Record in 24 bit with 44.1 OR 88.2 OR 192 ?

[Massive Master]

Think of it like this -- 44.1kHz exceeds the limitations (22050Hz) of basically everything in the chain (up to and including human hearing). 24-bit word allows for a dynamic range (>120dB) that exceeds the limitations of everything in the chain (and introduces no dither noise).

Except for a final delivery format, it makes no sense to go lower. But as either of those parameters exceed the typical limitations of everything else down the line, there's generally not much advantage to going higher.

Personally, I'm much more interested in the quality of the source and the quality of the gear. My AD at 44.1kHz sounds better than most I've tried at (whatever). A good source being captured by a quality chain in a quality space is going to make a far bigger impact than anything else.

Have you ever heard a really good sounding MP3...? You're basically hearing a *lossy* version of a 32kHz recording (usually, with a terrible and sloppy cutoff at the 16k limit). Yet, they can still sound totally decent and listenable - depending on the source recording.

 

[mactreouser]

Awesome! Now I'm re-ensure, source itself (before Audio Interface), the acoustic and cleanness are the key of Quality! Hallelujah...haha

 

[Farview]

The Metallica Black album was mixed down to a 16 bit 44.1k Sony DAT in 1990. (Now known for its crappy sounding converters) The DAT tape that used the 'good' converters had a clocking issue and was unusable due to all the clicking, so they used the backup which just used the internal converters in the DAT machine.

The album sold over 31,000,000 copies (as of sept, 2013) and is/was a touchstone for how that sort of music should sound.

Not sure bit depth and sample rate are the things that anyone should obsess over.

 

[mactreouser]

The Metallica Black album was mixed down to a 16 bit 44.1k Sony DAT in 1990. (Now known for its crappy sounding converters) The DAT tape that used the 'good' converters had a clocking issue and was unusable due to all the clicking, so they used the backup which just used the internal converters in the DAT machine.

The album sold over 31,000,000 copies (as of sept, 2013) and is/was a touchstone for how that sort of music should sound.

Not sure bit depth and sample rate are the things that anyone should obsess over.

Appreciate the respectable sharing. And, this seems not about technically facts, but the contents of those songs and stories. wonderful

 

[garww]

Going further, I don't have Black here, but I still grab a tune off an album to see what I can do to it, That will 24/88 for my FLAC and the original wave goes to CDRW. At this turntable, I have the SoundMAX integrated and usually the Tascam us 144mkii. One of the receivers tape loops is on 3.5mm stereo, so the integrated card get used often. The Tascam gets what ever noisemaker I have sitting off to my left to play around with - USB guitar amp and a synth, right now

 

[Farview]

Appreciate the respectable sharing. And, this seems not about technically facts, but the contents of those songs and stories. wonderful

...And that is kind of the point. When technical differences that you think you might be able to hear start taking up more of your attention than the actual content, your priorities are backwards.

If higher sample rates actually made a night and day difference that everyone could hear, there wouldn't be any discussion. No one is arguing about whether it's better to be given a million dollars or a swift kick in the nuts. It's obvious which one is better. The only time there is an argument is when the difference is small, situational or fictional.

 

[garww]

Tech facts ? You mean some numbers they print up ? ..To incite a wallet riot ?

Anyone listening to these ;
The Doors-The Doors-DSD Single Rate 28MHz64fs Download|Acoustic Sounds

What are you listening to ?

 

[drtechno]

on the older gear, 88.2K & 96K sounded better. But that is because of the dreaded filter known as pre-emphasis that was switched on at 44.1K & 48K.

 

[mactreouser]

on the older gear, 88.2K & 96K sounded better. But that is because of the dreaded filter known as pre-emphasis that was switched on at 44.1K & 48K.

That means we shall stick to 44.1 & 48?

 

[42low]

48 and 96 kHz is used for high definition equipment and professional audio

That for start says something IMO. That higher than 48 could be usefull.

And what to choose for analogue recordings edited (and mastered) digital then? So for those who mostly work analoque within their system?

As a digital signal with high sample rate will smooth the result more to the almost perfect smoooooth analogue signal much more. So then i would say a higher sample rate will save the original analogue smooth signal more.
That way i will keep the best possible quality sound to edit to the best possible quality result is what i would say. Near lossless, were each step higher would save more original sound, and where lower would be more loss.

So what would then be the best quality to work in? (I know, many discussions about this, but never about the analogue part of it)
Were i think higher is really better in this case.
So i'm really curious to substantiated answers for the analogue part. (please no non-declarative personal preferences)



Sample Rate and Bitrate: The Guts of Digital Audio | | The Stereo Bus Blog
What's the bitrate for cassette tapes? : audiophile

 

[drtechno]

That means we shall stick to 44.1 & 48?

If you have to record below 96khz, 48 khz would be the lowest sampling rate IMHO. The CD media standard is mid fi and conversion from 48K to 44.1K has advanced over the years so recording in a multiple of 44.1 is not needed any more.

48Khz is more widely used now. This includes movie sound tracks that get mixed down and formatted to the 96K formats (dolby, DTS).

But 96K dose sound better track and mix however, it takes good hardware to do that and with mixing with plugins, offloading to dsp cards, boxes and additional computers (with digital i/o) that they run instead of it all in one box seems the way to go with it. Of course real outboard can be used, but going this route, It seems that you need dedicated i/o that is not depended on the DAW computer (like Dante) because the less things that is depended on the DAW computer, the more efficient the DAW runs.

My pro friends record at 96K. The only one that didn't was Brian Carlstrom (rip) and he recorded in 88.2 because the sampling rate conversion from 96K to 44.1 was not that good in the 90's.

 

[42low]

But 96K dose sound better track and mix however, it takes good hardware to do that and with mixing with plugins, offloading to dsp cards, boxes and additional computers (with digital i/o) that they run instead of it all in one box seems the way to go with it. Of course real outboard can be used, but going this route, It seems that you need dedicated i/o that is not depended on the DAW computer (like Dante) because the less things that is depended on the DAW computer, the more efficient the DAW runs.

My pro friends record at 96K. The only one that didn't was Brian Carlstrom (rip) and he recorded in 88.2 because the sampling rate conversion from 96K to 44.1 was not that good in the 90's.

So if i understand you well higher than 48 (around 90) would be better? And you would recommend that if your system can handle that?
Can you give some technical explanation why this is?

Because for me (analogue sounds, system can handle 96khz) this would be the best recommendation? Don't i have to step back to 48khz as many advice?

 

[garww]

Listen. One 48k is not the same as another . In lower px ranges, one will pay for integrated features. There is no sense recommending 96k on a crap interface. It is there and you can test it out for value. With digital i/o, you can send all the devices to your reference D/A for comparisons. All the A/D will get your best analog out. You may have to burn DVD for a good output : ) hahaha

 

[42low]

There is no sense recommending 96k on a crap interface.

Yes offcourse i understand that recording 88khz with a 44bit IF has no sense. But that's not what i'm asking.

As i ask it for myself it has sense because i mainly use a professional interface which is also analogue (so unlimited rates), and my total system can handle a big step over 100khz too (so around 90 is no problem at all).

I ask it here because it's subject already. If it's not in it's place here and if desired i can open another topic for this question. Just say it then.
And i ask it (again) because i have never got a substantial answer on it. And i want to know (with good arguments).

 

[Farview]

As a digital signal with high sample rate will smooth the result more to the almost perfect smoooooth analogue signal much more. So then i would say a higher sample rate will save the original analogue smooth signal more.
That way i will keep the best possible quality sound to edit to the best possible quality result is what i would say. Near lossless, were each step higher would save more original sound, and where lower would be more loss.

The only thing that higher sample rates give you is the ability to record higher frequencies. It's not "smoother" at all. There is nothing not smooth about 44.1k

Check out this sticky thread Digital sampling and stair-stepping explained

 

[Farview]

Yes offcourse i understand that recording 88khz with a 44bit IF has no sense. But that's not what i'm asking.

As i ask it for myself it has sense because i mainly use a professional interface which is also analogue (so unlimited rates), and my total system can handle a big step over 100khz too (so around 90 is no problem at all).

I ask it here because it's subject already. If it's not in it's place here and if desired i can open another topic for this question. Just say it then.
And i ask it (again) because i have never got a substantial answer on it. And i want to know (with good arguments).

Analog does not equal "unlimited rates". 24 bit format has a better signal to noise ratio than any analog circuit could hope to achieve (including the analog side of the interface) 44.1k will let you capture sounds as high as 22khz. (do you really make sounds above 22khz?) The difference between 44.1k and 96k sample rate is the ability of 96k to capture sounds up to 48khz. If you think what you are recording has a lot of useful sound between 22khz and 48khz, then it is essential. If not, it isn't.

 

[garww]

hah. Well, the MFG will want you to think it is about the format. I think the toob vid has someone like Dr MIX sample a nice Prism, and I felt it was easy to hear

EDIT; "Doctor" Mix
YouTube

 

[drtechno]

So if i understand you well higher than 48 (around 90) would be better? And you would recommend that if your system can handle that?
Can you give some technical explanation why this is?

Because for me (analogue sounds, system can handle 96khz) this would be the best recommendation? Don't i have to step back to 48khz as many advice?

Some converters have better response @96K others @48K, but in the case of @96K they really have to have really good line stages to make the difference. Main stream (pro-sumer) grade converters with built in microphone preamps like scarletts, and the behringers are not going to be as great sounding compared to a burl converter with an API mic pre and a manly or a shadow hills compressor on it. If you want real quality, you get what you pay for but the division of price does not always reflect this. A while back E-Mu systems made a really nice stereo card that used the same converter ic used in Metric Halo and other nice converters. Nice thing is it was line level without the preamp and it worked an sounded like the tracks were tracked in the same studio as the other tracks. Granted the microphones, mic preamp and compressor that was in the signal chain was the same. But when it was plugged into the budget converters, you definitely tell the difference.

 

[42low]

The only thing that higher sample rates give you is the ability to record higher frequencies. It's not "smoother" at all. There is nothing not smooth about 44.1k

Check out this sticky thread Digital sampling and stair-stepping explained

About that the options are divided.
Sample Rate and Bitrate: The Guts of Digital Audio | | The Stereo Bus Blog

But that's not the discussion i'm looking for. I have no need for links with personal opinions which i can find myself, even as much 'againsts' as 'pros'. I'm asking for an answer out of reasonable technical arguments which can be checked.

Analog does not equal "unlimited rates".

It does. Read the link i gave.
Sample Rate and Bitrate: The Guts of Digital Audio | | The Stereo Bus Blog

Analog signal is smoooooth, just like the image you see to the right here. Like the real world, it just keeps going. In order to get this signal represented in the digital world, we need to measure it into little chunks by defining a rate.

And more little chunks make a smooooother curve.

But again, this is not the discussion i'm going to do again.

BTW, for that discussion this is a fun explanation.



To be clear at first. The dots are the bits, and the lines are the by electronics (transistors) implemented signal.

In 1 A you see an amount off black dots (=bits).
In 1 B these dots are connected by the reasonable lines.

In 2 A there are exact the same black dots. But i've added the blue ones to double the amount of dots (= double the bits).
In 2 B you can see that this creates a totally other line that in 1 B. But remind again, it has the exact same black dots. :wink:

2 is the actual signal (=line) in double counts. 1 doesn't reach that actual signal at all because it has losses.
If you would draw a new line for a new effect-signal over the #1 signal then the result will even have a more deviate result, which i call distorted.
To be clear: You hear the line, not the dots/bits.

This proves (slighly overdone that is ) that half the signal/dots/bit (1 A) do not make the line (1 B) as it actually seemed to be (2 B) when drawn with more information (2 A). And a more real original line (2) gives correcter effects as result.
The new effect-signal line over the deviated core will even deviate more because distortion over already distorted core (= double loss = lesser quality. So more dots/bits = more quality).

And if i would draw the analoque signal it would even have NO dots and corners, but it would be a fluent line. Analogue has no bits but a line, and a line can be seen as endless/uncountable amount of consecutively dots (= endless uncountable bits).

 

[42low]

@ garww. Thanks.
That prisme ada-8xr is a nice piece of gear.
But gives me the question why this professional gear would work on 24-bit and 192kHz if that should have no sence at all?
The prisme site also nowhere talks about 44.1khz.

"Now, the goals of ultimate sound quality, unbeatable performance and un-matched versatility have been realised in a modular multi-channel, 24-bit 192kHz-capable format."

"AES3,S/PDIF up to 192kHz. Handles & converts 2-wire formats for 96k and 192k"

"Built-in Prism Sound 'MR-X' word-mapping for lossless recording up to 24-bit, 96kHz on standard"

Looks to me like a professional statement that higher sample rates indeed are better.
Why should they otherwise have those high rates as professional goal?

Some converters have better response @96K others @48K, but in the case of @96K they really have to have really good line stages to make the difference. Main stream (pro-sumer) grade converters with built in microphone preamps like scarletts, and the behringers are not going to be as great sounding compared to a burl converter with an API mic pre and a manly or a shadow hills compressor on it. If you want real quality, you get what you pay for but the division of price does not always reflect this. A while back E-Mu systems made a really nice stereo card that used the same converter ic used in Metric Halo and other nice converters. Nice thing is it was line level without the preamp and it worked an sounded like the tracks were tracked in the same studio as the other tracks. Granted the microphones, mic preamp and compressor that was in the signal chain was the same. But when it was plugged into the budget converters, you definitely tell the difference.

Now this again is a kind of answer i'm looking for. Thanks again drtechno.

Although it isn't compleet yet i think. You mention scarletts and beringers and as far as i know these are digital. These are limited when on usb for instance, as usb has limits in speed.
I'm talking about analogue systems with high speed (HD) digitalization.