[jcx]

the op wants to claim that due to Nyquist digital "is only an aproximation" - I claim this is nonsense when you appreciate that real analog is also an aproximation - finite time record and limited bandwidth, S/N makes the digital math work in the real world - these limits of any signal chain mean you can't record "more" audio information with with good anaolg vs todays best digital

 

[poikkeus]

Quote:

the op wants to claim that due to Nyquist digital "is only an aproximation" - I claim this is nonsense when you appreciate that real analog is also an aproximation - finite time record and limited bandwidth, S/N makes the digital math work in the real world - these limits of any signal chain mean you can't record "more" audio information with with good anaolg vs todays best digital

JPT - ts (!) 

 

[EthanWiner]

Quote:

real analog is also an aproximation

No kidding. Some people obsess over infinitesimal amounts of digital jitter, ignoring that the flutter of even the finest analog tape recorder is literally 1,000 times worse.
 
--Ethan

 

[JerryLove]

Quote:

While today's best hi-res digital recording may be state of the art, a real world recording session may or not be hi-res. That's why not all digital is equal, though theoretically it should be. Older digital recordings in fact do not subscribe to current standards. The fact that the discussion references high res recordings suggests that there are different standards or recordings - analog and digital. 

Older analog recordings in fact do not subscribe to current standards either... unless those include wax cylinders and shouting into a cone. Perhaps I don't understand the point you were trying to get to.

 

[poikkeus]

I understand that some may not have a wide experience with analog and digital. Don't worry. What you don't know isn't going to kill you. Stick with beeswax cylinders. 

 

 

[xabu]

This is a science thread about sound (Headphones do reproduce sound too).
 
My premise was:
Analogous as well as digital recording and playback of sound (in German I would say "Schall" in this context) is lossy.
 
Based on  this premise I thought it would be possible to discuss which method might be the superior one today to reproduce an original sound by recording and playing it back (which means reproducing the sound by other means than the original sound producing instrument). The analogous/mechanical method or the digital method.
(What any individual might sense as better or "the Best" reproduced analogous sound in the end is not the question here).
 
Or is this differentiation perhaps not that important today due to countless practical factors and Heisenberg etc.
(What any individual might sense as better or "the Best" reproduced analogous sound in the end is the question here).
 
 
 
A  test arrangement could be e.g.:
 
1. An instrument which emits a sound.
1. A microphone which receives this sound (We need an "analogous" instrument to capture sound).
2.This sound is recorded on Tape or best available mechanical/analogous medium as well as digitally on a SSD or best available equivalent medium.
3.The sound is than played back from these media by mechanical/analogous respectively digital means.
4. The played back sound is emitted via headphones or speaker (We need an "analogous" instrument to emit sound).
 
Now, would it actually be possible to compare the sound emitted by the headphones or speaker to the original sound technically?
 
A test equipment would have to be superior to digital as well as analogous ... wouldn't it ?
 
 

 

[JerryLove]

Quote:

Originally Posted by xabu /img/forum/go_quote.gif
 
Now, would it actually be possible to compare the sound emitted by the headphones or speaker to the original sound technically?

Not without moving the results to the digital domain. Even if you could: separating quality from quantity is a challenge.

 

[xabu]

Well, if we agreed on my last assumption, and if we concluded that the currently available "digital domain" based sound reproduction testing equipment would be sufficient to test "analog" and "digital" wouldn't that also lead to the conclusion that "digital" reproduction had to be technically superior to "analog" reproduction.
 
(...for the record ... most of my life I listened to vinyl and I like the sound ... I even had the opportunity to listen to original shellac records... but with the advent of current technology developments ... vinyl still sounds good ... but the technical and social possibilities of the "digital domain" are far superior ... and regarding sound ... assuming good reproduction systems and healthy media in both domains it is just a difference in sound signature to me ... like different headphones have different sound signatures ...)

 

[brconner]

Digital reproduction has the ability to be substantially better than analog reproduction. Digital recording techniques are capable of having a much higher bandwidth than analog ones and their distortion ratings considerably lower. Most audio analog devices are limited to around 20kHz to 40kHz whereas with digital (if you wanted) can go to GHz. From an engineering standpoint, the faster you can convert analog to digital the better. Digital processing is less lossy and doesn't have those pesky phase shifts that analog processing will have. 
 
There are some good explanations of why analog reproductions sound better. Some distortions are actually very pleasing to the ear. "Analog" distortions tends to be from more even harmonics which sounds "musical" and "fuller". "Digital" distortion tends to be higher in odd harmonics which sounds "harsh" and "sharp".
 
Another interesting effect is channel separation. It's very easy for digital recording and playback to have better than 90dB channel separation, which for human hearing is basically complete separation. Tape and vinyl aren't anywhere close to that number. Those mediums are more in the range of 20dB to 50dB of separation. In real life, we never hear sounds that are completely from one side. It may sound like it, but the other ear is always getting a bit of that signal as well. When you're using speakers and not headphones this generally isn't an issue because the sound will be mixed in the air before it reaches your ears. Headphones on the other hand won't mix the two channels for you. So sometimes a digital mix won't sound as realistic and natural because of the channel separation. The analog mediums naturally mix the channels more which is reproducing the sound more similarly to how we'd actually hear it from a speaker or in real life.

 

[Syan25]

I record music in a studio and have worked with studio techs most of my life. I agree with the above. Digital formats are for working with sound mostly. For recording and listening - of course analog reigns supreme. Thank for for DACs...
 
We avoid compression when recording and try our best to NOT compression anything - a unique trend among studios...but more and more are going that way...
 
In this way - analog is then converted to digital to deal with the editing process - and finally - formatting to CD...
 
However, I guess you are all aware of that already...
 
Anyway - compression in the studio has been a main killer of SOUND over the 1990's...and it has ruined a lot of amazing acoustic recordings in particular...
 
Regardless of what you get from FLAC or ALAC...if the studio ruined acoustic sound with compression - there is no way to get that back...
 
H

 

[JerryLove]

Quote:

I record music in a studio and have worked with studio techs most of my life. I agree with the above. Digital formats are for working with sound mostly. For recording and listening - of course analog reigns supreme. Thank for for DACs...
 
We avoid compression when recording and try our best to NOT compression anything - a unique trend among studios...but more and more are going that way...

I deal with the flow of data for a living, including data integrity, and I don't understand what you seem to be inferring.
 
Recording is done in the format of the original waveform. If I am recording an acoustic guitar: the guitar dictates that the sound start analog. Since I must record (mic) prior to any change, it is neccessairily analog. But I might as well convert to digital right there at the input before committing to master (I'd start with a HDD).
 
Compression (the reduction in number of bits to hold data through use of algorethim, as opposed to "dynamic compression", the reduction in dynamic range) does not affect anything at all other than the overhead involved in processing. Unless the compression is lossy, and I assume we would be discussing lossless compression, there's no affect at all on the sound.

 

[Syan25]

Oh - right - well I am talking about dynamic compression - of course. In the studio - we simply refer to it as compression. But it means dynamic and consequently tonal compression. Sorry for not being so clear..

 

[Jodles]

Quote:

That said, there's a common misconception regarding the Nyquist theorem.
A sampled (and digitally recorded) sound wave could be restored equally without information loss ONLY if the samples would be taken endlessly for a given signal. That's never going to happen obviously ...

 
Source?

I'm not trying to refute what you said, but studying audio engineering I'm genuinely interested. Mathematically, I see no reason why it should not be possible to record and reproduce all frequencies below half the Nyquist rate perfectly (in theory) and unambiguously. Of course if you're talking about the whole frequency spectrum then I'm with you.

 

[jcx]

Not likely to learn much more than basic Trolling technique from the OP - he just keeps repeating the usual sophomoric understanding of Fourier Series, Nyquist theorem - which doesn't apply to digitally sampling finite time, bandwidth limited signals, with limited amplitude, unavoidable noise ("Brownian" motion of air, Johnson, "shot" noise in electronics)
i.e. we don't have to "perfectly" represent a series of infinite sines to reproduce real music performances - which don't have "infinite" duration (subjective impressions of high school band concerts aside)
we also don't try to capture/reproduce signal at the Nyquist frequency - we always limit signal frequency content to Less than Nyquist by filtering before the ADC
 
a much better perspective on the limits is the Shannon-Hartley Channel Capacity theorem - all parts of the recording-playback chain can be modeled as "information channels" - with hard limits
 
recorded audio signal bandwidth is 1st limited by the recording microphone's 2nd order roll-off at any where from ~15KHz to at most 50KHz - the latter being very rare in practice
 
then preamp and ADC front end analog electronics further filters the signal to remove signal content above Nyquist - modern oversampling audio ADC input sample rates are MHz - allowing lots of room for even low order anti-alias filters
the ADC then digitally filters and decimates to the final bit depth/sample rate output format

 

[Jodles]

jcx: Thanks for the explanation! That matches what my lecturer said and what I've read in books. I should stop using the internet, seriously. Just confuses