Popular Memory of Vinyl Records Probably a Shifting Memory

One phenomenon known in Psychology is, that as the years pass, memories which we have of a same thing that once happened, will change, so that, 10 or 20 years later, it becomes hard to trust those memories.

A modern phenomenon exists, by which many Baby-Boomers tend to recall their old vinyl records as having had better sound, than so-called modern, digital sound. And in total I’d say this recollection is partially true and partially false.

When “digital sound” first became popular (in the early to mid- 1980s), it did so in the form of Audio CDs, the sound of which was uncompressed, 16-bit PCM sound, at a sample-rate of 44.1kHz. Depending on how expensive a person’s CD player actually was, I felt that the sound was quite good. But soon after that, PCs became popular, and many eager people were advised to transfer their recordings, which they still had on LPs, to their PCs, by way of the PCs’ built-in sound devices, and then to compress the recordings to MP3 Format for Archiving. And, a bit-rate which people might have used for the MP3 Files could have been, 128kbps. People had to compress the audio in some way, because early hard drives would not have had the capacity, to store a person’s collection of music, as uncompressed WAV or AIFF Files. Further, if the exercise had been, to burn uncompressed audio onto CD-Rs (from LPs), this would also have missed the point in some way. (:2)

What some people might be forgetting is the fact that many LPs which were re-recorded in this way, had strong sound defects before being transcribed, the most important of which was, frequent scratches. I think, the second-most-common sound defect in the LPs was, that unless the listener had a high-end turntable, with a neutrally counterweighted tonearm, and a calibrated spring that defined stylus force, if an LP was listened to many, many times, its higher-frequency sound content would actually become distorted, due to wear of the groove.

(Updated 3/02/2021, 18h05… )

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My First Digital Audio Player

One of the facts which people have been aware of for several decades now, is that we can buy a portable player, specifically for MP3 files, and that if we do, the sound quality will not be so great.

But in more recent years, Digital Audio Players have emerged on the consumer market, that promise lossless playback of high-fidelity sound, the last part of which is just referred to as “High Resolution Sound” by now. This lossless playback-capability does not come, when we listen to MP3-Files with them, but rather, if we actually play back FLAC, or ALAC -Files.

I just bought This sort of device, which is a Fiio X1 II. One of the remarkable facts about this device is, that its Digital-Analog conversion can run at up to 192kHz, and it sports the possibility of 32-bit sound. What I assume in such a case is, that even if I was to listen to a 48kHz -sampled audio file, 4-factor oversampling would in fact take place, because the D/A converter would continue to run at 192kHz, and I’d also assume that the analog filter would stay as-is, with a cutoff-frequency around 20kHz. But because I am in fact listening to 44.1kHz -sampled sound, I also assume that the whole D/A converter is being slowed down to 176.4kHz. ( :1 )

I have this working with My recently-purchased headphones, and am listening to a mix of MP3, OGG and FLAC -compressed music. I would say that this combination has significantly better sound, than the sound-chip in my Samsung Galaxy S6 phone does. ( :2 )

When I received this DAP, it had firmware version 1.6 already installed. But, I updated the firmware to the latest, v1.7… In fact, formatting the SD card with ‘exFAT’, as well as applying the firmware update, worked easily for me, even from Linux computers. The SD Card is a Sony.

My only regret is, that I personally, don’t have the manual dexterity which would have been needed to install the supplied screen-protector properly. I had the presence of mind to pull it back off, when it did not align correctly, and to dispose of the screen-protector. So I can expect some scuff-marks in the future. :-)

Happy, with Music,

Dirk

(Updated 07/09/2018, 14h55 … )

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About the Origins of Pulse Code Modulation

The way most digital sound gets reproduced today, a sample-value that has 8, 12, 16, or 24 bits of precision gets sent to a Digital / Analog Converter, and transformed into an analog voltage. This gets repeated at a constant sample-rate, resulting in an analog signal. Usually, the bits of precision are treated by the D/A converter in parallel. And this poses the question, ‘Why is this format named Pulse Code Modulation?’

The reason this is sometimes referred to as PCM, is the fact that a technology once existed, in which the sample-bits were sent to a kind of analog circuit sequentially.

The fact was observed that when a capacitor was allowed to discharge over a simple resistor, the voltage decay curve was exponential, and that within a constant amount of time, the capacitor voltage would halve. That interval of time was then used as the timing constant for pulses, which were either present or absent in a digital stream, and if the pulse was present, a constant amount of charge was pumped into the capacitor, thus increasing its voltage again by a fixed difference.

This resulted in a signal-format in which the least significant bit was also the earliest, and in which therefore the most significant bit would be the one represented in the last pulse. The circuit was so simple, that it could be implemented with a vacuum tube. And thereby, some form of digital sound already existed for military use, in the early 1960s. However, the precision was limited to 6 bits. Also, suitable for military use, this form of digital sound might have sounded quite distorted. It was certainly not meant for music, but was suitable to tell troops in a battlefield situation what their orders were.

(Edit : )

As to how circuit-designers assured that the amount of charge added to the capacitor would be constant, I think they just relied on the anode voltage of the tube being much higher than the signal-amplitudes, so that if this voltage was allowed to flow through a cathode-resistor ‘down to the voltage of the capacitor’, a relatively constant amount of current would result. And the analog pulse-width of each pulse was also made uniform somehow.

Dirk

 

Successive Approximation

While Successive Approximation is generally an accurate approach to Analog-to-Digital conversion, it is not a panacea. Its main flaw is in the fact that the D/A converter within, will eventually show inconsistencies. When that happens, some of the least-significant bits output will either be an overestimated one, followed by nothing but zeroes, or an underestimated zero, followed by nothing but ones.

Although circuit specialists do what they can to make this device consistent, there are quantitative limits to how successful they can be. And, whether 24 bits can be achieved depends mainly on frequency. In analog circuits, voltages tend to zero in on an ideal voltage exponentially, even when there is no signal-processing taking place. So the real question should be, ‘Can 24 bits still be achieved, far above 48kHz?’

And, if we insist that the low-pass filter should be purely numeric, we are also implying that one A/D conversion must be taking place at the highest sample-rate, such as at 192kHz, while if the low-pass filter could be partially analog, this would not be required.

Dirk