CDs Vs LPs

As is usually the case, I am in 100% agreement with the post by Larryi. As an illustration of his point about dynamic compression of classical recordings, I’ll mention that I have had occasion to examine the waveforms of a couple of classical orchestral recordings that I believe have been engineered with no compression. Namely the Telarc recording of Stravinsky’s Firebird Suite, Robert Shaw conducting the Atlanta Symphony, and the Sheffield Labs recording of excerpts from Prokofiev’s Romeo & Juliet, Erich Leinsdorf conducting the Los Angeles Philharmonic. I have done this on a computer, using a professional audio editing program.

In each case I observed a dynamic range (the difference in volume between the loudest notes and the softest notes) of approximately 55 db, which I consider to be an amazing "tour de force." Correspondingly, when I listen to those recordings at my preferred volume settings, with average SPLs somewhere in the 70s at my 12 foot listening distance, the softest notes are in the vicinity of 50 db or so, with the loudest notes reaching close to 105 db at my listening position. Also correspondingly, a dynamic range of 55 db means that the amplifier must supply approximately 316,000 times as much power to reproduce the loudest notes as to reproduce the softest notes. (And 316,000 is not a typo).

I believe it is safe to say that many and probably the majority of audio systems, including the majority of systems used by serious audiophiles such as those who are members here, would not be capable of handling such recordings at volume levels that are high enough to satisfy many listeners. With the most common limiting factor perhaps being the maximum volume capability of the speakers. And perhaps in some cases, and at some times, the ambient noise level in the room.


On another subject:

Sleepwalker65 10-4-2018

@dynaquest4 and @cleeds you are glossing over the fundamental issue with going digital: it does not capture the infinite range of undulations. Rather, the process quantizes the input program material at the sampling frequency, and then stores it as a sequence of discrete samples.


What you are mistakenly thinking of, is the reverse process, taking that stored sequence of samples and generating a facsimile of the original analog program material, doing bit-sum averaging to compensate for dropouts.


Perhaps this is the missing information you needed to see why digital is inherently flawed.


Sleepwalker65 10-5-2018

@cleeds the fact is that while the data is still represented as a staircase, (lollipop diagram if you prefer), the nature of sampling means that some information is left out during digitizing and then on playback, it is artificially synthesized. That is the indisputable flaw in the process. No matter what quantization resolution and what sampling rate, you cannot escape this fundamental.

First, I’m not sure how familiar you may be with the Nyquist-Shannon sampling theorem. There is no "bit-sum averaging" involved in the process. And if by that you meant "interpolation" that is not involved either. And the reference to playback being "artificially synthesized" is a misconception. What is involved is low pass filtering, in the recording process at the input to the A/D converter, to prevent "aliasing," and in the reproduction process to reconstruct the analog signal. And the 16 bit quantization that is used in CDs can, if well implemented, provide a dynamic range of approximately 96 db (actually slightly more than that), which can be further enhanced by "dithering." And theoretically/potentially, at least in the case of an infinitely long series of samples, the 44.1 kHz sample rate can support a bandwidth of 22.05 kHz, without any "averaging" or "interpolation" or "artificial synthesizing" or information being "left out."

If you believe that the LP medium is inherently superior to the CD medium I suggest that instead of focusing on (and misapplying) theory you consider the potential side effects of the two filters that I mentioned, and probably more importantly on the engineering of the recordings, and perhaps most importantly on the quality of the circuit designs that are used in the specific equipment that is used for both recording and playback.

Regards,
-- Al

P.S: Regarding the mention of 96 db and 55 db in my previous post, to be sure it’s clear I should add that those numbers refer to two different things. The mention of 55 db refers to the dynamic range of the music (meaning the difference in volume between the loudest notes and the softest notes), on a couple of recordings having extraordinarily wide dynamic range. The mention of 96 db refers (approximately) to the dynamic range of the medium, which must be far greater than the dynamic range of the music, so that the information that is present within each note can be captured.

Regards,
-- Al

sleepwalker65"What you are mistakenly thinking of, is the reverse process, taking that stored sequence of samples and generating a facsimile of the original analog program material, doing bit-sum averaging to compensate for dropouts.'

You are confused, disoriented or misinformed provided that what is under discussion hear is as I believe it to be which is the Compact Disk Audio Standard as defined by the "Red Book" protocol as promulgated by Sony/Phillips there is no "bit sum averaging." There should be no "dropouts" unless of course there is a substantial failure, defect or fault in the playback system because the CD audio standard relies on Cross-Interleaved Reed-Solomon Coding by using 24 8 bit words and encoding them in a RS code with parity check symbols.

@sleepwalker65 said:

the fact is that while the data is still represented as a staircase, (lollipop diagram if you prefer), the nature of sampling means that some information is left out during digitizing and then on playback, it is artificially synthesized. That is the indisputable flaw in the process. No matter what quantization resolution and what sampling rate, you cannot escape this fundamental.

This is a oft repeated and persistent argument against digital audio, but it is a myth. It stems from a lack of understanding of sound, Fourier transforms, and the Nyquist-Shannon theorem.

People who argue that a discrete sampling protocol can never record and reconstruct continuous audio do not fully understand the nature of sound. Sounds are made up of waves. Sound waves are just sine wave compressions and rarefactions of various frequencies at various amplitudes. Granted, in a musical performance, there are a lot of frequencies and amplitudes, but they are just a lot of sine waves. If you apply a Fourier transform to an audio recording, you can decompose all of that noise into a collection of sine waves that have equations to describe their behavior. Because sine waves have mathematically regular behavior, once you know the frequency and the amplitude, you know everything you need. There isn’t any "information left out during the digitizing."

Since humans aren’t bats or dogs, we are only interested in waves we can hear, and for the absolute best of us, that spans the range from 20Hz to 20,000Hz. For most of us middle-agers and above, a more likely range is 20Hz to 12,000Hz. Nyquist-Shannon says that a filter can be constructed to accurately reconstruct a waveform from a stream of discrete samples if the sampling rate is 2 times the desired highest frequency. This isn’t magic, it is math. And it is why the CD Redbook standard of 16/44.1 is all that is really needed. Every rigorous listening test I have ever seen has shown that for the vast majority of listeners, even trained and professional listeners, it is very very difficult to tell the difference between a 16/44.1 recording and a 24/96 or 24/192 one.

Fourier analysis has been around since 1822, the Nyquist theorem was formulated in 1933, and Shannon’s extension was published in 1948. So these aren’t new ideas. Until the advent of the digital computer, applying these ideas to audio wasn’t practical. But since 1982, it has been practical, and over the last 36 years the technology has been refined.

If people prefer the sound of vinyl LPs to CDs, that’s fine. But they should stop justifying their preference by perpetrating myths.

To all who believe that CD sacrifices none of the original program material, I ask this: if it was not recorded in 100% entirety, how can you be 100% certain that the program is PERFECTLY reproduced? Here’s the issue: what comes out cannot be superior to the data it was based on.