Not everything, but you are right, much of it is also discussed in the MQA, Worse than FLAC? thread.
This is false just FWIW; when I worked at Allied Radio Shack service department back in the 1970s it was always super annoying when they left the ultrasonic motion detector on. Fortunately it had a big switch so you could just turn it off. I could hear it up the stairs before I even entered the space. It ran at 25KHz. I sure can't hear anywhere near that now!the 44.1KHz sample rate of a CD would faithfully reproduce frequencies of 22.05 KHz which is already above the threshold of even young humans.
This is false just FWIW; when I worked at Allied Radio Shack service department back in the 1970s it was always super annoying when they left the ultrasonic motion detector on. Fortunately it had a big switch so you could just turn it off. I could hear it up the stairs before I even entered the space. It ran at 25KHz. I sure can't hear anywhere near that now!
Plus DACs are way better now than they were 20 years ago! You can get really competent stuff now for less than $1K.Anyway CD is now obsolete, many excellent modern recordings are now available in 96/24, that does not suffer from bandwidth limitations.
Isn't the sample also supposed to be an analog sample?Minor correction: The Nyquist sampling criteria (not the stability criteria) is that the sampling rate must be >2x the signal (information) bandwidth. No equality. You cannot recover a signal precisely at fs/2; the signal bandwidth must be a little lower in frequency.
Bandwidth is essentially DC to 20 kHz or whatever for audio, but the "bandwidth" part is important for systems that capture or produce signals at much higher frequencies. An example is an ADC running at 100 kS/s can recover a 40 kHz signal band centered at 1 GHz, assuming the ADC has the front-end bandwidth to handle it. That eliminates a whole bunch of components otherwise needed to convert the signal all the way down to DC-40 kHz. A lot of radios, cell phones, and the like take advantage of that property.
FWIWFM - Don
I don't know what that means... Signal, whatever the signal looks like, must not have any frequency component equal to or greater than Nyquist if you want to prevent aliasing.Isn't the sample also supposed to be an analog sample?
Right. That's not the question.I don't know what that means... Signal, whatever the signal looks like, must not have any frequency component equal to or greater than Nyquist if you want to prevent aliasing.
Right. That's not the question.
When a sample is taken, a 16 bit word is used to represent the voltage value at that instant. I don't see anything in the Theorem that suggests it be that way; the Theorem seems to suggest that a 'sample' in this case a voltage, is the actual value of the voltage. If I have this right, that is why there are 24- and 32-bit codices now. The voltage of any given sample will be more accurately portrayed.
Right. That's not the question.
When a sample is taken, a 16 bit word is used to represent the voltage value at that instant. I don't see anything in the Theorem that suggests it be that way; the Theorem seems to suggest that a 'sample' in this case a voltage, is the actual value of the voltage. If I have this right, that is why there are 24- and 32-bit codices now. The voltage of any given sample will be more accurately portrayed.
Right. That's not the question.
When a sample is taken, a 16 bit word is used to represent the voltage value at that instant. I don't see anything in the Theorem that suggests it be that way; the Theorem seems to suggest that a 'sample' in this case a voltage, is the actual value of the voltage. If I have this right, that is why there are 24- and 32-bit codices now. The voltage of any given sample will be more accurately portrayed.
Isn't it the case that at -45dB, you only have 8 bits to express the signal if 16 bits expresses 0VU?On the output side the CD transport or streamer sends this digital stream (along with the clock signal) to the DAC which will decode the digital amplitude data and map that from 0 to 2 Volt output (usually, but this is DAC dependent).
Not sure I 100% understand the question but ... Each bit of resolution provides 6 dB of dynamic range. So 16 bits gives 96 dB of dynamic range while 8 bits of resolution would give 48 dB of dynamic range.Isn't it the case that at -45dB, you only have 8 bits to express the signal if 16 bits expresses 0VU?
Yes- this is what I was getting at. So isn't it true then that at -48dB you'd have half of the bits in a 16 bit word turned off? If I have this right, if any of the bits to the left of the remaining 8 were on, the word would be representing something greater than -48dB?Each bit of resolution provides 6 dB of dynamic range. So 16 bits gives 96 dB of dynamic range while 8 bits of resolution would give 48 dB of dynamic range.
Made for nice phase shifters/flangers too!BTW, the old bucket brigade delay lines where the signal amplitude was stored in capacitors is an excellent example of using the Nyquist theorem in the analogue domain.
We can hear into certain forms of hiss and 10dB is reasonable. But oddly, if the hiss doesn't have a 'natural' quality (like white noise) our ability to hear into the noise floor is curtailed. For example in power amplifiers, if feedback is poorly applied, it can result in a noise floor composed of harmonic and inharmonic (intermodulations) information. The ear has less ability to hear into this sort of noise floor (which sounds like 'hiss' when no signal; is present).Finally, we can hear, and processors can extract, signals from below the noise floor. How far below varies, but anywhere from a few dB to 10 dB or more is possible.
This is not new but it's the best explanation of jitter and digital sound I've seen on the net.My infatuation with the IFi hip dac has prompted me to learn more about dacs. I think it is the hardware that has improved. I don't know. Can anyone refer me to a good primer to a straightforward primer.
Thanking you in advance.
Greg
Good point bringing up jitter. Everything discussed above about bit depth, digital noise, sampling etc. is all assuming everything works perfectly. Of course, in the real world, nothing is ever perfect. Thus clock jitter rears its ugly head and screws up all of the beautiful theory.This is not new but it's the best explanation of jitter and digital sound I've seen on the net.
Jitter in Digital Audio Data Streams Article By Steve Nugent Of Empirical Audio
Jitter in Digital Audio Data Streams Article By Steve Nugent Of Empirical Audio - Enjoy the Music.com Review Magazinewww.enjoythemusic.com
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