or bypass.

    But wait, there's more. Is the output level even the same between the two formats' conversions to analog? Meaning to within 0.1 dB. If one is as little as 0.3 dB louder, all else being equal, it can be heard as sounding more dynamic and better. You need an SACD and a CD test disc with sine waves of equal amplitude to answer that.

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    still apples to oranges, part 3
    Written by Nick P. , August 29, 2008

    So there you have four things that can make all CD/DSD listening comparisons invalid. I know of only one real comparison, because the 16/44 part of this one comes straight from SACD and DVD-A, not through a separate chain, thereby eliminating differences in production and playback equipment (we're comparing formats, not gear). And it is by Meyer and Moran, "Audibility of a CD-Standard A/D/A Loop Inserted into High-Resolution Audio Playback" pages 775-779 in Sep. 2007 JAES (volume 55 number 9). Here's the first paragraph, you can purchase the details from the AES:

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    Written by JEJ , August 29, 2008

    Looks like digital has just as many problems as vinyl. I have to say it is kind of fun going back to having a turntable (two actually) as well as CD and SACD in the listening room. Rediscovering some classic jazz recordings of the late 1950's and early 1960's has been very enjoyable. The re-issued LPs are cut at 45 RPM instead of 33-1/3 so even the inner grooves sound great. Yes, even new, there are occasional pops and tick, but I can live with it.

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    Nick P - PCM is Continuous?
    Written by adcdac , September 01, 2008

    Hi Nick,

    I read your first comment with great interest. I don't think it's technically correct to call PCM data "continuous." By its nature, the PCM audio data consists of discrete digital words.

    Theoretically, a sinc interpolation filter can reconstruct the sampled data ideally.

    Practically, though, physical sinc interpolators are hard to come by.

    So, it's typically down to DAC's internal circuitry. In many delta-sigma audio DACs currently on the market, the analog output is formed by the really fast switching capacitors at the output. The target is to map an error-free digital word to a an exact analog voltage. Of course, there will be errors, but luckily they are pretty small. And the voltage between the words would basically depend on the discharge characteristics of the bank of switched capacitors. Things get even more complicated because the delta-sigma output is often multi-bit and noise shaped. But still, thanks to high speed circuitry...while not quite a *perfect* reconstruction, it's pretty darn close.

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    previous post didn't complete
    Written by Nick P. , September 02, 2008

    and the blog's error messages didn't give the right reasons but what the AES article came to was this - a year, pro systems, audiophile systems with expensive cables, audiophiles, university students in recording program, pro recording engineers, well over 500 trials... and correct answers no different from blind chance. The experiment: Does adding a high quality 16/44 A to D to A converter "bottleneck" after SACD and DVD-A players change the sound when the CD-quality loop is switched in/out?

    No. Not unless you turn things up enough to hear the noise floor. Perfect enough reconstruction for me and encoding to go with that. (Try that even with super-quiet JVC virgin vinyl.)

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    Apples to Oranges
    Written by JM , September 07, 2008

    I am not so concerned with DSD vs. CD, but rather with digital vs. vinyl. A vinyl slab cut from the same digital source used to produce a cd or sacd at least gets you closer to some baseline for comparison than a record wholly produced in the analog realm.

    We repeatedly have found that just about anyone can pick out the digitally mastered vinyl, once they recognize the sonic signature. When comparing original Stones pressings with DSD mastered vinyl, and the sacd/cd, everyone who heard agreed that the remasters sounded similar no matter the format, and that the original sounded possessed a much different signature.

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    ...
    Written by JEJ , September 07, 2008

    When the master is digital and the release is vinyl, whatever digital artifacts there are with digital recordings are there in the vinyl with whatever artifacts that vinyl cutting has. I have not listened to such a recording yet. I am pretty busy with all the classical jazz releases that were recorded analog, edited in analog, and cut in analog vinyl. I know they have more distortion, but it feels so good, I am having a heck of a good time.

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    lemons vs oranges not close enough
    Written by Nick P. , September 15, 2008

    The DSD-mastered RS vinyl was direct metal mastered, previous RS vinyl wasn't.

    To be in a position to say "I know what I heard" without sounding like the opposite, eliminate all outside variables a la Meyer/Moran but replace the hi-res disc source with a vinyl rig. Since cartridge output voltages and phono pre gains vary, make sure the combination is right for the ADC (use level control if necessary) or we're back to perceived digital signature = wrong hands at the controls.

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    lemons vs oranges, rehash
    Written by JM , September 18, 2008

    Eliminating outside variables will necessarily have no reliable effect when your measuring device is the human brain. Minimizing said variables may only skew the results somewhat. Have you ever conducted a test where you are asking for impressions?

    I am guessing that the point made with respect to the direct mastering is that the new RS vinyl offers a more accurate sound? Hence a more accurate copy of the digital signature? I am also guessing that you have never actually conducted such a comparison yourself?

    Typically, the results of listening tests have been quite obvious and consistent, despite what you might guess.

    You are right about one point, however, as the volume goes up, so does the listener's perception of the digital signature.

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    clarifications for JM
    Written by Nick P. , September 19, 2008

    I didn't say listener's perception of digital signature goes up as volume does though that can be true for any sound good or bad. I was referring to clipping - a hard limit not a progression. Common on pop CD and audible. Also occurs when an ADC is overloaded.

    In the Meyer/Moran experiment the testers knew that the CD-quality A/D/A was in the loop ONLY when the volume was cranked high enough to hear the noise floor - that's with NO source playing.

    Regarding RS vinyl, maybe I should have pointed out the obvious: The DSD-mastered vinyl and the SACD sound a lot more like each other than any other renderings simply because they're from the same master - same EQ, compression, generation etc. Are the things you're ascribing to digital artifact really that? How do you eliminate the possibility that those sounds on the analog master? Yes, direct metal mastering has less surface noise and no pre-echo to speak of which makes it more revealing than lacquer mastering.

    Since vinyl cannot be cut, pressed, and listened to in real time but digital can, we're back to switching A/D/A in and out of ONE source, ANY source, to see if the conversions change the sound. Research (real) says it doesn't.

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    Listen at times
    Written by JM , September 22, 2008

    The DSD-mastered vinyl and the SACD sound a lot more like each other than any other renderings simply because they're from the same master - same EQ, compression, generation etc. Are the things you're ascribing to digital artifact really that? How do you eliminate the possibility that those sounds on the analog master?

    I have heard numerous digital masters pressed to vinyl, some from cd. The difference is pretty dramatic. You need to actually listen some time rather than just conjecture or read about tests.

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    Think at times
    Written by Nick P. , September 23, 2008

    You're saying that I should put a lot more faith in your uncontrolled biased poorly thought out tests since I don't do any listening (by the way that's your assumption, who else but a media junkie would be reading this article)? If your conclusions are that good, shatter the world by having them published in the AES.

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    Read if you won't listen
    Written by JM , September 23, 2008

    My "assumption" is based on the fact that you said yourself earlier that you read about a study and the conclusions were good enough for you.

    "The (AES)experiment: Does adding a high quality 16/44 A to D to A converter "bottleneck" after SACD and DVD-A players change the sound when the CD-quality loop is switched in/out?

    No. Not unless you turn things up enough to hear the noise floor. Perfect enough reconstruction for me."

    If you won't listen, try reading something else. There are real engineering reasons why digital, to date, has not matched analog reproduction in many ways. Try googling Tom Scholz and this topic. He is an MIT trained EE who possesses a wealth of studio experience with his band Boston.

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    Even the digital world recognizes "analog" sound
    Written by JM , September 24, 2008

    http://www.digidesign.com/inde...temid=1561

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    More reading 
    Written by JM , September 24, 2008

    Please note the last line. As I stated earlier, no statistically reliable method to test matters of human perception is readily available. It is folly to begin to even make such a proposal.

    http://www.abcviolins.com/blindlistening.html

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    vinyl is very revealing... about some of its promoters
    Written by Nick P. , September 24, 2008

    http://www.digidesign.com/inde...temid=1561

    That's just another plugin that adds distortion created by analog tape machines and overdriven amps. Is this irrelevant link supposed to prove that analog sounds better than digital? Good burn there buddy.

    http://www.abcviolins.com/blindlistening.html

    Did you read the article? Let me answer that: You didn't, because its point is about people who chicken out of blind testing to avoid exposure of being pompous know-nothings, and it has little to do with a listening test using recorded music where hundreds of people can undergo the same test in the same conditions a year apart, control the switch from A to B, taking as long as they want to listen, as many times as they want.

    Keep saying I don't listen but I have collected and listened to literally tons of media starting with the acoustic 78 era. Musician? Me too. EE? Me too. Undergone blind listening tests? Me only it seems. Worked in the consumer and pro audio industries? Me too. For real. Big deal. It's the statistically proven facts that matter here if you're making a blanket statement. That's what I'm presenting and readers can decide for themselves. I'm mostly here to read about interesting new (re)releases.

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    Promoting vinyl?
    Written by JM , September 24, 2008

    Perhaps, you should actually take a moment to read my posts. At what point did I proclaim the superiority of any format? I merely stated (and not very assertively) that we found that vinyl cut from digital masters reminded us more of the cd, than of vinyl releases cut from analog masters. It was suggested that the editor try this for himself to see what he hears. Have you forgotten the test going on here?

    You should probably re-read the violin article. I think that you missed the point. Re-read the last sentence. It is what I have been saying over and over and over....

    The problems with digital are just as easily understood and measured as are the problems with analog. If you cannot understand this, then I have to question your engineering pedigree, which means that I have wasted much time here.

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    You certainly missed the point of the article
    Written by JM , September 24, 2008

    Perhaps, you can pick up the discussion of your "statistically prove" testing method here.

    http://www.audioasylum.com/cgi/vt.mpl?f=critics&m=40379

    I have grown tired of this nonsense. It would help a bit if you could ever post some correct information.

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    sorry JM maybe I was dismissive...
    Written by Nick P. , September 26, 2008

    ...because when you mentioned digital signature in vinyl from digital masters, I assumed you meant that a digital signature was a bad thing. If you meant that a digital signature is good, then you might want to clarify that, defining "digital signature" while you're at it.

    If you meant that *specific examples* of vinyl can sound better than digital counterparts (see your Sep 07 post), and vice versa, then giving them would have helped - that's what most of us are here to read about. For example, I found that the original Living Stereo Lt. Kije sounds better than Chesky reissue vinyl whereas with Gaite Parisienne it's the opposite, and with Leibowitz-conducted Beethoven, Chesky CD wins. Anything goes - see my posts following part 5 for other examples and their causes (and my answer to Tom Scholz which you later asked me to google).

    But...

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    on the other hand
    Written by Nick P. , September 26, 2008

    you have dismissed all blind testing. That last sentence you're referencing says that members of a panel of wine tasters may disagree with themselves the next day. Hel-lo! Use more trials and improve the conditions until the result is meaningful.

    The ADA loop test I talked about took a year and has a fundamental difference: Subjectivity was not involved. It was a very simple, does A sound *different* from B, yes or no?

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    getting tired too
    Written by Nick P. , September 26, 2008

    If subjectivity is involved in a different listening study, credentials are part of it. Your hearing is either normal or it isn't. During 40 years of testing Floyd Toole found that those with normal hearing agree very closely on what sounds nice and those without it disagee wildly (i.e. a terrible sounding speaker can be the perfect hearing aid). On top of that, bogus stats are weeded out with the following tricks to see if the listener is consistent:

    - A and B are the same thing, but don't tell that to the subject
    - re-test the same A and B the next week and the week after that, but tell the subject they're different things 
    - tell the subject that it's the same test as last week but don't say that A is now B and B is A

    That, to me, is highly trustable. I did the A/D/A test for fun using archival quality equipment some years ago. No change in sound. Just a small group's observation not worthy of mention when there's a much larger scientifically acceptable study to reference.

    So to summarize:

    A high quality 16/44 ADA chain has zero effect on sound.

    Looping is the only way to have a meningful comparison.

    Analog and digital may have distortion depending on which title you're playing but digital doesn't have to.

    I'm ready to get back to regularly scheduled programming. Not interested in responding to continually irrelevant info.

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    Ah ha
    Written by JM , September 26, 2008

    Okay, now I get it. You have been kidding us the whole time. Obviously, this is the case based upon your last 3 points, especially number 3.

    See, and I thought you were being serious. I should have known better. No one could seriously post the nonsense you have been spewing. You got me.

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    Distortions unnatural??
    Written by ts90 , October 08, 2008

    I am no expert, and maybe someone commented on this already - I apologize. But the distortions are present in REAL-LIFE as the different frequencies of sounds from different instruments bounce off the different surfaces in the room at different speeds. We know this in nature as "real life". True digital recordings do not capture these effects (the pick-up device on a guitar hears things differently than our ears in the concert hall) and; therefore, they present us with pseudo-sound - something that is "mathematically perfect", but absolutely unnatural. The distortions "added" by the phase differences in the electronics helps build back into the recordings what we perceive is natural.

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    distortion: intentional vs unintentional
    Written by Nick P. , October 10, 2008

    Recording and playback chains' job: Reproduce what the microphone hears as accurately as possible.

    Guitar amp's job and very desirable side effect: Make the guitar or blues harmonica louder *and* add distortion (frequency response change, clipping from tubes, saturated output transformers, sagging power supplies, speakers pushed to limits, resonance and feedback between speaker and guitar). Most electric guitars sound lousy when merely amplified, plugged into a "clean" system. The amp is part of the instrument. Voice, on the other hand, usually sounds lousy through a guitar amp.

    Sometimes the recording chain is also overdriven to add effect to mic signals, making them sound like Little Richard or Bo Diddley records (sound quality wise but not necessarily talent wise) or old movies. But that's not what our ears would hear in the same room as the performers, and I for one wouldn't want all my recordings to sound like that. What did movies sound like in theaters with a 1-channel system on overdrive? Nowhere close to natural as I remember it, and movie sountracks contain voices, sounds occurring in nature, and just about all musical instruments.

    Like I said at the end of part 5, a system should be evaluated based on how it sounds on all recordings, not just certain ones potentially through rose tinted glasses.

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    apples to oranges
    Written by Steven Sullivan , October 20, 2008

    Nick P, I agree with pretty much all you've said, except that when I compared rips of Rolling Stones SACD CD layers, to 88.2/24 digitized analog output recordings of the DSD layer, they were remarkably similar -- and I didn't find any clipped peaks in the CD rips. In fact, the CD rip looked very much like a simple transcode of the DSD version, which is the way it's 'supposed' to be done, but too often isn't (e.g. "Dark Side of the Moon", where the same comparison procedure reveals OBVIOUS tinkering with the CD layer, compared to the DSD layer).

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    for Steven
    Written by Nick P. , October 21, 2008

    Since the clipping is mild, an extreme close up is needed such that the width of the monitor represents around a thousandth of a second or fifty samples. Preferably, the wave viewer or editor shows the samples not just the wave.

    Randomly selected example: 26 seconds into I'm Going Down from Metamorphosis Mick sings Oh-Ah-Ahm Goin Down. Zoom in on the beginning of Ahm. I'm pointing it out because with a dozen samples clipped at a time it's more obvious than earlier parts of the song.

    With peak levels matched, the CD layer has higher average RMS power, which also spells dynamic range compression.

    Speaking of "simple" transcode, in this case it was done by Super Bit Mapping Direct which also adds noise shaping. Noise shaping vs straight TPDF dither... not so simple and that's a whole other story but it goes back to "I know what I heard" vs "Don't be so sure if you don't know the variables".

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    Why so much distortion in the hometheaterhifi.com tests?
    Written by Arnold B. Krueger , October 22, 2008

    http://www.hometheaterhifi.com/images/stories/april-2008/vinyl-vs-cd-mt10-stock-cartridge-1-khz-25-ohms-thd n-graph-large.gif

    Note that this graphic shows a 1 KHz tone, with the second harmonic about 20 dB down, which I call 10% second harmonic nonlinear distortion. 10% distortion is a lot of distortion by any standard.

    to:

    http://www.pcavtech.com/play-rec/rega-2/grado-SNR.gif

    Note that this graphic shows a 300 Hz tone, with the second and third harmonics each 40-45 dB down, which I call less than one percent second and third harmonic distortion.

    IME the latter results are hihgly typical and the former ones are rather atypical.

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    Why so much distortion in the hometheaterhifi.com tests? - correction
    Written by arny krueger , October 23, 2008

    http://www.hometheaterhifi.com/images/stories/april-2008/vinyl-vs-cd-mt10-stock-cartridge-1-khz-25-ohms-thd n-graph-large.gif

    Note that this graphic shows a 1 KHz tone, with the second harmonic about 40 dB down, which I call 1% second harmonic nonlinear distortion.

    It's still surprising that a far less SOTA system like the Rega could have somewhat less nonlinear distortion.

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    ...
    Written by JEJ , October 23, 2008

    The second harmonic may be a 1% distortion peak, but I measured the entire THD plus N for all the graphs, which was of course a much higher number than simply measuring the harmonic peaks. Most of the number is the noise component. You can simply look at individual harmonic peaks to see how they relate to the fundamental for that kind of comparison.

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    loudness war graphic examples
    Written by Nick P. , November 03, 2008

    http://spectrum.ieee.org/aug07/5429

    The older example is typical but not all-inclusive. Intentional unnecessary clipping still went on in 1990.

    The newer example is also representative but doesn't show how bad things got. The following is no joke and examples predating everything shown by a few years exist:

    http://www.cdmasteringservices.com/wallofsound.htm

    You can't do that with vinyl but I'm sure record producers would have liked to.

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    ...
    Written by JEJ , March 23, 2009

    I found a very interesting comparison of all audio formats in terms of dynamic range, frequency response, sampling rate, equivalent bits, and fidelity potiential index. Vinyl does not rate very high, but that still does not take away my enjoyment of them

    http://www.newformresearch.com/fidelity-potential-index.htm

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    Excellent loudness war tutorial video
    Written by Nick P. , September 22, 2009

    search "loudness war" on

    http://www.youtube.com/

    At the end of part 5 someone postulated that whether one's comfort food is vinyl or mp3, it's simply because of habit. I'll add massacred dynamic range to that idea after seeing the reactions of some friends who compared the dynamically rich 1986 CD of Led Zeppelin II vs the remastered/range-limited 1990 version and the even more limited 1994 remastering of the 1990 remaster, which also makes up the latest box set from Japan.

    Do enough research and you'll even find that people who were raised on classical music in the 78 rpm era might feel that only the sound of 78 rpm recordings can convey the power of the live orchestra. Such was the case with a friend with which I had many musical discussions spanning 25 years. He was born more than 40 years before I was and had a very simple rule for judging (classical) recordings: The best one is the one that you listen to most often, which is often the one you heard first.

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    quote from Bob Ludwig at Gateway on vinyl vs digital
    Written by Audionirvana , October 03, 2009

    Analog distortion in vinyl may sound euphonic but if you're interested in hearing what the source material really sounds like, look elsewhere.

    'With high resolution digital, it is almost impossible to pick out the original from the copy, while with vinyl, one can ALWAYS pick out the original vs the vinyl playback!"

    Bob Ludwig

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    a couple of points
    Written by Steven Sullivan , October 03, 2009

    NicP, I don;t own Metamorphosis but I do not recall clipped peaks even at high resoltuion on the Stones SACDs I do have. I can take another look though, using software that scans for consecutive samples at the same level.

    And JEJ, it's really quite misleading to show DAC output before reconstruction filtering had done its job. NO ONE hears the sine waves shown above.

    How about showing the results of all those AFTER filtering, and measuring the differences between them?

    Also, the contention by some posters that digital always is inherently unfaithful while analog is inherently faithful to the input signal, is absurd. There are no perfect recording processes of real world signals, the laws of physics prevent it; errors are ALWAYS introduced, and far more are introduced in 'good' analog than 'good' digital. After all 'analog' is called ';analog' for good reason -- it's analogous, but not exactly equivalent, to the original. (Digital of course is also 'analogous', but the term 'analog' had already been co-opted)

    And anyone who points to Audio Asylum as a source of scientific information on sensory/perceptual testing cannot be taken seriously.

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    vinyl rituals
    Written by ws , October 03, 2009

    Not voting one way or another, but I think many of the vinyl lovers enjoy the rituals associated with it.

    - the careful disrobing of the record 
    - the de-staticing and dusting 
    - the gentle lowering of the cartridge 
    - the tweaking

    After all that work, of course it's going to sound good. :-)

    PS For the record (no pun) I wish I had a lot of 50s era jazz classics on vinyl.

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    ...
    Written by JEJ , October 04, 2009

    A Couple of Points said - And JEJ, it's really quite misleading to show DAC output before reconstruction filtering had done its job. NO ONE hears the sine waves shown above. -

    It's not misleading. You just scanned the paragraphs too quickly. I specifically mentioned in paragraph 5 - The lines connecting the dots represent the signal that is fed to the output stage before any filters are applied. - and - The filter's job is to make these jagged lines more sinusoidal, so that it will be like the music that was recorded, which is also sinusoidal.

    The point of the discussion is to show how difficult a job the filters have with conventional Redbook CD encoding, and that even with 24/192, the 20 kHz region, which is on average, the limit of human hearing, the unfiltered waveform is not smooth.

    A few years ago, an engineer tested various sampling frequencies and listened for improvements in the reproduced sound with higher and higher sampling rates. He stated that it was at about 450 kHz that the limit of audible improvement was reached. I am beginning to think he might be right.

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    Limitations...
    Written by JM , October 05, 2009

    One has but to listen to a Beethoven symphonies on LP to hear the dynamic limitations of vinyl. And the ability of a CD to play an entire symphony with out "flipping" to side two is not just convenient, but essential!

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    ...
    Written by JEJ , October 05, 2009

    You mentioned flipping to side 2 as a limitation. With my set of classic jazz LPs, remastered at 45 RPM, I have to flip to side 2, then a second LP for side 3, and then to side 4. Each album is two LPs because of the increased tracking velocity (45 RPM instead of 33-1/3 RPM). Each side plays for about 10 to 15 minutes.

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    how to show dots and sine wave on top of one another
    Written by Nick P. , October 05, 2009

    Use Cool Edit Pro or its Adobe successors.

    Even though the dots riding on the signal have a ridiculously different pattern, especially at high frequencies, that's how PCM works and the low-pass filter is key.

    If anyone wants to talk about how things change as signal frequency approaches half the sampling rate, even while ignoring an invention called oversampling, first look through the same glass at what happens when the same signal is cut to vinyl and played (one of the reasons that discrete quadrophonic had no chance).

    Speaking of vinyl loving rituals, a must-see: the Everybody Loves Raymond episode where he gets his 50s-jazz-vinyl-loving dad a CD player for Christmas.

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    Straight Lines
    Written by Josuah , October 05, 2009

    You're not supposed to draw straight lines between the samples. Even if you are trying to show the result before the low-pass "analog reconstruction" filter.

    See "The Procedure" section under: http://en.wikipedia.org/wiki/N...ng_process

    The reconstruction filter is to remove aliasing artifacts. Not to make the curve "curvy" instead of "straighty".

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    Straight Lines (cont.)
    Written by Josuah , October 05, 2009

    Or:

    "Theoretically, the interpolation formula can be implemented as a low pass filter, whose impulse response is sinc(t/T) and whose input is [a Dirac comb function modulated by the signal samples]."

    Either way, the input into the low pass filter is not just the samples with straight lines connecting them.

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    ...
    Written by JEJ , October 06, 2009

    Regardless of the reconstruction algorithms, the graphs show that there is not much information to reconstruct the sinusoidal waveform at high frequencies. It is certainly not perfect. Otherwise, an article on Vinyl vs. CD would not even have been written by anyone, including us. It is because of the imperfect reconstruction that so many consumers don't like the sound. It is, after all, a reconstruction of the analog signal, not a reproduction. The process tries to fill in the blanks, and at high frequencies, the blanks are pretty big with 16/44 sampling. I think that if we were to go to 500 kHz sampling, 24 bit, no filter would be necessary because typical studio microphones don't respond beyond 20 kHz, and some even roll off at 15 kHz.

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    Re: Straight Lines
    Written by Josuah , October 06, 2009

    My main disagreement is with the statement that the graphs you posted are a good representation of the pre-filter signal. I don't believe them to be so, nor do I believe 44.1kHz is fundamentally lacking any information needed to reproduce sounds below 22.05kHz (non-inclusive).

    I do agree that a higher sampling rate both when recording and playback will allow you to more accurately reproduce the sound, given practical considerations that come into play outside the realm of pure math. That's why modern ADCs and DACs oversample.

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    ...
    Written by JEJ , October 07, 2009

    If you make a statement that you don't believe the graphs are true representations of the signal before filtration, you will have to say why you don't believe it, i.e., provide a link to a mathematical analysis that supports your statement. Otherwise, it is only an unsubstantiated opinion with no basis. Secondly, as to the 44.1 kHz lacking any information for reproduction of frequencies below 22 kHz, you contradict yourself in the next sentence by saying higher sampling rates will allow more accurate reproduction. If it's more accurate, then something was lacking in the lower sampling. And there is no "outside the realm of pure math" with digital sampling. It is all math.

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    Re: Straight Lines
    Written by Josuah , October 08, 2009

    I don't believe the graphs are true representations of the signal before the low-pass filter because the dots were connected with straight lines instead of connecting them using the mathematical formulas described on the Wikipedia page. Specifically the last bullet point under "Mathematical basis for the theorem" which I quoted earlier, with respect to the Dirac comb function.

    http://en.wikipedia.org/wiki/ Nyquist–Shannon_sampling_theorem#Mathematical_basis_for_the_theorem

    44.1kHz does not lack the required information for sound reproduction below 22kHz, but you can deconstruct/reconstruct more accurately if your process is performed at a higher sample rate. It is somewhat analogous to performing a chain of multiplication and division operations using decimal places even though your original numbers are integers. By using the decimal places, you help avoid rounding errors. This is mentioned under the "Practical considerations" section of the Wikipedia page.

    http://en.wikipedia.org/wiki/ Nyquist–Shannon_sampling_theorem#Practical_considerations

    Dan Lavry has a nice paper about this:

    http://www.lavryengineering.com/documents/Sampling_Theory.pdf

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    Post vinyl
    Written by JM , November 16, 2009

    It would be nice if we could see posts of vinyl at 10K and 20K from both an all analog recording, as well as one cut from a digital master.

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    re: Post Vinyl
    Written by Nick P. , November 18, 2009

    Analog outputs of analog and digital sine wave generators (as master as a master gets) can be compared directly so what would be the point of adding vinyl to each of their outputs?

    The real test is still this - analog generator feeds A/D/A stage so output of A/D/A can be compared to the source. Measurable difference? Maybe. Audible difference using 10k and 20k sine waves? Good luck.

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    The point is....
    Written by JM , November 19, 2009

    to actually see what the vinyl output looks like vs. the digital, since this thread was all about vinyl vs. cd, and not about your tired, unimaginative, repetitive pontification regarding the merits of 16/44 via engineering 101.

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    You missed your point
    Written by Nick P. , December 01, 2009

    You asked for a vinyl vs vinyl comparison with the analog vs digital master being the variable. That's not a vinyl vs CD comparison which is why I asked what I asked. It was a question not a pontification. So without complicating things, what's the answer?

    And this is not engineering 101, it's addition and subtraction. Take the master and the copy, level-match, invert the polarity of either one and mix the two signals. Whatever they have in common gets cancelled, whatever they don't have in common is what's left and it's called distortion plus noise.

    Since you asked for 20 kHz in the vinyl vs vinyl comparison you might also want to specify whether it's the first play or the 20th, the 50th etc. Maybe 20 kHz according to number of plays can be its own comparison. Wouldn't you find that interesting?

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    CD vs vinyl vs ... MP3!
    Written by Enrique , February 02, 2010

    Now that everybody is listening to MP3 through their iPods, computers, even hi-fi equipment, the whole topic is marginal at best. Perhaps it would be interesting to know how much are we missing by using MP3.

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    CD vs vinyl vs ... MP3!
    Written by Piero , February 03, 2010

    Enrique, you need a graph or machine to tell you the difference? Just use the best instrument we have, your ears!

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    Nothing is perfect but...
    Written by Juan Pablo Cuervo , June 24, 2010

    LPs from a good master sounds more real & detailed than the same song from a CD.

    usign decent analog equipment vs. the best digital equipment.

    the details are not IMD distortion or Noise.

    technically its far more complex than just IMD and noise floor.

    there are many other variables not even considered in the test.

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    What do you think?
    Written by Just a Guy , July 20, 2010

    OK. You guys are so far out there on the technical end from my perspective. I read everything posted and understood some of it and the jist of most of it. (I'm a Degreed Mechanical Engineer) I most enjoyed the Sandbox Kicking the most!!! That was amusing!! I liked the posts in the first half that say, "just enjoy it" etc.

    Here is the deal. I liked to this blog because I am interested in buying a new Stereo. You guys have shown me something different and educational. Let me paraphrase from what I understood so a normal person can understand it.

    * CD Technology is limited to 16 Bit Resolution/44.1Khz Sampling rate. when an analog recording is recorded on CD, the music is sacrifcied.

    * Multimedia PC's can handle uncompressed 24 Bit / 96Khz 7.1 sound.

    * A good recording is at 96Khz.

    * Modern Recording are above 100Khz., but have to be compressed to 44.1Khz to be playable on a CD.

    * SACD and DVD-A are compressed. Soon to be obsolete technology

    * 7.1Ch lossless Codecs on Blueray is emerging standard.

    * The recording and engineering make the difference.

    * Telarc produced some of the best sounding CD's

    (feel free to correct or add)

    I was mostly saddened to hear JEJ state that 450Khz would be the rate to displace analog. It's ok, it just means I have to wait. . .

    As an outsider looking in; I have a bigger picture than the nitty/gritty technical arguments. This post by C. Weber June 22, 2008 says it all for me.

    Multi-stereo surround recording and High Resolution Recordings. Call me a Zen, Freak, or tell me to go back in my hole. Whatever. I'm just calling it as I see it from 30,000 feet. To me, this looks like the future, and the technology will come because of what you described. This is where to invest your stereo dollars, or equipment It's where I am headed as a consumer.

    IMO

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    What about crosstalk?
    Written by Jupiter8 , December 20, 2010

    What about crosstalk? Maybe I've skipped it, but a main thing to consider is the crosstalk between the two channels. With regard to vinyl this is a huge kind of distortion. Crosstalk simply doesn't/can't exist in digital formats.

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    What is your method of making 'digital 20khz sine wave'?
    Written by KYS , May 06, 2012

    Did you make it by 'microphone recoding'? or 'generating in software'?

    Please answer me. Thank you.

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    ...
    Written by JJ , May 09, 2012

    The digital signals shown in the graphs were generated digitally in a software program. They were not recorded with a microphone and a speaker. They have not passed through an A to D converter or DAC yet. They are simply a graphical representation of the levels (higher or lower recording levels) in the bitstream. Each of the dots represents one of the 44,100 samples per second (in a 16/44.1 bitstream) and indicates the level with respect to 0dBFS. The lines connect the dots so that a reader can see changes in the levels more easily. The DAC is fed the sample values in a bitstream (not the lines that connect the dots in the graphs), which is converted to the analog signal using a filter that results in a clean sine wave at the output.