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tell me what you think you know about dBs Studio Monitors
Old 22nd September 2011
  #61
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Quote:
Originally Posted by Ben B View Post
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
righto
and you are not them

you do not know what they are using
so cannot predict what happens when they do the d/a for themselves
Old 22nd September 2011
  #62
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Quote:
Originally Posted by psycho_monkey View Post
No one has disagreed with you on this point. Why do you keep stating it?

Then they shouldn't be in a job.

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!
like the OP noted
there are many standards
you do not know what someone else is using
so have no way to know what they get after their d/a
Old 22nd September 2011
  #63
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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.
you know that there are also amateur studios
and noobs and mere consumers

you have no idea what results they will get from your digital signal
Old 22nd September 2011
  #64
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Quote:
Originally Posted by unitymusic View Post
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.
and a lot of wars have been started because of it
even when they speak the same language problems result

not using simple clear meaningful logical language is a sure way to end up having problems
Old 22nd September 2011
  #65
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audiogeek's Avatar
 

I feel like I'm taking crazy pills!
Old 22nd September 2011
  #66
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Quote:
Originally Posted by oldeanalogueguy View Post
not using simple clear meaningful logical language is a sure way to end up having problems
I totally agree! So failing to do so should be avoided at all costs. Fortunately or unfortunately meaning comes from context and standards. I'm just joking with you by the way, because it seems like you like to be intentionally difficult.
Old 22nd September 2011
  #67
Hey...shouldn't you be using a .997 test tone anyway? heh

j
Old 22nd September 2011
  #68
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minister's Avatar
Quote:
Originally Posted by oldeanalogueguy View Post
you know that there are also amateur studios
and noobs and mere consumers

you have no idea what results they will get from your digital signal
Which is why you should deal with Intersample Peak Potential in Mastering.

I am sensing a theme here.
Old 22nd September 2011
  #69
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Quote:
Originally Posted by fastlanestoner View Post
  • 0dB is the threshold of human hearing
Well.. 0dB SPL is the threshold at one certain frequency or in a narrow band if you will. Around 3kHz the threshold is negative at aprox. -7dB SPL.

At 20-30Hz the threshold is about 70dB SPL.

Quote:
  • 120dB is the threshold of physical pain
At some frequencies/bands yes, at others, nope.. and we also need to consider time, the duration of the sound stimuli.


Quote:
  • humans don't respond to linear increases in dB
Linear increases in dB/logarithmics? Clarification needed.

Quote:
  • we perceive higher amplitude as louder volume (dB)
Yes, a higher amplitude implies higher SPL.

Quote:
  • dB unit is logarithmic/exponential reference scale (non-linear)
Yes

Quote:
  • 3dB increas is 2x amount of volume (pressurized air)
Three dB what? And volume what?

If you move 2x the amount of air (displacement increased with a factor 2 from your sound source, for example your speaker) you get +6dB SPL.


Quote:
  • we don't perceive 3dB change as doubling a sound's volume
A 3dB change of what?

Quote:
  • we perceive 10dB as doubling of volume
Depending of type of signal (discrete tones or wider bandwith signals), frequency and SPL different the results differ. Some studies indicates that an increase of 6dB SPL is perceived as a double as strong signal (which it also is sound pressure wise).


/Peter
Old 22nd September 2011
  #70
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Ben B's Avatar
 

+3dB represents a doubling of power (Watts), acoustical or electrical. Most people do not perceive a 3dB increase as a doubling of subjective loudness.

Having had the chance to conduct several perceptual tests at multiple colleges and universities over the past 22 years, I have found that most people perceive a 6dB change in SPL as a subjective doubling of loudness.

+6dB represents a doubling of voltage or pressure. Since the ear is basically a pressure-suspension device, it makes sense that doubling the pressure exerted over the area of the tympanic membrane would come close to doubling the perceived loudness, psychoacoustically speaking.

This is not a universal finding, however. Different frequencies played back at different levels can confound these findings, where (due mostly to the Fletcher-Munson Effect), more change within a certain frequency range at certain SPLs would be needed to produce that same perceived change in loudness as a lesser change in other frequency ranges. So can bandwidth limitations imposed on otherwise full-bandwidth figures.

But, overall, with full bandwidth pink noise or music signals played back at moderately loud volume, the +6dB SPL figure always seems to win in my experience with regard to the "twice as loud" perception.

-Ben B
Old 22nd September 2011
  #71
Quote:
Originally Posted by oldeanalogueguy View Post
like the OP noted
there are many standards
you do not know what someone else is using
so have no way to know what they get after their d/a
Well, providing it's documented, it doesn't matter.

If I want you to replicate my gain staging, all I need to do is write down what standard I'm using. Then when it comes to a recall or the project is continued at a later date, it's replicable.

If someone can't follow that, then the flaw is not in the system, it's in the operator. End consumers generally can't adjust their converter lineups but since they're generally just playing back, they're only concerned with listening levels.

I don't know why you think there is a problem, with correct documentation it's possible to recreate gain staging accurately. That's all we need to do, it's just not a problem if the operators know what they're doing.
Old 22nd September 2011
  #72
Quote:
Originally Posted by oldeanalogueguy View Post
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
Like I said - you're not being clear at all, you've not defined a thing other than spouted random technical words. No-one really knows what you're arguing for or against!
Old 23rd September 2011
  #73
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Quote:
Originally Posted by psycho_monkey View Post
Well, providing it's documented, it doesn't matter.

If I want you to replicate my gain staging, all I need to do is write down what standard I'm using. Then when it comes to a recall or the project is continued at a later date, it's replicable.

If someone can't follow that, then the flaw is not in the system, it's in the operator. End consumers generally can't adjust their converter lineups but since they're generally just playing back, they're only concerned with listening levels.

I don't know why you think there is a problem, with correct documentation it's possible to recreate gain staging accurately. That's all we need to do, it's just not a problem if the operators know what they're doing.
doesnt matter to you with your gear

matters to everyone else
nobody writes it down and ships it with their tracks
you wont find it on any commercial music so you can adjust your gear to play it better

it is a problem as long as most "master"ing guys are trying for louder not better
Old 23rd September 2011
  #74
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Quote:
Originally Posted by minister View Post
Which is why you should deal with Intersample Peak Potential in Mastering.

I am sensing a theme here.
you can use ghostbusters too
makes no difference
intersample peaks dont exist
and if they did they are not a problem
unless the d/a was badly designed

what you should deal with
is making it better not louder
and stay way below 0dBFS

two problems solved at once
plus less distortion too
Old 23rd September 2011
  #75
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Quote:
Originally Posted by psycho_monkey View Post
Like I said - you're not being clear at all, you've not defined a thing other than spouted random technical words. No-one really knows what you're arguing for or against!
nonsense
i clearly used english words
and LOGIC (mathematical type)
to show the words as you use them in juxtaposition
have zero meaning in the real world
Old 23rd September 2011
  #76
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Just out of curiosity, isn't this entire thread basically the same topic as this other "why itb sucks" thread-
The Reason Most ITB mixes don’t Sound as good as Analog mixes (restored)
Old 23rd September 2011
  #77
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woodhenge's Avatar
 

Wow... Had a bad experience lately?

If you think your recordings are suffering due to these issues, you're probably looking in the wrong direction.
Old 23rd September 2011
  #78
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Quote:
Originally Posted by oldeanalogueguy View Post
a brazillion standards exist
and they are ARE IGNORING ALL THE OTHER ONES !!
Bloody South Americans, always doing things differently...
Old 23rd September 2011
  #79
Quote:
Originally Posted by oldeanalogueguy View Post
nonsense
i clearly used english words
and LOGIC (mathematical type)
to show the words as you use them in juxtaposition
have zero meaning in the real world
Not really! Hence the reason people struggle to understand what you mean, and it took 17 pages to understand you were just arguing about terminology on the other thread!

You may think you're being clear, but go back and count the number of "what are you trying to say?" posts... You might be surprised..
Old 23rd September 2011
  #80
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Quote:
Originally Posted by Ben B View Post
Having had the chance to conduct several perceptual tests at multiple colleges and universities over the past 22 years, I have found that most people perceive a 6dB change in SPL as a subjective doubling of loudness.
That is what me and others over here found out as well.


/Peter
Old 23rd September 2011
  #81
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ssaudio's Avatar
 

Quote:
Originally Posted by oldeanalogueguy View Post
nobody writes it down and ships it with their tracks
I don't know anyone that doesn't - standard practice when I was taught, anywhere I've worked and still is today. Clearly, you have no experience in this respect.

Quote:
Originally Posted by oldeanalogueguy View Post
you wont find it on any commercial music so you can adjust your gear to play it better
It's called a 'volume knob'. Said knob used to be required fairly often in the 'oldanaloguedays', but less so due to modern practice.

Quote:
Originally Posted by oldeanalogueguy View Post
it is a problem as long as most "master"ing guys are trying for louder not better
You can't just simplify the argument by blaming 'mastering guys', but given your lack of knowledge of the field in question, I can understand your use of crass hyperbole.
Old 23rd September 2011
  #82
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narcoman's Avatar
 

Quote:
Originally Posted by oldeanalogueguy View Post
Recently read an AES paper that certainly appears to make a noob error.

Help me out here and tell me where I am wrong.
(And I know there are lots of you who enjoy doing that)

In the digital domain we have dBFS
the reference is maximum digital value
(eg 111111111111111111111111) and goes down from there to lower binary values ending at all 0's with a dBFS value relating to the number of bits used (eg 16 24 32 binary).

There is no way to relate any dBFS to a voltage as there is no voltage in digital. And in analog the absolute voltage is set by the d/a converter not the digital value.

In the analog domain we have things like dBu/dBv dBV dBm etc. They all relate to a specifiec voltage in the analog domain and can be compared to others in the same domain if you consider the different references. EG +4dBu compared to -10dBv is not 14 dB higher but about 12 dB because of the different references.
[dBV reference = 1V; dBu reference about 0.775 volts]
But since both references are in the same type of units eg voltage we can calculate the difference.

In the physical domain we have dBSPL for sound pressure.

There is NO way to convert between the dB types in different domains.

Yet that is exactly what the AES paper did!. It claimed that things in the analog domain were higher in terms of dBFS.
NONSENSE! There is no correlation between dBFS and dBu or dBV. With this error the AES paper merrily claims that there is an alleged intersample peak. Clearly only if you confuse digital and analog domains and mix units that do not relate.

bash away !
Set a reference first.

Plus we've already had this chat!! Can you link to the paper old chap?
Old 23rd September 2011
  #83
Quote:
Originally Posted by oldeanalogueguy View Post
you know that there are also amateur studios
and noobs and mere consumers

you have no idea what results they will get from your digital signal
Quote:
Originally Posted by oldeanalogueguy View Post
doesnt matter to you with your gear

matters to everyone else
nobody writes it down and ships it with their tracks
you wont find it on any commercial music so you can adjust your gear to play it better

it is a problem as long as most "master"ing guys are trying for louder not better
Well, no, I don't think it does matter to anyone else.

The professionals need to know because they need to calibrate their gear to provide the same reproduction chain in case of recalls or continuing work on the same project. The ME needs to know (even though he'll be playing it out of different converters)...or does he? at one point, he needed to know because he'd be playing back from 1/2" so he needed the calibration info. Now I guess it's much less important, most MEs seem to receive WAV files so they just load them in and off they go. Their DA conversion is calibrated (and repeatable) to their analogue stages, but it doesn't matter if it's not the same as the mix room's calibration, only that it is repeatable and recordable.

The prosumer - well, most interfaces you can't change the lineup anyway, it's an all in one solution. Providing you're interfacing your preamp with your audio interface at the correct nominal level (+4dBu or -10dBv) your converters are going to be fixed at something like 0VU=+4dBv= -18dBFs or -16dBFs, so your gain staging will be good. Your output stage - well, the main outs will be fixed again, the monitor outs have a volume pot on them so it doesn't really matter, presumably the output stage and the monitor amp are gain staged correctly within the interface, and the end user is always going to adjust their own volume!

So, to the consumer? again, taking the Hi-fi enthusiast (since a guy buying an all in one just has a volume control to deal with), his CD player has a fixed nominal reference level (-10dBv) and his amp expects to see this same level. As far as I'm aware, hi-fi gear DOES have a set of standards that means that any amp with a "CD" input expects to see this -10dBv reference level, hence again the only thing the end user has to worry about is the final volume.

Of course, there will NEVER be a guaranteed "x dBFs = x dBSPL", nor should there be. What we CAN do is calculate the end SPL from a given digital input signal, assuming we know the gain staging in the chain. This is basic physics.

If you're suggesting that because there are no standards to the end user as to what any given dBFs signal level generates as an SPL reading, then the whole thing is meaningless, then I really don't know what you are trying to say! you can never control what a consumer does with their gear, nor should you try to.

Our job is to make something that translates well onto well set up equipment, regardless of the spec of that equipment. If someone's got a boomy sounding setup, my mix should sound boomy on it - otherwise it's going to sound bass light on a proper setup. It's not possible to compensate for user error or plain crap sounding systems, nor should we try to. It's impossible to make a mix that sounds flat and balanced on boomy, bass light AND truthful playback systems - all we can do is use speakers like aurotones to make sure the mixes translate as well as possible.

In practice of course, providing you make sure your mix isn't clipped (or containing intersample peaks or flutabogs (copyright narcoman), any DAC will deal with it just fine, and you'll get reproduction that is in keeping with the limits of any given system. The rest just doesn't matter.

So - remind me what the original argument is again? that dBFs means nothing once you go through a DAC?
Old 23rd September 2011
  #84
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narcoman's Avatar
 

Quote:
Originally Posted by oldeanalogueguy View Post
and a lot of wars have been started because of it
even when they speak the same language problems result

not using simple clear meaningful logical language is a sure way to end up having problems
You're concepts are actually pretty good (yes, even the inter sample peak problem being a DA design issue) but you cannot fly in the face of language used by technicians and engineers the world over or issues that cross domains - without reference and translation then you're shutting off whole areas of math concepts and computing techniques to solve or predict issues (one use of inter sample peak detection helps one predict the peaks in reference based voltages - it gives a digital model of what the reconstructed analogue wave will be like. It's a statistical model if you prefer - maybe you could call it "inferred peaks correlated with virtual referenced voltages in the real world against standardised digital sample points along an equivalent time domain measure!! ).
Old 23rd September 2011
  #85
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Quote:
Originally Posted by thenoodle View Post
Just out of curiosity, isn't this entire thread basically the same topic as this other "why itb sucks" thread-
The Reason Most ITB mixes don’t Sound as good as Analog mixes (restored)
dont know
cant say
didnt read that thread

this thread was why do they build a box with higher sampling if people claim that it is not beneficial

then it morphed into debating the big lie urban myth conventional internet whizdumb and what high school seniors know about the NON EXISTENT INTERSAMPLE PEAK PROBLEM
Old 23rd September 2011
  #86
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narcoman's Avatar
 

Quote:
Originally Posted by oldeanalogueguy View Post
dont know
cant say
didnt read that thread

this thread was why do they build a box with higher sampling if people claim that it is not beneficial

then it morphed into debating the big lie urban myth conventional internet whizdumb and what high school seniors know about the NON EXISTENT INTERSAMPLE PEAK PROBLEM
please differentiate the existence from the problem. I don't believe in a problem - but the concept has existence (even though the medium difference makes only reference possible).
Old 23rd September 2011
  #87
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Halloween's Avatar
Remember when recording was fun?
Old 23rd September 2011
  #88
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DaveUK's Avatar
No this thread was tell me what you know about dBs . same ****
different packaging
Old 23rd September 2011
  #89
thank you so much for correcting what I posted. like the title of the thread...I told you what I thought I knew about dbs.

Please feel free to correct me, I'm loving the learning here.

As far as the production notes aspect goes...I annotate everything. What devices were disabled in my system to allow better performance...even on down to a minute measurement of mic placement or what cream:sugar ratio is in the coffee I'm chugging during the session. EVERYTHING matters in order to do a proper recall.

Someone said that intersample peaks don't exist? Can that poster elaborate?

I also have some questions about what I posted:

So 0db is the threshold of human hearing depending on the frequency?

As far as response to linear increases in dB, how does that work?

We don't perceive a 3dB change in amplitude as being 2x as loud?


cheers!
Old 24th September 2011
  #90
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Ben B's Avatar
 

Quote:
Originally Posted by fastlanestoner View Post
So 0db is the threshold of human hearing depending on the frequency?
At low listening levels, our hearing is most sensitive in the 3-4 kHz range. Look up the Fletcher Munson Effect, and study the Equal Loudness Contours.

Quote:
Originally Posted by fastlanestoner View Post

As far as response to linear increases in dB, how does that work?
Think of it this way: Linear additions of decibels results in a multiplication of power, voltage, or pressure. Every decibel corresponds to a fixed percentage change of the overall quantity. For example, with each addition of 3dB, the power multiplies by a factor of two. With each addition of 6dB, the pressure or voltage multiplies by a factor of two.

Quote:
Originally Posted by fastlanestoner View Post


We don't perceive a 3dB change in amplitude as being 2x as loud?
That is correct. 3dB is a doubling of electrical or acoustical power. Doubling the acoustical power of a signal does not double its perceived loudness. While loudness is a complex topic, it can be generalized that a doubling of pressure (+6dB) corresponds to a doubling of perceived loudness in most cases (but not all).

-Ben B
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