I guess my other thought is that analogue recording engineers just EQ’d things in a very different way to take advantage of the limitations of vinyl. Creating a very different (not necessarily more detailed) sound, with more emphasis around the EQ points of the RIAA curve as that is where the emphasis / de-emphasis takes place. Thus creating a ‘vinyl sound’. Data density wise, and this has to be a very real consideration, vinyl has a similar data density to CD. All things been equal so vinyl is unlikely to contain significantly more data than a CD, so other factors must influence the ‘warmth’ of vinyl.
Yes Analogue is theoretically infinite,with infinitely short risetimes, ....until it goes into a bandwidth limited network, then the picosecond time resolutionjust vanishes ,and gets smeared as the impulse response can only change with the time constant of the network..
Isn't the human the human hearing architecture, by its very design from the pinner all the way through to the synapses and neurons a band-width limited network? Especially the cochlea which is specifically designed to process auditory data within the 20hz to 20 khz spectrum? These are the frequency range that the (inner ear) cilia are designed to respond to? Sub 20Hz sound is the remainder of the body perceives sound pressure waves, Similarly above 20 khz its more about cranial vibration.
The body has some strange responses to sound pressure waves outside its hearing range. in the 60s the US military experimented with Ultra Low Frequency sound cannons, which generated high intensity sound waves aroundthe 2Hz and 3Hz frequencies by causing interference waves using a pair of MASERS at offset frequencies. These were intended for crowd control, by causing the rioters to defecate themselves (a side effect of exposing the human body to ultra low frequency sound) unfortunately it also causes less desirable (read destructive) effects on key organs, which is why it was eventually abandoned.
Digital here is only considering CD. How does an original DSD recording on SACD or even higher res download compare?
Is this software and the associated research why we are seeing microtiming presented as an issue for hifi?They have devised software that can measure musical microtiming, and are making the case that it has psychoacoustic properties that strongly influence listening.OK, its not news that no one wants to listen to a crappy drummer.
LARA
/proxy.php?image=https://i1.rgstatic.net/publication/237423294_Music_on_the_timing_grid_The_influence_of_microtiming_on_the_perceived_groove_quality_of_a_simple_drum_pattern_performance/links/551d1e690cf2a15336260d3a/largepreview.png&hash=1356ccc5fdd8e5e87416a02c260206b7&return_error=1 (PDF) Music on the timing grid: The influence of microtiming on the perceived groove quality of a simple drum pattern performance PDF | In general, microtiming is considered to be an important factor for the perceived quality of rhythms. Our experiment analyzed the influence of... | Find, read and cite all the research you need on ResearchGate /proxy.php?image=https://www.researchgate.net/apple-touch-icon-120x120.png&hash=2c65958a8244216b1b458864733bf8e7&return_error=1 www.researchgate.net
In the researchgate link, the timing shifts are in the/- 25millisecond range.Haven't found any audio research yet that delves into the picosecond timing range, but I'm guessing that is the limits of human neurological function, not the limit of structures like our ears, skin/fascia, etc. that are the physical interfaces to sound.
The unavoidable flaw in using computer software to analyze music microtiming is the music must be recorded by conventional methods... step one of the Circle of Confusion.And put in the "fake" digital domain to boot.
If you are thinking in terms of frequency response, RIAA curves and samples etc. you are sort of missing the point of the microtiming article. It's about the musicality of the track rather than the accuracy.
From what I understand, the speed of a single neuron is not the limiting factor here, but the interaction of hundreds or even thousands of them at very slightly different times is what makes the difference.
I'm not sure how you could measure this with any ease. Double blind testing?
I am not sure how that works. CD and vinyl have a similar data density, so if there is extra information that makes vinyl more musical (even if we cannot perceive it cognitively) where is it coming from? It can only either be at the expense of missing data elsewhere. Ipso facto vinyl has a less audible data and the data it has is less accurate than CD? Microtiming becomes irrelevant at this point or it is psycho-acoustic effect which means that it is generated in the brain and is not actually present on the media?? Which again makes vinyl an inferior medium for faithful reproduction.
Its the equivalent of saying that a laserdisc is a more natural form of Movie reproduction than a 4k Blue Ray as there is some imperceivabledata that we cannot see (or measure) on the analogue laserdisc that makes it truer ?
I understand the what the Author of the the paper is positing but he is doing it with a very narrow dataset that does not take into account many of the other real world variables present.
You cannot compare laserdisc, as it's much lower resolution, but what about 35mm film?
Just a few random observations -
Super Session - Bloomfield, Kooper, Stills - I have this on both Vinyl and CD. The Vinyl is a very old original pressing album that I bought used. It is in rough shape, tremendous noise due to scratches and abrasions, but you can listen past that and hear that the music itself is bold and exciting. Because the Record is in such bad shape, I bought a recent release of Super Session on CD. It is dull and lifeless. Yes the music is there, but the excitement is gone.
So, who or what is to blame for this? It is certainly not CD that is the problem, it is the people who mixed the sound, and squashed all the life and dynamics out of it.
Again, relative to Analog vs Digital, it is not the format, it is the content that is the greater problem.
Next, someone mentioned the space between samples, but for an change in the music to occur between samples, it has to be at a frequency well above the 20khz cut off. Bearing in mind that at normal levels, you probably can't actually hear 20khz, and if you are over 30, you will be luck to hear over 16khz at normal levels.
https://www.avforums.com/styles/avf/logo.og.png Frequency Response Test of your System I stumbled across this by accident, and though some people might find it helpful.This is intended to be a test of Headphones. But it works for any system -The Ultimate Headphone Test!!! - YouTube - First it sweeps down from 200hz to 10hz. My speakers are rated at 64hz on the low end, yet... https://www.avforums.com/styles/avf/logo.og.png www.avforums.com
Set the volume at a comfortable and FIXED level, then without changing the Volume, listen to the frequency sweep and see what you can actually hear.
Then someone mentioned our mind filling in the gaps between sound, except in reality there are no gaps. A very simple basic Digital to Analog Converter will produce stair-steps, but modern DACs have tremendous computing power behind them, and the resulting signals are smooth.
I have Graphics that would help me illustrate this, but the new Forum Software, as of yet, does not give me access to the dozens of graphics I have stored on the forum in my attachments. So, we work without them.
I have concerns that the most simply and basic Digital to Analog Conversion can not resolve the Amplitude or Phase of high frequency signal, because with standard 44.1k Sample Rate, at 20khz you are only taking 2.2 samples per cycle (I have graphics on this). That is not enough to know for sure where a sinewave begins, nor to accurately measure the amplitude right at the peak. Once again, Science and Computing power seem to do a good job of anticipating these problems, and over coming them.And ... as pointed out, very unlikely that you can hear 20khz without cranking the volume up.
Higher Sample Rates can help solve this problem too. A 96k, you are taking 4.8 samples per cycle at 20khz. With 192k, you are taking 9.6 samples per cycle at 20khz. Either make reconstructing the signal at that frequency easier. Though again, there is a very considerable amount of computing power on the job reconstructing the signal. But with more samples, the job of reconstruction become easier.
But ...all that said... I stand by the statement that it is not the compromises of the Format that matter, but rather the compromises that were made in the content. A bad mix in the absolute best format is still a bad mix.
Regarding SACD. - DSD, the file format for SACD, uses 1 BIT / 2.8224Mhz. There is really no way to translate that to a PCM equivalent, but, though estimates vary, most put it at about 20b/96k.
Keep in mind that the original 16 bit rate gave you 65,536 samples across the working voltage range. If for the sake of this example, we assume the working voltage range was 5v, then each sample is capable of resolving in units of 76 micro-volts.
20 bits gives a 1,048,576 sample range. Again across a 5v range, that resolves to 4.8 micro-volts. That is to say, changing sound can be measured in 4.8 micro-volt increments across the working range.
24 bits give a 16,777,215 sample range. Across 5v, that is 0.3 micro-volts.
32 bits, 4,294,967,296 across 5v give you the ability to resolve 1.2x10^-9 volts.
So, the amplitude sample resolution is generally not a problem.
But ...in might view... at higher frequencies, low sample rates could potentially be a problem relative to accurate phase and amplitude. But as mentioned, this is usually offset by massive computing power. And as also mentioned, you probably can't hear those higher frequencies anyway.
Again, just a few general thoughts.
Steve/bluewizard
Fair comment. need to chew that one over