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tell me what you think you know about dBs Studio Monitors
Old 19th September 2011
  #31
Quote:
Originally Posted by oldeanalogueguy View Post
nonsense

you can relate dBu and dBv exactly
you cant relate dBFS to any voltage
because everybody can relate it to a different voltage
which makes it meaningless

tell me what dBFS value = -11dBSPL
Quote:
Originally Posted by oldeanalogueguy View Post
to see if anybody knew wht spl really was
and perhaps trap some blowhard who claims there is a relation to dBFS
Well, aren't you the big clever guy.

Yes, lots of us know what dB SPL relates to. Some of us even studied it and have degrees in the subject.



Quote:
Originally Posted by roelverberk View Post
Johnnyc has a few good (and correct) points. Let me try to bring it closer to the practical side of this theory.

The AES standard for calibrating adc's and dac's is a sinewave at 1kHz peaking at -18dBFs in the digital domain.

Many converters are calibrated this way when the come out of the factory.

Many studios are therefore calibrated to -18dBfs = +4dBu = 0dB VU. Only movie mixers have an official relation with dBSPL and Bob Katz has suggested a K-system where a dBSPL is linked (Audio CD Mastering, Mixing & Replication)

This would mean we could boost the sinewave with 18dB before the signal would clip in the digital domain. Nothing special yet.

Is this the paper you are refering to?
http://www.tcelectronic.com/media/le...per_aes109.pdf

A sinewave has no harmonics. A triangle wave for example, though, has harmonics and will cause intersample peaks when the audio is converted to analog. It is inherrent to the process of digital-to-analog conversion as the process is not linear. You will always need a lowpassfilter and/or decimation filter correcting the harmonics created by the process of digitization and it depends on the signal and on these filters how much it suffers from this process . It is also true that well designed dac's should take this into account and provide enough headroom to allow these peaks.

However, this never should never be a problem if you work with 'normal' levels, staying well under 0dBfs.

Only mastering engineers have to take it into account, as some poorly designed consumer dacs could have more of a problem with those signals.

All the best.
Very very well put. You won't get any sense out of the guy because we're not on planet OAG which has a population of precisely one, but there you go.

Anyone who doesn't understand this stuff - listen to this guy, and disregard OAG and his nonsensical ramblings.


Quote:
Originally Posted by oldeanalogueguy View Post
and nobody uses them
or they all use different ones
or theyignore them because they cause problems and want better results than the standard allows
That's funny - in all the PROFESSIONAL studios I've worked in, whether working digital or analogue, we make copious amounts of notes as to what's being used - which means that when you come to recall a mix, even if you weren't the original assistant or engineer, you know exactly what was used and how to recreate it.

Using different converter lineups is really no different to using different compression settings. Provided you document what you did, it shouldn't cause problems further down the line.

Saying "nobody uses them" is daft. EVERYONE coming into our studios uses the standard 0dBFs = -18dB VU lineup unless they request different. It's part of our job to provide that, so we can make recalls.
Old 19th September 2011
  #32
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Quote:
Originally Posted by psycho_monkey View Post
EVERYONE coming into our studios uses the standard 0dBFs = -18dB VU lineup unless they request different. It's part of our job to provide that, so we can make recalls.
A bit off topic but could you clarify how you do this? Are you doing effectively -18dbfs = 0vu = [email protected] for example by using a test tone?

Also a 2nd question, where do most consumer grade converters cal at? Is it -18dbfs = -10dbv?
Old 19th September 2011
  #33
Quote:
Originally Posted by johnnyc View Post
A bit off topic but could you clarify how you do this? Are you doing effectively -18dbfs = 0vu = [email protected] for example by using a test tone?

Also a 2nd question, where do most consumer grade converters cal at? Is it -18dbfs = -10dbv?
Can't really comment on the consumer grade converters, whenever I've used products like this it's usually not in conjunction with other hardware - speakers hooked up directly to interface for example. I would imagine even consumer level gear with balanced IO references +4dBu..I seem to remember the 002 being fixed at -16 or -18dBFs = 0VU. At any rate, connecting an outboard pre @ +4dBu nominal operating level works fine with the 002...which you could switch between +4dBu and -10dBv.

Re the desk lineups, our consoles (SSL/Neve etc) are operating at 0VU= +4dBu, so using a test tone @ 1kHz and -18dBFs from PT, we line up the outputs to read 0VU on the meters. Send (calibrated using an oscilloscope in the first instance) 1kHz tone at 0VU from the console to the groups, line up the PT inputs to -18dBFs.

Essentially, you're tweaking the PT analogue line gains to match the slight variations between the groups and line inputs on the console. Some "pro" level converters, such as the early revisions of the lynx auroras, you couldn't do this....weird omission.
Old 19th September 2011
  #34
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Quote:
Originally Posted by psycho_monkey View Post
Re the desk lineups, our consoles (SSL/Neve etc) are operating at 0VU= +4dBu, so using a test tone @ 1kHz and -18dBFs from PT, we line up the outputs to read 0VU on the meters. Send (calibrated using an oscilloscope in the first instance) 1kHz tone at 0VU from the console to the groups, line up the PT inputs to -18dBFs.
Ok thanks, confused me a bit when people refer to this because the VU meters are typically averaging rms reading type while dbfs digital is a peak reading, so wasn't sure exactly what people were doing.

If we look at some numbers with this 0dbfs ends up being +22dbu = ~13.8V peak. With typical +/-15v rail power supply rails this is getting pretty close to the edges of the analog circuitry, especially since lots of circuits don't run to the edges of the power supplies. With these levels intersample peaks (or whatever you wanna call them) can pose a very real potential problem. So we end up in a compromise between analog and digital headroom trying to decide an optimal operating level.
Old 20th September 2011
  #35
  • 0dB is the threshold of human hearing
  • 120dB is the threshold of physical pain
  • humans don't respond to linear increases in dB
  • we perceive higher amplitude as louder volume (dB)
  • dB unit is logarithmic/exponential reference scale (non-linear)
  • 3dB increas is 2x amount of volume (pressurized air)
  • we don't perceive 3dB change as doubling a sound's volume
  • we perceive 10dB as doubling of volume
Old 20th September 2011
  #36
Gear Guru
Quote:
Originally Posted by stinkyfingers View Post
intersample peking duck-sauce heh
Now I'm really confused.
Won't ducking affect the peak?
Old 20th September 2011
  #37
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Old Goat's Avatar
 

Not if the sauce is properly made. Duh!
Old 20th September 2011
  #38
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Quote:
Originally Posted by oldeanalogueguy View Post
and perhaps trap some blowhard who claims there is a relation to dBFS
Just lookin' fer a friend, eh?
Old 20th September 2011
  #39
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Quote:
Originally Posted by ToxicAvenger69 View Post
I prefer to set my converters so 0 dBfs = -16 Bels, or -18.420446688 Np, whichever best suits the music - it provides more headroom.
when i have a circularly polarised antenna i do that too
Old 20th September 2011
  #40
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Quote:
Originally Posted by roelverberk View Post
Johnnyc has a few good (and correct) points. Let me try to bring it closer to the practical side of this theory.

The AES standard for calibrating adc's and dac's is a sinewave at 1kHz peaking at -18dBFs in the digital domain.

Many converters are calibrated this way when the come out of the factory.

Many studios are therefore calibrated to -18dBfs = +4dBu = 0dB VU. Only movie mixers have an official relation with dBSPL and Bob Katz has suggested a K-system where a dBSPL is linked (www.digido.com)

This would mean we could boost the sinewave with 18dB before the signal would clip in the digital domain. Nothing special yet.

Is this the paper you are refering to?
http://www.tcelectronic.com/media/le...per_aes109.pdf

A sinewave has no harmonics. A triangle wave for example, though, has harmonics and will cause intersample peaks when the audio is converted to analog. It is inherrent to the process of digital-to-analog conversion as the process is not linear. You will always need a lowpassfilter and/or decimation filter correcting the harmonics created by the process of digitization and it depends on the signal and on these filters how much it suffers from this process . It is also true that well designed dac's should take this into account and provide enough headroom to allow these peaks.

However, this never should never be a problem if you work with 'normal' levels, staying well under 0dBfs.

Only mastering engineers have to take it into account, as some poorly designed consumer dacs could have more of a problem with those signals.

All the best.
my point exactly
you can design the d/a to avoid the problems

there are lots of standards
lots of people ignore all of them
there is no way to know what a given digital signal
will end up as in somebody elses d/a analog result
Old 20th September 2011
  #41
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Quote:
Originally Posted by Rikharthu View Post
Intersample peaks are related to digital level metering, yes? It's not about the complete set of samples that make up the audio data, it's about the samples that the metering pulls out from that data.

Imagine each "O" as a sample of audio data.

OOOOOOOOOOOOOOOOOOOOOOOOOOOOOO

Now, the metering takes "samples" from the complete audio data. These are represented by the X's.

OOOOOOXOOOOOOXOOOOOOXOOOOOOXOO

The meter tells you what level the Xs are running at, but if a peak happens in between the "X" samples (an "intersample peak"), you won't know. So you could be clipping without it showing up.

The ADC samples at 44.1/96/whatever and the metering in your DAW samples at a much slower rate.
nonsense
there is NO clipping in the anlog domain unless the d/a is badly designed. nyquist guarantees that. you get back the original unclipped signal even though you think it is peaking in the digital domain.
Old 20th September 2011
  #42
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Quote:
Originally Posted by psycho_monkey View Post
Well, aren't you the big clever guy.

Yes, lots of us know what dB SPL relates to. Some of us even studied it and have degrees in the subject.

Very very well put. You won't get any sense out of the guy because we're not on planet OAG which has a population of precisely one, but there you go.

Anyone who doesn't understand this stuff - listen to this guy, and disregard OAG and his nonsensical ramblings.




That's funny - in all the PROFESSIONAL studios I've worked in, whether working digital or analogue, we make copious amounts of notes as to what's being used - which means that when you come to recall a mix, even if you weren't the original assistant or engineer, you know exactly what was used and how to recreate it.

Using different converter lineups is really no different to using different compression settings. Provided you document what you did, it shouldn't cause problems further down the line.

Saying "nobody uses them" is daft. EVERYONE coming into our studios uses the standard 0dBFs = -18dB VU lineup unless they request different. It's part of our job to provide that, so we can make recalls.
degrees in SPL !!!!!!!!!!!!
you all must really be smart
mine are only in EE , math, & CS

since you have the degree tell me what a dBVU is ?
and what is the reference for a VU dB ?
and is it the same everywhere or do VUs differ depending on extraneous factors.
Old 20th September 2011
  #43
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Quote:
Originally Posted by oldeanalogueguy View Post
degrees in SPL !!!!!!!!!!!!
Hey man, that's so he can record some hot rockin' tuneage.
Old 20th September 2011
  #44
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Quote:
Originally Posted by oldeanalogueguy View Post
my point exactly
you can design the d/a to avoid the problems

there are lots of standards
lots of people ignore all of them
there is no way to know what a given digital signal
will end up as in somebody elses d/a analog result
Of course that is true for any type of replication, as every media has it's own non-linearities. Though if you understand the underlaying processes you can get closer to a more ideal situation.

Like mentioned before, it helps if you stay well below the clipping point (i.e. respect the headroom of a digital system) you will not suffer a lot from these intersample peaks.

If a mastering engineer keeps the dBfs peak level around -0,3dB there is little to worry about.

Better adc's, dac's and more resolution (in bits and sample rate) can further reduce the problem.

Kind regards
Old 20th September 2011
  #45
Quote:
Originally Posted by oldeanalogueguy View Post
degrees in SPL !!!!!!!!!!!!
you all must really be smart
mine are only in EE , math, & CS
Well, if you want to play cocky and take things out of context, my degree is in Music, Acoustics and Recording, from Salford University, which is one of the centres of Acoustics research in England.

I could have guessed your degrees weren't in English or Science Communication...

Quote:
Originally Posted by oldeanalogueguy View Post
since you have the degree tell me what a dBVU is ?
and what is the reference for a VU dB ?
and is it the same everywhere or do VUs differ depending on extraneous factors.
Why, aren't you able to look these things up for yourself?!

Tell you what, since whatever I say you'll no doubt twist and write a stream of nonsense about, why don't YOU tell me what YOU think a dBVU is and all the rest, and I'll tell you if everyone else agrees with you or not, and if you're able to communicate what you actually mean - something you seem to have failed at significantly so far.
Old 20th September 2011
  #46
Quote:
Originally Posted by oldeanalogueguy View Post
my point exactly
you can design the d/a to avoid the problems
No one has disagreed with you on this point. Why do you keep stating it?

Quote:
Originally Posted by oldeanalogueguy View Post
there are lots of standards
lots of people ignore all of them
Then they shouldn't be in a job.

Quote:
Originally Posted by oldeanalogueguy View Post
there is no way to know what a given digital signal
will end up as in somebody elses d/a analog result
Of course there is - it's called (as I've already pointed out) calibration.

Differing "sounds" of converters aside, I can give you a digital recording and tell you what lineup I was using, and you can recreate EXACTLY the same levels, because you send a lineup tone out of your conversion and calibrate to that. If your converter has 2dB of internal headroom - then that will be accounted for. You'll get exactly the same output voltage as I do.

If you disagree with this, please say why - in English, formulated sentences, with punctuation, not just random fragments of thought. If you can't be bothered with this, don't be bothered to reply.

At any rate - I've done it, it works in practice!
Old 20th September 2011
  #47
Deleted 6ccb844
Guest
This is all fantastic!!

But what I would recommend is not getting hung up on this sort of crap and finding the best ways to get your clients to feel comfortable, sing / play there hearts out. Capture that performance with equipment that will suit the client and then make some damn good records.

Degree or not, voltages etc. there are many engineers that would look quozeyed with what your talking about, then make some mindblowing tracks and beat you down with experience.

I did a degree in sound engineering, I know enough about the technical side of it (Even if I have forgotten a little / lot :P)... But I would rather have many years of experience any day, because there are engineers without degrees and not exactly what you would call scientific that can do what many many engineers cannot (Including myself). I strive every day to get there on a proffessional and social level.
Old 20th September 2011
  #48
Quote:
Originally Posted by Dannyvect View Post
This is all fantastic!!

But what I would recommend is not getting hung up on this sort of crap and finding the best ways to get your clients to feel comfortable, sing / play there hearts out. Capture that performance with equipment that will suit the client and then make some damn good records.

Degree or not, voltages etc. there are many engineers that would look quozeyed with what your talking about, then make some mindblowing tracks and beat you down with experience.

I did a degree in sound engineering, I know enough about the technical side of it (Even if I have forgotten a little / lot :P)... But I would rather have many years of experience any day, because there are engineers without degrees and not exactly what you would call scientific that can do what many many engineers cannot (Including myself). I strive every day to get there on a proffessional and social level.
Goes without saying, of course!

But you need a good handle on the technical side as well to be a good engineer - either that, or concentrate on the production and hire someone who does! If you can't do an accurate recall (or more importantly, if your assistant isn't able to, because no-one's impressed upon them the need for converter calibration or lineup tones) then as a commercial studio you're not going to have very good client relations - and having been there, a mix that won't recall correctly is a total vibe killer, just as computer problems are, just as badly maintained gear is.

It's not an either/or situation, and I'm certainly not saying you need to be able to design gear to call yourself a sound engineer. But you should be aware of the relationship between the different dB scales, and how they relate. I think you'll find that the very best engineers have a good handle on this sort of thing, or they hire people to make sure it's taken care of.
Old 20th September 2011
  #49
Deleted 6ccb844
Guest
I get what your saying, but nothing has changed in a long long time. It has been set discussed and put to bed a long time ago, I don't see how this is going to benefit anyone...

You either know what your doing or you don't and should find out from someone who does...

I don't agree with the teflon shoulders approach, don't just pass it to someone else.. Even if your an assitant or just someone really dedicated to music engineering and production, go find out..

It's been around for a long time, there is plenty of material on the subject from component designers to engineers.
Old 20th September 2011
  #50
Quote:
Originally Posted by Dannyvect View Post
I get what your saying, but nothing has changed in a long long time. It has been set discussed and put to bed a long time ago, I don't see how this is going to benefit anyone...
I agree. But if a board like this is worth anything, misinformation needs to be argued with.

Quote:
Originally Posted by Dannyvect View Post
You either know what your doing or you don't and should find out from someone who does...

I don't agree with the teflon shoulders approach, don't just pass it to someone else.. Even if your an assitant or just someone really dedicated to music engineering and production, go find out..
Oh, on a personal level, I totally agree - it goes double if you're an assistant. I was just making the point that even the guys who are known as top engineers even though they're not that technical, they have someone on their team who is and who deals with that side of things. Personally, I want to know and understand everything!

Quote:
Originally Posted by Dannyvect View Post
It's been around for a long time, there is plenty of material on the subject from component designers to engineers.
read a few of oldanalogueguy's posts on the subject, and please add your voice to those of us telling him that!
Old 21st September 2011
  #51
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Quote:
Originally Posted by psycho_monkey View Post
Well, if you want to play cocky and take things out of context, my degree is in Music, Acoustics and Recording, from Salford University, which is one of the centres of Acoustics research in England.

I could have guessed your degrees weren't in English or Science Communication...

Why, aren't you able to look these things up for yourself?!

Tell you what, since whatever I say you'll no doubt twist and write a stream of nonsense about, why don't YOU tell me what YOU think a dBVU is and all the rest, and I'll tell you if everyone else agrees with you or not, and if you're able to communicate what you actually mean - something you seem to have failed at significantly so far.
i can and do look things up for myself

except when they are total nonsense and dont exist
Old 21st September 2011
  #52
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Are you saying calibration standards do not exist?

As far as I can tell, no one here is debating the fact dBFS has no direct relationship to absolute voltage in the absence of calibration standards. You are correct that there is no direct "conversion" between the two, however (as I've said before), it's all about calibration.

Such standards do exist, and despite your claims, they are widely used by both equipment designers and audio practitioners throughout the world.

(From Wikipedia)
  • EBU R68 is used in most European countries, specifying +18 dBu at 0 dBFS
  • In Europe, the EBU recommend that -18 dBFS equates to the Alignment Level
  • European & UK calibration for Post & Film is ?18 dBFS = 0 VU
  • UK broadcasters, Alignment Level is taken as 0 dBu (PPM4 or -4VU)
  • US installations use +24 dBu for 0 dBFS
  • American Post: ?20 dBFS = 0 VU = +4 dBu
  • The American SMPTE standard defines -20 dBFS as the Alignment Level
  • In Japan, France and some other countries, converters may be calibrated for +22 dBu at 0 dBFS.
  • BBC spec: ?18 dBFS = PPM "4" = 0 dBu
  • German ARD & studio PPM +6 dBu = ?10 (?9) dBFS. +16 (+15)dBu = 0 dBFS. No VU.
  • Belgium VRT: 0dB (VRT Ref.) = +6dBu ; -9dBFS = 0dB (VRT Ref.) ; 0dBFS = +15dBu.
When such standards are adhered to, a relationship between dBFS and voltage is established. No, it's not a conversion between the domains; but it is a fixed relationship.

So, if you merely repeat that "there is no relationship between dBFS and voltage," I can only take that to mean that you deny the existence of calibration standards (or as you've said before, "nobody uses them"). On that we'll have to just disagree.

On the other hand, if you'd like to change your claim that there is no direct conversion between dBFS to voltage, I can agree with that. (It would fall under the category of "no sh!t, Sherlock.")

-Ben B
Old 21st September 2011
  #53
Quote:
Originally Posted by oldeanalogueguy View Post
i can and do look things up for myself

except when they are total nonsense and dont exist
go ahead then and let us know how the world according to OAG defines things. Because no doubt it's not like the rest of us define it!

Quote:
Originally Posted by Ben B View Post
Are you saying calibration standards do not exist?

As far as I can tell, no one here is debating the fact dBFS has no direct relationship to absolute voltage in the absence of calibration standards. You are correct that there is no direct "conversion" between the two, however (as I've said before), it's all about calibration.

Such standards do exist, and despite your claims, they are widely used by both equipment designers and audio practitioners throughout the world.

(From Wikipedia)
  • EBU R68 is used in most European countries, specifying +18 dBu at 0 dBFS
  • In Europe, the EBU recommend that -18 dBFS equates to the Alignment Level
  • European & UK calibration for Post & Film is ?18 dBFS = 0 VU
  • UK broadcasters, Alignment Level is taken as 0 dBu (PPM4 or -4VU)
  • US installations use +24 dBu for 0 dBFS
  • American Post: ?20 dBFS = 0 VU = +4 dBu
  • The American SMPTE standard defines -20 dBFS as the Alignment Level
  • In Japan, France and some other countries, converters may be calibrated for +22 dBu at 0 dBFS.
  • BBC spec: ?18 dBFS = PPM "4" = 0 dBu
  • German ARD & studio PPM +6 dBu = ?10 (?9) dBFS. +16 (+15)dBu = 0 dBFS. No VU.
  • Belgium VRT: 0dB (VRT Ref.) = +6dBu ; -9dBFS = 0dB (VRT Ref.) ; 0dBFS = +15dBu.
When such standards are adhered to, a relationship between dBFS and voltage is established. No, it's not a conversion between the domains; but it is a fixed relationship.

So, if you merely repeat that "there is no relationship between dBFS and voltage," I can only take that to mean that you deny the existence of calibration standards (or as you've said before, "nobody uses them"). On that we'll have to just disagree.

On the other hand, if you'd like to change your claim that there is no direct conversion between dBFS to voltage, I can agree with that. (It would fall under the category of "no sh!t, Sherlock.")

-Ben B
What he's said repeatedly is that people ignore them (who these professionals are who ignore them, I don't know...I'm not sure I've met any!) so they're meaningless. Totally fails to understand or acknowledge that once someone defines it, the arbitrary relationship between dBFs and dBu is recorded and repeatable, unless wilfully ignored. IE if you know what the f you're doing, you've got a fixed system going on.

But as you say - the other answer is "no sh1t"...
Old 21st September 2011
  #54
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Quote:
Originally Posted by Ben B View Post
Are you saying calibration standards do not exist?

As far as I can tell, no one here is debating the fact dBFS has no direct relationship to absolute voltage in the absence of calibration standards. You are correct that there is no direct "conversion" between the two, however (as I've said before), it's all about calibration.

Such standards do exist, and despite your claims, they are widely used by both equipment designers and audio practitioners throughout the world.

(From Wikipedia)
  • EBU R68 is used in most European countries, specifying +18 dBu at 0 dBFS
  • In Europe, the EBU recommend that -18 dBFS equates to the Alignment Level
  • European & UK calibration for Post & Film is ?18 dBFS = 0 VU
  • UK broadcasters, Alignment Level is taken as 0 dBu (PPM4 or -4VU)
  • US installations use +24 dBu for 0 dBFS
  • American Post: ?20 dBFS = 0 VU = +4 dBu
  • The American SMPTE standard defines -20 dBFS as the Alignment Level
  • In Japan, France and some other countries, converters may be calibrated for +22 dBu at 0 dBFS.
  • BBC spec: ?18 dBFS = PPM "4" = 0 dBu
  • German ARD & studio PPM +6 dBu = ?10 (?9) dBFS. +16 (+15)dBu = 0 dBFS. No VU.
  • Belgium VRT: 0dB (VRT Ref.) = +6dBu ; -9dBFS = 0dB (VRT Ref.) ; 0dBFS = +15dBu.
When such standards are adhered to, a relationship between dBFS and voltage is established. No, it's not a conversion between the domains; but it is a fixed relationship.

So, if you merely repeat that "there is no relationship between dBFS and voltage," I can only take that to mean that you deny the existence of calibration standards (or as you've said before, "nobody uses them"). On that we'll have to just disagree.

On the other hand, if you'd like to change your claim that there is no direct conversion between dBFS to voltage, I can agree with that. (It would fall under the category of "no sh!t, Sherlock.")

-Ben B
okay
call it conversion

and if there is no direct conversion
then NSS says there are NO INTERSAMPLE PEAKS
cause you dont know what the conversion will be
Old 21st September 2011
  #55
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Quote:
Originally Posted by psycho_monkey View Post
go ahead then and let us know how the world according to OAG defines things. Because no doubt it's not like the rest of us define it!



What he's said repeatedly is that people ignore them (who these professionals are who ignore them, I don't know...I'm not sure I've met any!) so they're meaningless. Totally fails to understand or acknowledge that once someone defines it, the arbitrary relationship between dBFs and dBu is recorded and repeatable, unless wilfully ignored. IE if you know what the f you're doing, you've got a fixed system going on.

But as you say - the other answer is "no sh1t"...
i define things that are real
not make up bogus definitions for things that do NOT exist

there are no intersamples in the digital domain
hence logically can be no intersample peaks

there are no samples at all in analog domain
hence logically no intersample peaks there either

intersample peaks are ghosts in the minds of people
who do not realise that nyquist says you get back the original signal if you do a/d/a (without making changes in the digital domain) hence since most digital samples are below peak analog voltage originally it should be no surprise that if you erroneouslyl draw the digital samples in the analog domain that they are still below the anlog peaks.
as you said NO $hit SHERLOCK
Old 21st September 2011
  #56
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Quote:
Originally Posted by psycho_monkey View Post
go ahead then and let us know how the world according to OAG defines things. Because no doubt it's not like the rest of us define it!

What he's said repeatedly is that people ignore them (who these professionals are who ignore them, I don't know...I'm not sure I've met any!) so they're meaningless. Totally fails to understand or acknowledge that once someone defines it, the arbitrary relationship between dBFs and dBu is recorded and repeatable, unless wilfully ignored. IE if you know what the f you're doing, you've got a fixed system going on.

But as you say - the other answer is "no sh1t"...
like the other guy showed
a brazillion standards exist
and they are ARE IGNORING ALL THE OTHER ONES !!
which is like NO STANDARD at all

You do not know which one they are using
and so you have no idea what their analog signal from your digital samples will be.
Old 22nd September 2011
  #57
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Quote:
Originally Posted by oldeanalogueguy View Post
like the other guy showed
a brazillion standards exist
and they are ARE IGNORING ALL THE OTHER ONES !!
which is like NO STANDARD at all

You do not know which one they are using
and so you have no idea what their analog signal from your digital samples will be.
Different standards exist for different purposes. It's not important for a music engineer to adhere to standards for film mixing, for instance. Not sure what your point here is.

When you say "You do not know which one they are using," that is an untrue statement. I always know what calibration standard I am using depending on the type of work I'm doing (music, post, etc.). So do my colleagues.

-Ben B
Old 22nd September 2011
  #58
Lives for gear
 
woodhenge's Avatar
 

I think OAG doesn't realize that a sample, being a small sliver of real time, still has a duration nonetheless. It's entirely possible for the loudest peak to be somewhere in the middle of a sample... Even though its only a fraction of a second at whatever your chosen sample rate is, those samples still contain more than just a static point in time.

Einstein actually breaks this down in his relativity papers. Seriously! heh

Of course, I'm mostly preaching to the choir here, but what the heck...
Old 22nd September 2011
  #59
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minister's Avatar
Quote:
Originally Posted by Ben B View Post
When you say "You do not know which one they are using," that is an untrue statement. I always know what calibration standard I am using depending on the type of work I'm doing (music, post, etc.). So do my colleagues.

-Ben B
And it's so hard know..... I mean you have to get a signal generator in the digital domain, and we know how hard that is. And it has to generate a 1kHz sine wave, which is well nigh impossible. And we have to set it to one of the ZILLION standards there are. Then we have to have a Multi-Meter for the analogue side with resolution enough to show us 1.228 V RMS. It is all so hard to know which one is being used, I don't know how anyone gets anything done.
Old 22nd September 2011
  #60
Lives for gear
People throughout the world speak a ton of different languages, so there is no way to communicate with another person properly. There are so many languages there is really no language.
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