Please clarify. Are you saying that sampling at 1 Hz can provide the same timing precision as sampling at 1 MHz?
Maybe yes, for a tone below 0.5Hz (1/2 sample rate)
Please clarify. Are you saying that sampling at 1 Hz can provide the same timing precision as sampling at 1 MHz?
The opposite, surely? Until everyone understands the beauty of digital audio, sensible discussion is impossible.Montgomery's law. When an ASR poster has to be pointed at Montgomery's video the quality of the thread will trend towards zero.
You'll have to use one of those rotary woofers for this. They work below 1 hz.Maybe yes, for a tone below 0.5Hz (1/2 sample rate)
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You are using a scope and a frozen graph of the signal amplitude. The hearing system doesn't have such luxury. It can't "see" even instantaneous value of a signal, leave alone a high-resolution recording of the signal over time.
How many people, to this day, listen to digital audio under a burden of doubt due to half understanding how it works?
Until everyone understands the beauty of digital audio, sensible discussion is impossible.
If it's not about the money then what is it about.
De-blurring was a process that a one time was claimed to be separate from the compression scheme. OK, so offer de-blurring by itself and if people find it of value they'll buy it.
The compression is a answer to a non-existent problem. The small amount of bandwidth it saves when balanced against the data losses over a PCM 24/96 stream doesn't compute, it's not worth it. Most people say it sounds different but is that better or worse, maybe it's the compression losses heard?
The record companies have jumped on board cause it gives them what they wanted from DRM all along, no public access to a uncompromised digital copy of the original master tape. MQA is not Master Quality Authenticated, it's something else, a bastardized version of the original master done with the excuse of bandwidth savings for streaminng, a savings not needed in 2014 and all but laughable today when compared against the video streaming going on.
It's all about the money, and cutting off public access to bit perfect copies of the original masters
Just say NO.
This has been a sad corruption of the truth, again done in the name of the almighty dollar.
If CD 16/44 is standard res, then high res has to be something greater. But not just on the distribution end, but from the mic forward. In short a high res offering should have started as a digital recording at maybe 24/96 or better and never sampled to less than that.
You can't take a master analog tape done in 1960, transfer it to 24/192 and call it a "high resolution recording". That's a scam put forth simply in the name of $, once again being able to resell all the old catalogs of music with a promise of better SQ. There's nothing on those tapes that can't be captured at Red Book.
Once again the public gets screwed in the name of record labels profits.
Paraphrasing you, "a sad corruption of the truth" is that CD was advertised for so many decades as a perfect music carrier, whereas in fact it falls short of the PCM parameters that as we now understand are essential: at least 20 bits and 120 KHz sampling rate.
The 20 bit part is defensible actually. Ironically it is from the seminal paper by Bob Stuart in J. of AES, Coding for High-Resolution Audio Systems: https://secure.aes.org/forum/pubs/journal/?elib=12986Cite badly needed here.
Google found a nice short article on this:
https://science-of-sound.net/2016/02/time-resolution-in-digital-audio/
"So for a worst case of a 10 Hz sine with -60dBFS magnitude quantized at 24 Bits resolution, we get into a range of around 4 microseconds resolution. Which is already pretty coarse, but you wouldn’t hear that sound anyway. For a more realistic 100 Hz sine at -20 dBFS we are in the range of 4 nanoseconds. By the way, the samplerate doesn’t even show up in the equations!"
It's useful to emphasize that last point: the error in reconstructing a band-limited analog input is due to quantization, not the sample rate. The timing error (I surmise) for a particular input signal would be the maximum amount you can shift the signal in time before the quantized representation would change.
The trouble with PCM is that it seems simple, but our intuitions about it (at least for those not well versed in DSP) can be very misleading. And there is double trouble when marketers are happy to exploit those incorrect intuitions and engineers who know better look the other way.
For band-limited signals, yes. The bit depth being equal, and for a signal band-limited to 0.5 Hz (your example, not mine!), the 1 MHz sample rate provides no more timing precision than the 1 Hz sample rate. The same is true for for a signal band-limited to 22050 Hz: a 768 kHz sample rate provides no more timing precision than a 44.1 kHz sample rate.
That's right. Sample rate does not determine the timing resolution. However, higher sample rates allow wider bandwidth and if you take advantage of that, then the higher frequencies will improving timing.
Your hearing system is not perceiving digital samples one by one. It perceives a continuous waveform post reconstruction filter. In that sense, the sample rate is immaterial.
The timing resolution depends on how much resolution (bit depth) you have relative to frequency of interest. The higher the frequency, the better precision you need to determine it accurately. The minimum timing is = 1/ (2*pi()*f*A*(2^b-1). "f" is the frequency of interest; A is the amplitude; and b is the number of bits. Note the absence of sampling rate in the formula.
For CD at 22.05 kHz and 16 bits this becomes 110 picoseconds which is far smaller than the sample rate of 44.1 kHz (22 microseconds).
The 20 bit part is defensible actually.
I agree with your line of thinking. However, it is only applicable to the output signal. When we are considering the real-life-signal => ADC => DAC chain, we ought to understand that a real-life-signal, if Fourier-transformed, may have frequency components beyond the range of human hearing, yet still important for representing the onsets of transients with a temporal precision matching that of the human hearing system.
You can do some comparisons from 2L.
http://www.2l.no/hires/
BTW, I've missed where they had an article with mastering engineers doing such comparisons. Can you show where it is?
Just a note regarding the alleged benefits for reducing bandwidth requirements and cost for streaming providers. Most people are happy with MP3 quality streaming. Those that arent (us) are a small subset of the market. That subset are willing to pay a premium for access to the uncompressed streams. As such the cost reducing argument holds no water. The customer will pay.