Sanders Sound Systems - electrostatic

[Angela]

we are crazy busy, but I'll get Roger's ear this week on all this. Thanks for the replies and questions, earlinarizona.

and Steve, let us ponder on this one. Thank you so much for the offer!

 

[Steve Williams]

My offer stands. Please let me know. We could create a dedicated forum in the Mfrs Forum and move all of his threads to it

 

[earlinarizona]

Found an interesting paper named Low-Frequency Optimization Using Multiple Subwoofers* Does this have any bearing if I use 2 or 3 10c speakers side by side and not multi locations in the room. This question is in addition to the earlier ones posted.

 

[Gregadd]

I assume your speakers image well. In assessing the speakers imaging characteristics, what measurements do you rely on? Or do you simply voice it by ear?

 

[Angela]

I assume your speakers image well. In assessing the speakers imaging characteristics, what measurements do you rely on? Or do you simply voice it by ear?

Imaging is a geometric phenomenon having to do with the phasing (timing) and shape of the wave front produced by the speaker. This is different than "voicing" which is done by altering the frequency response of the speaker.

Imaging is determined by the physical shape and characteristics of the drivers and is not something that can easily be altered. For example a cone is a cone, and a panel is a panel. There is no practical way to change them. Yet their shape and behavior will largely determine the imaging of the speaker.

While the fundamental phase behavior of the speaker will be determined by the type of driver used, phasing can also be altered by the time alignment of the drivers, the behavior of the crossover, the polarity of the drivers relative to each other, the orientation of the drivers, and the dispersion of the speakers. It is possible to alter the phasing to a limited degree by working with these factors. But the effect will be minimal compared to the inherent imaging produced by the type of driver unless truly dramatic changes are made.

For example, a typical cone or dome driver is a point source, which produces a characteristic image. But you can change this quite dramatically if you use many such drivers and aim them all over the room -- think Bose 901 speakers with one cone facing forward and 8 cones facing backward to bounce sound off the walls behind the speaker.

In this case, most of the sound you hear is that reflected off the walls rather than that coming from the speaker and the image is much different than that coming from a single cone driver. This effect can also be produced by using leaf drivers (MBL) or inverted cone drivers (Ohm Acoustics) that produce 360 degree radiation.

By comparison, "voicing" a speaker involves altering its frequency response so that it is not linear. Generally speaking, most audiophiles will perceive a speaker with perfectly linear frequency response as sounding too bright and lacking in midrange "warmth" and "fullness". It will also not have enough bass to satisfy most listeners. In other words, audiophiles do not like speakers with truly linear frequency response.

As a result, most speaker manufacturers will deliberately alter the frequency response of their speakers to make them sound more pleasing to most listeners. Such "voicing" commonly includes suppressing the highs slightly, increasing the bass substantially, and boosting the midrange modestly.

But voicing a speaker in this way is tricky because loudspeakers interact with the room a great deal. It is the room/speaker interaction that largely determines the frequency response, and because most speakers are sensitive to room placement and shape, it is impossible for a manufacturer to know exactly what his speakers will sound like in any particular room.

For this reason, all good speakers have at least some adjustability so that the owner can tailor them to their room. This means that it is the audiophile/owner who ultimately determines the voicing of their speakers.

Most speakers do not really have enough adjustability to do a really great job of getting the frequency response "right" for a particular room. So most installations will benefit from using a DSP (Digital Signal Processor). By using the RTA (Real Time Analyzer) that is built into all good DSPs, you can see the actual frequency response of your room/speaker system. You can then make adjustments to eliminate problem areas (large peaks and dips) and then tweak the overall frequency response to that which you find most pleasing.

DSPs are extremely powerful devices. As such, they can make major improvements in the sound of your system. But if used poorly, they can truly ruin the sound too. They take practice and patience to use effectively. But it is fair to say that virtually all audio systems can be improved by their use.

As for measurements, the frequency response of speakers are rather difficult to measure accurately in a typical, reverberant, listening room like you find in most homes. Measurements cannot be made using sine waves and SPL meters.

Many audiophiles try to do so by using a test CD and a RadioShack SPL meter. This is not only a complete and utter waste of time, but it is deceiving and will lead you astray because you will think the measurements are accurate when they are not.

The reason that sine wave testing is useless is because the room will interact with the direct sound from the speakers and generate peaks and dips in the response that makes it appear that the speaker has poor frequency response. These peaks and dips will change quite dramatically if you move the microphone just a few inches to a different location. Therefore, the frequency response you will measure is based on microphone position, and does not represent the actual performance of your speaker.

This room interaction problem is the reason that accurate sine wave measurements are always done in anechoic chambers. An anechoic chamber absorbs all the sound from the room so that only the output of the speaker is measured.

But it is possible to measure the frequency response of a speaker in a live room using special techniques. The best is MLS (Maximum Length Sequence) testing. This is also known as pseudo-anechoic testing.

MLS measurements are done by using a quick noise burst (which contains all frequencies) and then using FFT (Fast Fourier Transform) analysis to measure each frequency. A computer is required for this complex measurement technique. But FFT analyzers are now available as computer software for quite reasonable prices that any serious audiophile can use.

To eliminate the room from the measurement, the MLS test is "gated." This means that the system will "listen" to the sound coming from the speaker for only a few milliseconds. Then the microphone is cut off before the room reflections have time to reach it. The result is that the FFT analyzer only hears the sound from the speaker -- not from the room. As a result, it can capture the true frequency response of only the speaker.

But gating has its limitations. It only works for short wave lengths (midrange and highs) which are much shorter than the dimensions of the room. The wave length of bass frequencies are longer than the dimensions of the room. Therefore, you cannot leave the microphone live for a long enough time to "hear" an entire bass frequency without some of the room reflections also getting through. So MLS testing does not work for the bass unless the room is extremely large.

A better way to measure bass is using an RTA. This involves using pink noise and measuring frequency bands (typically 1/3 octave bands). The room is involved in the measurement, but by obtaining many measurement samples and averaging them over time, you can get a reasonably good picture of the bass frequency response. The resolution and detail is low because bands rather than individual frequencies are used, but bass frequencies are non-critical in this regard. So an RTA works quite well for measuring bass.

Because an RTA includes the room in the measurement, it does not tell you what the speaker's bass performance alone is. However, to achieve powerful bass, a room must be involved to support it. So the bass measurement MUST include the room to be meaningful in a typical listening environment.

Think of a marching band. When outside, the band has very poor bass. If you close your eyes and just listen, you will note that the frequency response is similar to what you would hear from a telephone. It's really poor.

But take that same marching band an put it on the stage in a concert hall. Now the bass is spectacular and powerful. That's because the room is confining and supporting the bass energy and making it much louder and impressive.

The same is true of your speakers. Take them outside and they will have very weak bass. But in a room, the bass is really powerful.

In short, you need to measure bass in a room using an RTA. But an RTA doesn't give very detailed information so it doesn't work very well for the midrange and highs. MLS testing is far superior for the midrange and highs.

Also, because MLS testing uses a computer and FFT analysis, it can also display the transient response information. You can see this as impulse response, energy/time graphs, and waterfall graphs. Transient performance is extremely important and a speaker should be evaluated for this performance just as much as its frequency response.

Imaging is a function of our brains that take the phase (timing) and loudness information from two speakers and process it into an image of the original sound. No machine can do this, therefore there is no specific measurement of "imaging."

You can get some idea of how a speaker will image by measuring its polar frequency response because this will give you some idea of its dispersion. Generally wide dispersion will produce relatively poor images because the reflected sound from the room will delay and confuse the phase information. For this reason, narrow dispersion speakers give a much better and more realistic image than wide dispersion speakers. You can get more information on this topic from my White Paper at:
http://sanderssoundsystems.com/technical-white-papers/dispersion-wp

Phase measurements will also give you a clue to the quality of the image. A properly "time aligned" speaker will have the sound from all the drivers arriving at your ears at the same time. With precise phase information, your brain can reconstruct a more accurate image.

Impulse testing is also helpful because it shows the phase behavior between the different drivers. For example, most speakers use passive crossovers and Butterworth filters operating at 12 dB/octave. This causes a 3 dB peak at the crossover point. To eliminate this peak, most speaker manufacturers deliberately put the drivers 180 degrees out of phase.

While this flattens the frequency response, it degrades the phasing and hence the imaging. It would be far better to use different crossover slopes or Linquitz/Riley filters to eliminate the frequency response error. Then the drivers could be wired in-phase. But this costs more money, so most speaker manufacturers sacrifice the phase to improve the frequency response.

But none of these measurements will truly predict the imaging of a speaker. Only your brain can do that. So you will need to listen to the speakers and judge the imaging for yourself.

Always keep in mind that a quality image require a quality recording that is done in true stereo. Few recordings today are made that way. Most are processed to death with artificial reverb, compression, and close mic techniques that are actually done in monaural. As a result, they cannot produce a realistic image -- even if the speakers can.

Older recordings were often made without much if any processing, using pairs of microphones in a natural "hall" environment. So older recordings are often able to produce much more natural imaging than modern recordings. I am convinced that much of the popularity of older vinyl recordings is due to the more natural recording techniques used on them rather than to the fact that they are recorded on vinyl. It is the realism of the recording techniques that make the sound so enjoyable.

With that background, I can now answer your questions in order. Specifically, I pay a great deal of attention to imaging. I deliberately use narrow dispersion speakers to eliminate the room acoustics and I use planar speakers get accurate phasing. I use electronic crossovers and electronic delay to get perfect time alignment of my drivers. I use both MLS and RTA testing to measure frequency response. I measure transient behavior with impulse testing, and produce both waterfall and energy/time graphs.

I "voice" my speakers by giving customers tremendous adjustability in my electronic crossovers so that they can get exactly the sound they want in their particular listening room. Of course, I supply the speaker system fully programmed for what is close to linear frequency response, so the customer need only make two small adjustments to get the sound just right. And it is not absolutely essential that the customer make adjustments. But they certainly can do a lot to make the speaker sound exactly right in their unique room.

-Roger

 

[kareface]

With that background, I can now answer your questions in order. Specifically, I pay a great deal of attention to imaging. I deliberately use narrow dispersion speakers to eliminate the room acoustics and I use planar speakers get accurate phasing. I use electronic crossovers and electronic delay to get perfect time alignment of my drivers. I use both MLS and RTA testing to measure frequency response. I measure transient behavior with impulse testing, and produce both waterfall and energy/time graphs.

Narrow dispersion will only help till around 500hz. You can never escape room acoustics, lol.

 

[kach22i]

Always keep in mind that a quality image require a quality recording that is done in true stereo. Few recordings today are made that way. Most are processed to death with artificial reverb, compression, and close mic techniques that are actually done in monaural. As a result, they cannot produce a realistic image -- even if the speakers can.

Older recordings were often made without much if any processing, using pairs of microphones in a natural "hall" environment. So older recordings are often able to produce much more natural imaging than modern recordings. I am convinced that much of the popularity of older vinyl recordings is due to the more natural recording techniques used on them rather than to the fact that they are recorded on vinyl. It is the realism of the recording techniques that make the sound so enjoyable.

Great information Roger, nailed it.

 

[kareface]

I really enjoyed your post Angela, it was a good read. Now that I'm home and have a little more time to focus, I would like to add my 2 cents. I've noticed a trend of high directivity and pinpoint imaging in a lot of commercial speakers. To achieve this you usually require a thin front baffle, which raises the edge diffraction frequency. It's not the best solution if you have more than 1 listener, as it's harder to achieve wide sweet spots. Also narrow dispersion, as I've mentioned before, will only apply to frequencies above ~500hz. That's not to say it isn't a solution, but whenever possible I always prefer trying to achieve an ideal acoustic environment over the directive option. If you look at how most sound is produced naturally, there isn't this high degree of directional. It becomes more of a way of avoiding room interference, but even then the reverb is still going to pick up tonal qualities of the acoustic space, which will be summed. I would agree that in situations where the room will be overly influential in the mids-highs and correction would be difficult, this would be a great solution. I've found myself suggesting this very thing to others in that very scenario. I don't want it to sound like I'm suggesting that it's the wrong way to go, it's just another solution, with it's own set of trade-offs like everything in acoustics. Thanks again for the great read.

 

[MylesBAstor]

Great information Roger, nailed it.

If so, then why do the digital issues of the original recording pale alongside the original vinyl release?

As I've said before, there are a number of reasons why recordings from the Golden Age of Stereo stand out fifty years later.

1. The hall/studio were preselected for sonic quality and once that was established, the engineers knew where to more or less place the microphones for different recordings. Corrollary: there were actually some good sounding halls out there.

2. Tube electronics and reel to reel analog tape. I'm sorry but if you have the chance to compare the same recording done digitally and analog, there's simply no comparison. IMHO. We can also talk about how mikes were better-and how the industry went though a period of horrible mikes that measured flat.

3. The orchestra nailed the playing the first time around, not playing one bar at a time and then correcting it say now with ProTools, etc. The orchestra played the whole movement with mininal takes. That's in contrast to today's attitude of fixing the issues after the fact; back then, the musicians actually could play a whole movement through without a mistake.

4. Simple miking patterns and techniques. 2 or 3 track recordings instead of 64 track.

5. The producers/engineers were former musicians (as opposed to the majority of todays technogeeks who feel the incessant need to twidle a dial or hit a key) who know music and what it sounded like. There was also less pressure to make a buck since many of these were small garage operations and instead were say as in jazz, paying homage to this "new" musical genre.

6. Minimal post production editing and messing around with.

6. Better conductors in classical music. Ozawa couldn't carry Munch or Fiedler's jockstrap. Look at the conductors of yesteryear (I'm talking stereo era here): Fiedler, Munch, Leinsdorf, Reiner, Dorati, Sargent, Hanson, Fennell, Kertesz, Ansermet and the list goes on and on and on.

7. Early digital recordings, in the first ten or so years, simply sucked. We've lost a whole generation of music because of this.

 

[kach22i]

If you look at how most sound is produced naturally, there isn't this high degree of directional. It becomes more of a way of avoiding room interference, but even then the reverb is still going to pick up tonal qualities of the acoustic space, which will be summed.

1. The hall/studio were preselected for sonic quality and once that was established, the engineers knew where to more or less place the microphones for different recordings. Corrollary: there were actually some good sounding halls out there.

I'm going to try to tie these two posts together, as I've just started reading Roger's white paper and had a brain storm (or tropical depression).

I am working on some "ripple" speakers, this means the sound wave/radiation pattern is 360 degrees or a sphere. The biggest thing I've noticed and my wife has noticed is that outside of the room it still sounds good. The room is being loaded much the way live music loads a room. And I suppose, much like live music, there are bad seats in the house and good seats in the house.

Don't like what you hear, move your head an inch to the left and avoid that sound node (not yet experience with my ripple speakers, but experienced in auditoriums).

Now to quote Roger's white paper:
http://sanderssoundsystems.com/technical-white-papers/dispersion-wp

I eventually came to understand that there are three serious problems caused by wide-dispersion in speakers. These are poor frequency response, poor transient response, and poor imaging.

I don't know if Roger had in mind or heard the Museatex Melior One speakers (ripple type) when he wrote this, but so far with my experimental speakers it is only the last part of "imaging" which is somewhat lacking. I say this with the clarification that detailed imaging of highly resolving stereo systems for the most part is an artificial construct not typically found in a live performance, be it acoustic or otherwise.

To go out on a limb here; I am not a purest, and not a slave to the original signal and strict reproduction of that artificially derived/recorded signal. I believe that there can be good colorations which restore and enhance the listening experience and that these colorations can include the way the room is loaded.

Room loading to me means several things, but essentially it is the way the sound bounces around the room. Saying that, I'd rather have an under acoustically treated room than an overly treated room as I have never heard live music in a dead room. Yes, this means I have never listened in on a live recording session in a recording studio, something I'd like to change one day.

 

[kareface]

Saying that, I'd rather have an under acoustically treated room than an overly treated room as I have never heard live music in a dead room. Yes, this means I have never listened in on a live recording session in a recording studio, something I'd like to change one day.

For the purposes of reproduction I agree. I would like to add that a lot of methods I've seen for correcting LF involves nothing short of the abuse of mids & highs, I always try to avoid this route with the rooms I design.

 

[Gregadd]

Roger-Disgree or agrree you always present things in a common sense manner that translates into real world information the audiophile can act on. It appears you were discussing Stereo imaging-or left to right balance. Could you consider how we would go about measuring front to back imaging and the spatial relationships between musical istruments and voices? Often described as air around the instruments or voices.
gregadd

 

[Angela]

OK boys, put your feet up for this one: (angela)

If so, then why do the digital issues of the original recording pale alongside the original vinyl release?

[FONT=&quot]Your question of " . . . why then do the digital issues of the original recording pale alongside the original vinyl release?" is a good one. The answer is much more complex and surprising than you might think and requires a lot of background information before I can answer it. So sorry, but this will be a lengthy discussion. [/FONT]

[FONT=&quot]I agree with your observation that many digital re-issues of the original recording are far inferior to the original vinyl release. Like you, I found this to be very perplexing and troubling. So I embarked on a quest to find the reasons. I think you will find the following story to be most interesting. [/FONT]

[FONT=&quot]In the mid 1960's I had become a serious audiophile and had figured out that better source material was going to be essential if I wanted to get serious improvement in my audio listening experiences. I became so frustrated with the quality of the source material of the day (vinyl LPs and 2-track, open reel tape) that I vowed to start doing my own recordings.[/FONT]

[FONT=&quot]Due to all the restraints on recording imposed by the musician's unions, I found it was really tough to find musicians to record. But after considerable effort, I was finally managed to start using the recording booth at the concert hall of the University I was attending in 1968.[/FONT]

[FONT=&quot]As a poor college student, this was a godsend as I was able to use the finest equipment of the day (Ampex 354 studio recorders, Altec mixers, Neuman and Telefunken condenser mics, etc.) without my having to buy this equipment myself, which I could not have done. I also received excellent training by the staff. But probably the most important advantage was the fact that concerts were always recorded by the University, so I was able to record some truly great performers without union interference.[/FONT]

[FONT=&quot]As part of my recording training, I was also taught to align, maintain, and repair all sorts of recording and electronic equipment -- particularly the studio tape decks, which were always in need of attention. I had access to the University's test laboratory facilities for this purpose. [/FONT]

[FONT=&quot]I soon became one of the top technicians at the University and was able to make all manner of measurements and tests to evaluate and compare equipment to the sound of the recordings I was making. This lucky turn of events put me in the unusual position of being able to do serious, rigorous, scientific testing to figure out the actual cause/effect relationships of what I was hearing. [/FONT]
[FONT=&quot]
What made this so unusual is that I was a concert musician, audiophile, and technician all at the same time. Therefore, I didn't have to just accept someone's opinion about audiophile topics, nor did I have to take the word of engineers who had no knowledge of music or audiophile issues. I could actually do both listening and measurement tests myself to find out the true causes for what I was hearing.[/FONT]

[FONT=&quot]When I graduated from the University, I was able to use my experience there to get permission to do recordings for our local public radio station. I was then able to get past the musician's union problem to be able to record the region's symphony orchestra, opera, and pipe organ. [/FONT]

[FONT=&quot]Because I was doing that, I was also able to get local musical groups to allow me to record them as well. I often could then get them air time on the radio station for which they were very grateful. [/FONT]
[FONT=&quot]
By that time, I was also able to buy my own equipment and start a recording studio as well as do live, on-location recording. So I was very lucky, did a lot of live recordings, and have a wonderful library of music that I have recorded over the last 40 years, much of which is superior to what you can find on the commercial market.[/FONT]

[FONT=&quot]I quickly learned that natural settings in good-sounding concert halls were far superior to the artificially processed recordings that were done in recording studios. Of great importance is that most on-location work was done with a simple 2-microphone setup in true stereo using either the Bleumline, crossed cardioid mics, or widely spaced, omni mic setups. These techniques produced true stereo recordings with natural hall ambience, while studio recordings were done with single microphones that produced inherently dry, monaural sound. [/FONT]

[FONT=&quot]To get the mono sound from a studio to work in a stereo system required panning, artificial reverb, compression, and equalization. Furthermore, studio recording meant that each instrument and performer had to be have their own individual microphone and be recorded in isolation on a single track. The result were multi-channel recordings that then had to be "mixed" to produce the final sound. [/FONT]

[FONT=&quot]This meant that the sound of the final 2-channel mix depended on the judgment of a "recording engineer" to make it sound good. There was no natural or real sound to compare to, so the quality of the recording was totally dependent on the engineer. This was (and remains) a critical flaw in the recording stream. [/FONT]

[FONT=&quot]The close-miked sound from the recording studio was not very realistic, particularly when it was heavily processed and altered to sound "good." I much prefer a simple 2-mic stereo setup in a concert hall.[/FONT]

[FONT=&quot]I continued to do all my own alignment and maintenance work on my analog recording equipment. This was always frustrating because good performance simply was not available. While high-speed, 2-track tape had much better performance than LPs, it was impossible to make recordings good enough that you couldn't tell the difference between the source (usually a live microphone feed) and the recording.[/FONT]

[FONT=&quot]Specifically, even the best recorders were noisy and it was impossible to record a symphony orchestra without hearing tape hiss on quiet passages. The quietest studio recorders had a S/N (Signal to Noise ratio) of around 72 dB. By comparison a symphony orchestra has a dynamic range of about 80 dB. So it physically was not possible to make recordings with a silent background.[/FONT]

[FONT=&quot]The development of noise reduction systems by Dolby and DBX were a help, but they introduced other problems that degraded the sound in exchange for lower noise. The biggest of these was "pumping" or "breathing" noises as their compander circuits opened and closed in response to the music levels.[/FONT]

[FONT=&quot]Linear frequency response in analog recorders was impossible to achieve. It was considered outstanding to get plus/minus 2 dB tolerance from 30 Hz to 15 KHz, which is really quite poor performance.[/FONT]

[FONT=&quot]Distortion was almost a joke. While you could get distortion slightly under 1% at midrange frequencies, the frequency extremes were far worse. In any case, it was normal practice to heavily saturate the tape on loud passages in order to have a quieter background for quiet musical passages. [/FONT]
[FONT=&quot]
While magnetic tape saturates "softly" (like tubes), the distortion at saturation often shoots up to well over 50% on loud passages. Due to the soft nature of the overload, this was sonically tolerable. But the music lacked full dynamic range and had a muddy and confused quality on loud sections.[/FONT]

[FONT=&quot]But the parameter that most annoyed me was the instability of both the frequency and amplitude of the signal. Frequency variations (measured as wow and flutter) rarely were as low as 1%. The slightest mechanical flaw (dirty or worn heads, capstan shafts, or tape guides) would dramatically worsen this. Wow and flutter could easily be heard on critical material like sustained piano tones as a warbling of the sound that was very unnatural.[/FONT]

[FONT=&quot]Amplitude instability made the flutter even worse. You could play back a steady test tone as you recorded it and its amplitude would vary plus/minus 2 dB! [/FONT]

[FONT=&quot]This was highly dependent on the quality of the magnetic coating on the tape. Later advancements in tape technology, particularly the polishing of the tape surface and the use of smaller magnetic particles improved this. But even so, the best reading I ever saw was plus/minus 1 dB at the midrange frequencies (high frequencies were much worse). This was always audible to a critical listener.[/FONT]

[FONT=&quot]Then there was the problem of bias drift. Analog tape is totally dependent on a supersonic bias signal (typically 100 KHz) for achieving low noise, low distortion, and linear frequency response. As the bias oscillator heated up during a recording, the bias current would change. [/FONT]

[FONT=&quot]This would cause a tape deck that I had spent hours "tweaking" to the highest performance possible to change its performance during the recording session. This was truly frustrating as I could never get the best performance from the equipment, even though I was using the finest equipment of the day.[/FONT]

[FONT=&quot]LPs were much worse than tape. I had many pressings of LPs made for customers. The process degraded the sound at every step of the process from using a lathe to produce the lacquer master, through the metal casting, and subsequent pressing with vinyl that was always contaminated with foreign particles that caused the clicks and pops of surface noise. [/FONT]

[FONT=&quot]Note in particular that most of the rumble heard on LPs is actually recorded into the master disk during the lathe cutting process. In most cases, the rumble from the customer's turntable bearing contributes an insignificant amount of rumble compared to the large, noisy bearings in a lathe. Rumble that is cut into the disk cannot be removed by using a super quiet turntable on playback.[/FONT]

[FONT=&quot]So the resulting LP inherently had a large degree of variation from the original master tape. But even if the pressing were perfect and had no errors (impossible), when playing an LP, you have the problem of phono cartridges. [/FONT]

[FONT=&quot]Cartridges are like loudspeakers in that they are transducers and therefore there are large differences in the sound of cartridges. Between the errors introduced during the production of an LP and the variances of phono cartridges, the sound from an LP sounds quite obviously different than the sound on the original master tape -- which sounds significantly different from the live microphone feed.[/FONT]

[FONT=&quot]Although a high-speed, master tape was the best storage medium we had, it still corrupted the sound quite obviously. Everyone could easily hear the difference between the recording and the live microphone feed. In fact, every preamp of the day had a tape monitor loop so you could compare the source to the recording in real time (if you had a 3-head tape deck), and there always where differences you could hear between the two. [/FONT]

[FONT=&quot]In short, analog recording is seriously flawed and never sounds like the source. Something better was badly needed.[/FONT]

[FONT=&quot]By the 1980's, digital recording had been developed that had the potential to solve the technical problems that were insurmountable with analog equipment. Of course, with an entirely new process, there were some teething problems. Initially there were serious problems with insufficient data storage, low-level accuracy of DACs, and a weird problem with 1/3 order harmonic distortion that was eventually eliminated with the introduction of dither. [/FONT]

[FONT=&quot]But the quality of digital recording quickly progressed and by the mid '80s, it was possible to make digital recordings that sounded identical to the live microphone feed. They had perfectly linear frequency response (DC to 20 KHz +/- 0.1 dB), lower distortion than most instruments could measure (less than 0.002% THD), unmeasurable wow and flutter, unmeasurable amplitude instability, and a totally silent background (S/N of better than 92 dB). [/FONT]

[FONT=&quot]Let me take a momentary detour here and comment that most audiophiles today still believe that linear PCM (Pulse Code Modulation) such as used on CDs and that requires a DAC (Digital to Analog Converter) produce digital steps in the wave form. This is simply untrue. [/FONT]

[FONT=&quot]You need look no further than to observe the recorded wave form on an oscilloscope to see that modern DACs work so well that the wave form is absolutely smooth and cannot be distinguished from the source wave form. Even a 20 KHz tone from a CD, which has only two samples at that frequency will be perfectly formed, utterly smooth, and will have distortion of around only a thousandth of a percent. [/FONT]

[FONT=&quot]The purpose of a DAC is to produce smooth wave forms and they do so brilliantly. There are simply no steps in the wave form of a PCM recording. [/FONT]

[FONT=&quot]These audiophiles then further believe that there is some mysterious measurement called "resolution" and that higher sampling rates improve the "resolution" of the sound. This is also nonsense. There is no such measurement as resolution and there are no steps in the wave form of a PCM recording.[/FONT]

[FONT=&quot]So if the sampling rate does not improve "resolution", what does it do? The sampling rate defines the highest frequency that the digital recording system can record and store. The sampling rate in a PCM system must be twice the highest frequency to be recorded. [/FONT]

[FONT=&quot]The CD "Red Book" that specifies the performance of a CD requires a 40 KHz sampling rate. So a CD can record sounds up to 20 KHz -- the limit of human hearing.[/FONT]

[FONT=&quot]"But wait," you'll say, "CD's are sampled at 44.1 KHz, not 40 KHz." True. The additional 4.1 KHz above 40 KHz are used for the anti-aliasing filter. This filter is required to remove any frequencies above 20 KHz, which would confuse the digital converters and cause errors and flaws in the recording. [/FONT]

[FONT=&quot]A sampling rate of 96 KHz will record up to 40 KHz (requires 80 KHz sampling). The extra 16 KHz are used for the anti aliasing filter. The 192 KHz sampling rate will use the first 160 KHz to record up to 80 KHz with the remaining 32 KHz being used for the filter.[/FONT]

[FONT=&quot]So the sampling rate only defines the high frequency limit of the recording. It has nothing to do with "resolution" in PCM recordings.[/FONT]

[FONT=&quot]I have been careful to state repeatedly that I have been talking about PCM recordings. This is because there are other digital recording and playback schemes that DO have steps in them and require different techniques to correct. [/FONT]

[FONT=&quot]For example SACD does not use a DAC. It detects the difference between samples only (delta-sigma processing). Therefore, there are discrete steps in the wave form. [/FONT]

[FONT=&quot]To get adequate smoothing, extremely high sampling rates and storage of massive amounts of information are required. So SACD samples in the MHz region. It then must eliminate the tiny steps that remain (which is noise) by using noise shaping to move the noise into the supersonic region up around 50 KHz. [/FONT]

[FONT=&quot]This system works, but it clearly is inherently inferior to PCM recording. The only advantage of SACD is that no DAC is required. But this is a moot point since SACD has now been abandoned by the industry.[/FONT]

[FONT=&quot]I might also add that to enjoy any true benefits (if any exist) from the SACD medium, the musicians had to be recorded with SACD from the start and all processing must have been done in the SACD domain. But most SACD releases simply copied PCM masters onto SACD for distribution to customers. So even if SACD were perfect, it could do nothing more than present a PCM recording to its listeners. [/FONT]

[FONT=&quot]In this regard, the industry was deceiving its customers. If you pay for an SACD recording, it must be SACD at every step of the recording chain.[/FONT]

[FONT=&quot]Digital steps are also a problem for Class D amplifiers. They too sample at very high frequencies to minimize the size of the digital steps. Their wave forms must be smoothed using a Zobel network.[/FONT]

[FONT=&quot]For the Zobel network to work well, it must be precisely tailored to the load (the speaker) that the amplifier "sees." If the two are not perfectly matched, the frequency response of the amp/speaker combination will not be linear.[/FONT]

[FONT=&quot]This is a huge problem for manufacturers of Class D amplifiers because usually they cannot know the load to which the amplifier will be attached. So they produce "universal" Zobels. These may work well or not with a specific speaker system -- you just don't know until you try it and measure it. [/FONT]

[FONT=&quot]Because Class D amplifiers will not produce linear frequency response with most speakers, I don't consider them to be high fidelity devices. But they do work very well in selected applications. [/FONT]
[FONT=&quot]
For example, powered sub woofers are ideal for Class D amps because the load is known, high frequency response is not required, but high power and cool operation are. So Class D amps are an excellent choice for manufacturers to include in their sub woofers.[/FONT]

[FONT=&quot]Now turning back to digital recording, the "word length" of a digital PCM sample is the number of bits in it. So what do the bits do?[/FONT]

[FONT=&quot]They define the dynamic range and S/N of the recording. In general, you can consider a bit to be 6 dB of S/N. [/FONT]

[FONT=&quot]The CD Red Book specifies 16 bits. Therefore, the S/N of a CD can be as high as 96 dB. [/FONT]

[FONT=&quot]There are some subtle technicalities that I won't get into that alter this slightly. For example the addition of dither (very quiet white noise) will reduce the S/N slightly. I measure an actual S/N on most CD equipment of around 92 dB for these reasons. But using 6 dB per bit is a good rule of thumb.[/FONT]

[FONT=&quot]Today's "hi resolution" [sic] recordings usually are made using the 24/96 (24 bit, 96 KHz sampling), linear PCM specification. This means that the highs will extend to 40 KHz and the theoretical S/N will be 144 dB.[/FONT]

[FONT=&quot]We can't hear above 20 KHz, so doubling the frequency response to 40 KHz serves no useful purpose. And while the digital S/N may be as high as 144 dB, no analog electronics are anywhere near that quiet. [/FONT]

[FONT=&quot]The quietest analog electronics have a S/N at best of 120 dB, and Browning motion of the air molecules around microphone membranes limits them to about 92 dB. So there is nothing to be gained by using 24 bits during playback. [/FONT]

[FONT=&quot]The industry recognizes these facts and that is why the CD remains the highest quality music storage medium available. No human can hear any difference between a properly make Red Book recording and the source. [/FONT]

[FONT=&quot]Many audiophiles doubt this. But I have a standing bet of $10,000 (or any amount of money you are willing to bet) that nobody can hear the difference on a properly controlled test. I've never lost this bet. Contact me anytime for details and arrangements to place your bet and do the test.[/FONT]

[FONT=&quot]This finally brings us to answer your question. If digital recording is so good, why do many old LPs sound much more enjoyable and realistic than their CD counterparts?[/FONT]

[FONT=&quot]To answer this, let me tell you a story about the best recording of Respighi's "Pines of Rome" that I have ever heard. It was recorded in 1959 by Fritz Reiner and the Chicago Symphony by RCA "Red Seal." [/FONT]

[FONT=&quot]The LP had superb dynamics, essentially full frequency range (30 Hz to 15 KHz), great "hall sound", and a high degree of realism. It was pure joy and very exciting to listen to, despite all the obvious faults that were epidemic in LPs of that era (surface noise, distortion, wow and flutter, poor S/N, and general instability). [/FONT]

[FONT=&quot]When CD's became available years later, I couldn't wait for RCA to re-release that recording on CD so that I could eliminate all the faults heard on the LP. RCA finally did so in the late 80's and I couldn't wait to bring the CD home and play it.[/FONT]

[FONT=&quot]Boy, was I disappointed. The CD had essentially no dynamic range, no bass, many of the instruments could barely be heard, and it in general sounded like I was listening through a telephone! [/FONT]

[FONT=&quot]I was furious. I knew that digital recordings could (and should) be superb, since I was making them myself and knew this to be true. So I was determined to find out what was going on at RCA to ruin this recording.[/FONT]

[FONT=&quot]After enduring considerable hassles finding my way through the telephone maze at RCA, I finally got to those responsible for releasing the recording. After hearing my complaint, they explained what had happened this way:[/FONT]

[FONT=&quot]The original master tape recording was NOT made in 2-channel stereo. It was made using a 16 track recorder and multiple microphones -- in stereo -- on each orchestral section (violins had 2 mics, trumpets had 2 mics, etc.) They also placed mics out in the concert hall to record the sound of the hall sound.[/FONT]

[FONT=&quot]They then mixed down the 16 track tape to get a 2-channel stereo recording that could be pressed to produce LPs. The recording engineer who did this work obviously really knew his stuff and did a great job of getting the right balance between the various orchestra sections, blending in the hall sound, and maintaining nearly full dynamic range and frequency response (particularly in the bass). [/FONT]

[FONT=&quot]Although this was a mixdown, he kept it reasonably simple, and did not use compression, equalization, or artificial reverb. The performance was superb and his mix showed it off extremely well.[/FONT]

[FONT=&quot]Twenty five years later, when RCA wanted to re-release the performance on CD, they did not have the mixdown used for the LP. So they had a different engineer do another mix of the original 16 channel tape for the CD. He totally butchered the job. [/FONT]

[FONT=&quot]No matter how good the recording medium, if you put garbage in, you get garbage out. So the awful sound on the CD version of this recording was due to an horrible mix done by an incompetent sound engineer who had probably never been to a live, symphony orchestra concert.[/FONT]

[FONT=&quot]The answer to your question should now be clear. It is usually the later re-processing of an old master tape that is responsible for the poor quality of sound you hear from a CD compared to the LP. It is like comparing apples and bicycles, the recordings simply are not at all the same.[/FONT]

[FONT=&quot]Obviously this problem is not the fault of the digital recording medium, which is actually far better than any analog recording process. You should not assume that the digital medium is the cause of the problem as it clearly is not. [/FONT]

[FONT=&quot]In short, LP recordings are often far better than their CD counterparts because they were mixed in a more natural and realistic manner than what happens in a modern recording or mix. So the LP is much more enjoyable than the CD in spite of all the serious flaws and inaccuracies inherent in the LP medium. [/FONT]

[FONT=&quot]Of course, not all CD recordings are inferior to LPs. A good example is Willie Nelson's album "Stardust." It is available on both formats and was apparently recorded using the same mixdown tapes. The LP is a modern pressing with excellent quality vinyl. As a result, the CD sounds a bit better than the LP because it has none of the technical flaws that are obvious in the LP. But it is obvious that the two are more similar than different in that the actual recording is identical on both. Try them and see for yourself.[/FONT]

[FONT=&quot]This experience should drive home the fact that audiophiles need to be very cautious when making cause/effect judgments about what is heard. Audiophiles far too often make assumptions and assign fault to components or design features without actually knowing that these are the cause of what they hear. [/FONT]

[FONT=&quot]This brings up another major topic -- subjective listening techniques. But this opus is already far too long, so that discussion will have to wait for another day.

-Roger
[/FONT]

 

[kach22i]

I want a certificate of completion for reading all of this great information Roger wrote.

Thank you once again for sharing your vast knowledge and experiences.

 

[audioguy]

Great post and it helps explain why, for example, the Sheffield Drum and Track LP kills every CD version I have of it.

Thanks!

 

[Gregadd]

Well Roger that begs the question-Why did so many all digital CD's uhm...suck?

 

[Angela]

I want a certificate of completion for reading all of this great information Roger wrote.

We are going to turn a lot of these into general audio white papers. Thanks for asking such intelligent and thoughtful questions about audio to prompt Roger to put thoughts on "paper". It is amazing how much people can learn and how much BS is out there (imho to part audiophiles with their money in vast quantities).

a quick story to demonstrate:

One of the latest customers to come visit brought his $20K CD player on the plane with him. He left understanding that what he had paid for was the DAC that is one of two that are in ALL CD players and the rest is really hype. With a AB test (Roger has the whole setup built for ease of use so that listener can press a button to change the device without knowing which device is which); this customer could not tell the difference between his CD player and the one we use in our living room that costs under $200. They spent a whole day in the factory testing out all kinds of "misunderstandings" that most folks have about audio. He left a believer.

oh and he was going to listen to a pair of Sound labs after our visit. Roger lent him a Magtech to take along to listen so he could be assured of adequate power. Can you believe that? Anyway, he ended up deciding to buy a complete system from Roger.

Was there a post by Roger on what is required for true ABX testing? Anyway, these devices met all the criteria for testing and they both had the same CD playing so the switch is instantaneous which makes it easier for the listener to hear any differences than the way most of have to do it at home. Listen, stop, get up, change out gear, sit down, and listen.

We have folks fly in from all over the world to spend a day or half a day with Roger. They stay in our guest room and really spend some quality time understanding things and listening, etc.

 

[microstrip]

Roger essay is very interesting, but in my opinion oversimplifies some aspects of the situation.

You are 100% true about some symphonic music – unhappily no vinyl system can properly reproduce the finale of Shostakovich 5th symphony or Mahler 2nd and the pianissimo – bad luck that symphonies many times end with fortissimo, just at this part of the LP where linear velocity is at a minimum.

But I own both Sonny Rollins Analog Productions LP of Way Out West and the XRCD24 CD, the best of the two CD versions I own. The LP version blows away the digital version. And I can not consider that the JVC mastering team did not know what they were doing. My best quality chamber music recordings are LPs. I own some in both formats, and for most of them y, I prefer the LP.

I listen most of the time to digital by convenience and because the music I want to listen only exists in this format. I have heard great sound form digital and would love to believe that CD can be superior to LP. However, unhappily my experience goes the other way. If the technical limits of the LP are not reached during the recording and the pressing is of very good quality ( not easy to get) the LP sounds better for me in my system. These last words should be stressed – the top analog systems I heard did not sound great with digital and also the best digital systems did not show their best with analog.

Disclaimer – I own Sound Lab A1s PX

 

[Angela]

I want to make it clear that my previous post was intended to address the question of why CDs and LPs sound differently. I was not criticizing LPs or those who enjoy them. I have no doubt that you have some great sounding recordings that you like better on LP than on CD.

I note with interest that you state that you also like the CD versions of some music better than what you hear on the LP. You also state that you have heard some great sound from CDs. This confirms my point in my previous post that digital recordings do not sound inherently awful and are capable of reproducing great sound.

The main point of my previous essay was to debunk the myth that because some CDs sound badly, that the digital recording medium is to blame. This simply isn't true and those audiophiles who hold that belief have drawn a false cause/effect relationship.

I went to great pains to point out the fact that a properly done digital recording is so good that it is indistinguishable from the source. Therefore one cannot blame the digital storage medium for the poor sound that is present on some CDs. We must look elsewhere for the cause of the poor sound.

I carefully explained the technical issues involved with both formats and showed that in many (but not all) cases, the difference in sound between an LP and a CD of the same performance is that the mixdown was different. As a result, the recordings were not the same on the CD and the LP and that was the main cause of the differences that were heard.

You are correct that I simplified the situation and didn't cover all the possible reasons for the differences that you will hear between an LP and a CD. My opus was too long. So I will take this opportunity to elaborate on other differences that may contribute to the differences you hear.

Throughout the following discussion, I will be referring to "differences" in sound. This is an objective way of explaining things without making judgments on the quality of the sound. Obviously, if one recording sounds better (or worse) than another, it must by definition sound different.

Once it is determined that a difference is present, then we can then explore the causes for what we hear and make judgments on the quality. But let's keep it simple during this discussion by just listening for differences in sound.

I will also take the time to explain a lot of technical issues regarding LPs because the technology is over a half century old and therefore most audiophiles are not familiar with the issues involved. I apologize in advance if some of this will be review for some of the older readers on this forum.

For purposes of this discussion, I will assume that the recording mixdown (if one is used) will be identical. Obviously, if different mixdowns are used, then that is likely to be the major cause of any difference you will hear between formats. So we must assume identical mixdowns to examine the other reasons for the differences you hear.

The major cause of the differences you hear are frequency response differences. So let's closely examine what can and will alter the frequency response of a recording in both the LP and CD formats.

The industry standard "Red Book" CD specification states that the frequency response of a CD will be from 20 Hz to 20 KHz +/- 0.1 dB. In other words, it will have perfectly linear frequency response across the entire audio bandwidth. Because of the precision and computational nature of digital recording, all properly operating digital equipment easily meets this specification and you may safely assume that the frequency response will be flawless.

By comparison, the "RIAA" (Recording Industry Association of America) LP equalization standard operates in a limited frequency range of 30 Hz in the bass and 15 KHz in the highs. The RIAA specification specifies that the recording and playback will be extensively equalized to compensate for inherent limitations in the LP format.

As an aside, there is a similar equalization process used for analog tape. It is specified by the NAB (National Association of Broadcasters).

Specifically, during the LP disk mastering process, the highs must be significantly boosted and the bass must be heavily attenuated. Upon playback, a mirror image equalization must be applied to depress the highs and boost the bass to re-establish the same frequency response that was present on the master tape.

The purpose of boosting the highs is to reduce noise. Exaggerating the highs on the disk increases the S/N (Signal to Noise ratio) by making the high frequencies of the music substantially louder than the background noise of a stylus dragging on vinyl. When the highs are then depressed back to their original level on playback, the high noise level on the LP will be depressed by a similar amount, thus improving the S/N.

Bass frequencies contain a great deal of energy (imagine the initial impact of a bass drum). This would result in far too much excursion of the stylus on playback and cause it to jump out of the groove. Also, even if the stylus could handle the large excursion, the groove would have to cover a relatively wide area of vinyl to capture this large bass excursion, and this would greatly reduce the amount of playing time of the LP. Since the term "LP" is an abbreviation for "Long Play", it was essential to compress the bass excursion so the that the disk could play for a reasonably long period of time.

Regrettably, attenuating the bass on the LP and then boosting it on playback tremendously increases the low frequency noise. The slightest low frequency motion will be exaggerated and the result will be the well-known problem of "rumble."

I have only supplied a brief description of the RIAA equalization issue. For more complete details, I would refer you to www.wikipedia.org where you can ask for "RIAA Equalization" and get a complete history.

The RIAA equalization specification is difficult to meet with precision because it is spread over three components, all of which must to be quite accurate for the result to have accurately linear frequency response -- and none of them are. The first of these is the preamp that drives the lathe's cutter amplifier, the second is the phono preamp, and the third is the phono cartridge.

Since transistors hadn't been invented yet, both of the preamps had tube electronics, and tubes change their behavior as they age. So the equalization would drift over time.

Furthermore, since digital wasn't yet invented, the equalization had to be done in the analog domain using capacitors and resistors instead of by using computer computations such as is done with digital signal processing. It was therefore critical that the capacitors and resistors had exactly the correct values.

Precision, 1%, metal film resistors were not yet invented, so manufacturers were stuck using carbon composition resistors with a tolerance of around 5%. Even worse were the capacitors. Close tolerance capacitors are hard to produce even today. Back then, most capacitors had a tolerance of 20%.

Furthermore capacitors tend to change their tolerance with changes in temperature. They also change as they age over years. The heat generated in tube equipment caused a significant amount of drift.

So even if the manufacturer of the preamp was very skilled and his equalization components were carefully chosen, the relatively poor tolerances of the parts and the fact that they changed their values with age guaranteed that a significant amount of frequency response error would be present. The problem was worsened by the fact that several components were involved and each had errors that could add up to major flaws in the overall frequency response of the playback chain.

Modern transistor preamplifiers using precision 1% resistors and 2% precision capacitors will have much more accurate equalization than older equipment. But they still aren't accurate like digital signal processing. And many audiophiles prefer to use older tube preamplifiers so do not take advantage of the improved equalization accuracy that is available from modern equipment.

Of course, old LPs were made before transistors and precision components were available, so unless you only listen to recent LPs, the LP itself is likely to have relatively poor RIAA equalization accuracy.

I think you can now better appreciate that the RIAA equalization/mirror imaging process was fraught with error. So the frequency response in the electronics used to produce LPs was far from linear.

But that is only part of the frequency response problem. The phono cartridge is a major player here.
Phono cartridges (like speakers) are transducers and like speakers they all sound quite different. There are big differences in the frequency response of cartridges from different manufacturers. Even apparently identical cartridges from the same manufacturer will not have identical frequency response.

These frequency response differences are caused by variations in the mass of the stylus, with the compliance and age of their suspension system, and with the distances between their coils and magnets.

Yet another problem is that cartridges have a high frequency resonance. This resonance is affected by the resistive and capacitive loading on the cartridge. To make matters worse, it changes as the suspension system of the stylus ages (you should change styli every year for this reason).

To deal with this, quality cartridge manufacturers will specify the ideal input resistance and capacitance of the phono preamp to achieve reasonably linear frequency response from their cartridges. Top of the line phono preamps have facilities for changing their input resistance and capacitance to best meet the cartridge manufacturer's recommendations. But even they will have limited choices and you will not likely get these values perfect. Of course, if your phono preamp doesn't have adjustable input resistance and capacitance, then you can safely assume that your cartridge will have significant errors in its frequency response.

Yet another issue affecting frequency response is the age and wear on an LP's vinyl. Due to the tiny surface area of the stylus that is in contact with the vinyl, the pressure applied per unit area on the vinyl is immense. This produces a great deal of friction as the stylus slides along the groove. The friction causes tremendous local heating of the vinyl.

The heat causes the vinyl to soften and deform. The stylus, although relatively low in mass, has a great deal of inertia at high frequencies. The hot, soft vinyl is therefore stretched and deformed by the stylus as it passes over it. This distorts the groove and when it cools, it looses some of its shape. The result is that the high frequencies on the disk are gradually reduced in amplitude as the disk is played repeatedly. So an older, well-used disk will have lost much of its high frequencies.

Yet another factor that affects the apparent (not measured) frequency response is noise. The hiss that is present on all analog recordings gives the impression of extended high frequencies and "air" around the instruments. This is completely unnatural and inaccurate. But it may produce an effect on the music that subjectively is pleasing, even though the noise itself is not.

As you can see, there are several points where frequency response errors can and will occur in an LP playback system. Therefore, the total frequency response error can be quite high when they are all added together. I have seen over 6 dB of error in some systems, and even the best will show at least a couple of dB of error. Two dB is quite obvious to any skilled listener, and 6 dB will seriously change the sound of the music.

You can now easily understand why an LP playback system simply will not have linear frequency response. The error can vary from minor to dramatic depending on many factors.

A CD has none of these problems and will have perfectly linear frequency response that will not deteriorate over time and multiple playings. The bottom line here is that there will be significant differences in the frequency response of an LP compared to a CD -- even if the mixdown used on both recordings is identical. So you will hear a difference in the sound of the between them.

Our ears are very sensitive to frequency response differences, so you will clearly hear a difference between the two formats due to this factor. Which one you prefer is a matter of personal preference. But you can bet your first born child that the frequency response reproduced by the CD is the more accurate of the two. The inaccurate frequency response of the LP is correctly classed as "euphonic coloration."

Let me put all this in perspective and summarize the facts as follows:

1) Analog recording, particularly when done on LP, is inaccurate and has many flaws. Anybody can easily hear differences between an analog recording and the source.

2) Digital PCM recording is subjectively flawless. No human can hear any difference between a properly produced digital PCM recording and the source.

3) There is some outstanding music and recordings available on both formats. However, modern recordings typically are more heavily processed than older ones and often sound awful. Since most modern recordings are only available on CD, many audiophiles have come to believe that the it is the digital CD medium that sounds bad when in fact, it is superb and it is the recordings that are responsible for the lousy sound quality. There is much really excellent music that is only available on LP because of the simpler and more natural way recordings were made scores of years ago.

4) Wonderful music on an LP sounds great IN SPITE of the flaws of the LP medium -- not because of them.

5) Music on CD may sound awful IN SPITE of the perfection with which digital media stores music. Garbage in gets you garbage out, no matter how good the recording medium. If music sounds badly from a CD, it is not because CDs "sound bad" (they don't), but because the music recorded on them was of poor quality.

6) The differences you hear between an LP and a CD of the "same" recording is usually because the recording is not the same. Probably a different mixdown was used on each. Because you are comparing apples and bicycles, no valid comparison of the two formats can be drawn. You can usually safely assume different mixdowns were used if there is a dramatic difference between the sound of the LP and CD.

7) If the differences in the sound between an LP and CD are minor, then probably the same mixdown was used and you are hearing the frequency response differences between the two caused by the non-linear frequency response inherent in the LP recording/playback process. The CD will have linear frequency response while the LP will not. The frequency response you like best is a matter of personal preference.

In closing, let me make a recommendation that you might find wonderful -- I certainly did. Record your LPs to a quality digital medium so that you can easily listen to them and you won't ruin your precious LPs by playing them over and over.

Because digital PCM recordings are subjectively identical to the source, you will not be able to hear any difference between the music on the LP and the digital recording you made of it. So you will not suffer any loss of musical enjoyment by making and listening to your own digital recordings.

There are other reasons for making digital copies. They include the fact that most albums (LP or CD) usually have only a couple of tunes on them that you really like. So by making digital copies, you can make compilations of the best music of a particular genre off many LPs to which you can listen and enjoy song after song that you really like. And of course, digital media gives you the ability to instantly select various selections, which is far more convenient than trying to queue up LPs.

Additionally a great deal of space can be saved by using digital compilations because the media is so tiny. I have taken the music from thousands of LPs and stored them on just a few flash cards or digital tapes that take up just a few inches of shelf space.

It's really great to be able to come home after work and start listening to my "LPs" on digital format. I can then just sit down, relax, and enjoy the music without having to get up and down constantly while I pick and choose tunes from LPs that must be found, cleaned, played, cleaned again, and put away.

What digital format should you use? The most common is CD-R. But I don't like it or use it. CDs simply don't hold enough music to be useful.

A much better alternative is hard disk recorder or server. Even your computer can make flawless digital copies (use WAV format). But a computer tends to be rather bulky and is not a convenient "component" to put in your audio rack.

Nowadays, the best format is a flash recorder. These are simply awesome as they record on solid state memory flash cards (the same cards used in digital cameras).

Unlike a hard disk recorder, server, or computer, a flash recorder has no moving parts. It therefore is silent and has nothing to wear out. You can use any size flash card and as many as you wish, so storage is unlimited. A large flash card can record hundreds of hours of music. For easy identification of music, you can have a different flash card for each genre of music.

Unlike CD-R, you can copy and erase music from a flash card at will. So you can keep expanding the music of a particular genre on that flash card all you wish as you find more music that you like.

Because a flash recorder is a small computer, the music files on the flash card can be moved, edited, erased, and renamed as you would any other computer file. The music files can also be copied to your computer where you can work with them as you wish (I often make CD-R copies of music from my flash cards on my computer for customers who don't have flash recorders).

Finally, because flash recorders are totally electronic and have no mechanical parts, they are inexpensive. You can get many for less than $1,000 retail. An excellent example is the Tascam SS-R1 that sells for around $500. Marantz also makes many models.

Flash recorders are used by the professional music industry, which is why you probably haven't heard of them. No "audiophile" company makes them. So look for them in the "Pro" market. I sure wish they were available back in the "good old days" when I was hauling 80 pound, analog, tubed, analog open-reel recorders up several flights of stairs to do live recordings in concert halls!

So focus on the music. Enjoy whichever format contains the music you like best.

-Roger

 

[microstrip]

...
2) Digital PCM recording is subjectively flawless. No human can hear any difference between a properly produced digital PCM recording and the source.
...
-Roger

Another great post.

But could you explain how did you prove that "No human can hear ... " and what you mean exactly by a "properly produced digital PCM recording"? CD, 24bit 192 kHz? Apologies if you explained it in another topic and I missed it. But stated exactly as written without justification it seems to distort the line you were so well presenting.

Disclaimer 2 - I am a subscriber of the Tape Project
Disclaimer 3 - I read all TELDC and even built a small panel in the 90's! Great fun.