Old 11-08-2010, 05:11 PM   #41
Lawrence
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Originally Posted by jnif View Post
Can you show some measurement data where distortion caused by signal below clipping level is shown? And of course in this case the recorded signal has to be a real signal from some mic, instrument or mixing/effect device. Not some unrealistically low or high level signal that has to be amplified or attenuated unrealistically in the pre-amp.
jnif
I don't think it's necessarily speculation. Just look at the specs of some of them on the lower end. It mostly depends on two things... how the preamp is designed, and if it has it's own converter circuit, which of course can be calibrated a little higher, to -15 or so to close that gap to make it less likely.

Some cheap pres will start to clip internally at +14dBu (a good bit below digital 0, the -20 calibration anyway) and others like my M101, not until +26, so I'd be clipping in digital well before the preamp clips. Even then, the warning (clip light) is set at +16.

With some standalone prosumer ASIO devices (pre + converter) I think that may be why they calibrate them a bit higher than the standard pro level of -20, to get the clipping point of the internal electronics closer to digital 0, close that gap... knowing the audience will be much less likely to be informed and thereby leading to better recordings (for that group in general) than calibrating the AD at -20 while the pre is clipping at +14 internally.

Just because home recordists can't always hear the early stages of analog clipping or unwanted distortion doesn't mean it's not happening. It's not as obvious as digital clipping. I suppose you could run sine waves through some cheap pres at -0.01 to a daw to see what happens... see if a particular pre is prone to distortion or clipping at those levels or not.

I do think many, like ART, have generally started to give their prosumer pres lots more internal headroom though. So not a major issue there at least.

Yep's only point was that there's nothing to be gained - only potentially lost - from doing that anyway so it makes no sense either way. But most people doing it (imo, with standalone pres) aren't clipping the pre, they're just gaining up the output.

Last edited by Lawrence; 11-08-2010 at 05:54 PM.
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Old 11-08-2010, 05:23 PM   #42
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Originally Posted by Lawrence View Post
Just send a steady tone into a preamp, so that the VU meter needle is sitting right at 0, send it to the DAW, with no trim or gain on the output if it has an output trim knob and see where it lands on your DAW input meter.

It should land at -20, -18 or -15 depending on your converter hardware. If it lands somewhere nutty like -10 your analog gain staging is probably off somewhere. If tracking through a console or similar the path(s) should be calibrated all the way through to the daw where you set input fader and maybe bus output fader to unity gain and hit that mark from a 0 reading on a VU meter.
Thanks Lawrence. Appreciate the help!!
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Old 11-08-2010, 06:10 PM   #43
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http://www.msr-inc.com/downloads/pdf...system_daw.pdf

http://www.homestudioguide.com/HowTo...ingLevels.aspx
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Old 11-08-2010, 06:41 PM   #44
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Originally Posted by PitchSlap View Post
Individual tracks.

By the time things get to the master, if you have any problems it's already too late. Reapers FX all all pre-fader, so lowering the master fader doesn't help if there is distortion somewhere in there.
Unless of course, if I am following correctly, I have one of those "vintage" plug-ins on my Master fader. Not that I would use processing there, but conceptually...
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Old 11-08-2010, 10:14 PM   #45
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I'm reading this with great interest, and the discussion about dBu/dBFS got me to look up teh specs of my interface (Lexicon Lambda). So, for this to make more sense to me... could someone please explain what the "Maximum Input Level" and output levels below really mean in relation to Reapers metering? Thanks.

Code:
Microphone Inputs	(2) Female XLR Pin 2 Hot
Input Impedance		600 ohms balanced
Phantom Power		+48 Volt
GAIN			+44 dBu
EIN			-115dB A-weighted @44dB gain (150 Ohm source impedance)
Maximum Input Level	+6.5dBu
Frequency Response	+0, -0.5 dB 20 Hz - 20 kHz, ref. 1 kHz
THD+N			<.005%, 20 Hz - 20 kHz

Insert Inputs:		        (2) 1/4" TRS
Send Level (tip):		+10 dBu maximum
Maximum Return Level (ring)	+11 dBu maximum

Line Inputs		(2) 1/4" TRS balanced or unbalanced
Input Impedance		20 kOhm balanced, 10 kOhm unbalanced
Maximum Input Level	+13 dBu
Frequency Response	+0, -0.5 dB 20 Hz - 20 kHz, ref. 1kHz
THD+N			<.009% A/D, 20 Hz - 20 kHz

Instrument Input	(1) 1/4" mono jack
Input Impedance		1 MOhm unbalanced
Maximum Input Level	+8.5 dBu
Frequency Response	+0, -1 dB 20 Hz - 20 kHz, ref. 1 kHz
THD+N			<.0125% A/D

Line Outputs		(2) 1/4" TRS balanced or unbalanced
Level			+16 dBu maximum
Impedance		1 KOhm

Headphone Output	(1) 1/8" stereo jack 25 mW per channel at 50 Ohms

MIDI Interface		5 pin DIN connectors for MIDI in and MIDI out
Sample Rate		44.1 kHz or 48 kHz (determined by computer application)
Dynamic Range		A/D (24 Bit) 104 dB typical, A-weighted, 20 Hz - 20 kHz
			D/A (24 Bit) 96 dB typical, A-weighted, 20 Hz - 20 kHz
A/D/A (24 Bit)		100 dB typical, A-weighted, 20 Hz - 20 kHz
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Old 11-09-2010, 05:36 AM   #46
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Originally Posted by Fabian View Post
I'm reading this with great interest, and the discussion about dBu/dBFS got me to look up teh specs of my interface (Lexicon Lambda). So, for this to make more sense to me... could someone please explain what the "Maximum Input Level" and output levels below really mean in relation to Reapers metering? Thanks.
The maximum input level normally corresponds to the max level before analog saturation becomes too important. It is not necessarily related to digital clipping...

Unfortunately, nothing in the lambda specs relates the digital and analog scales. For the Omega, they state that +4dBu nominal input = -15dBFS. So if Reaper tells me I record below -15dBFS RMS, I have 15 dB headroom to handle the peaks before digital clipping. In the same time, the line input can handle +22dBu, and this means that digital clipping will occur before I can hear analog saturation. But I don't think this also applies to the lambda,because the max input levels are really lower on the lambda.
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Old 11-09-2010, 07:26 AM   #47
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I'm also following and learning a lot from this thread. This first page seems to sum it up nicely for me ; especially if you scroll down to the bottom of the page where it compares an analog hardware meter vs. a DAW digital meter.
http://www.homestudioguide.com/HowTo...ingLevels.aspx

But, how does the +28db analog hardware meter relate to this PSP Vintage meter plug-in below? It only goes to plus 3 (or is that supposed to mean 30 or something)?

http://www.pspaudioware.com/plugins/...vintagemeter/#
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Old 11-09-2010, 10:55 AM   #48
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...However, i got told in that other thread that those plugins would 'automatically' trim the input to the 'proper' input range (and applying the appropriate gain after the plugin if needed).

So the question is : what is true ?...
Some plugins DO automatically sort this stuff out, and some DO NOT (including very well-thought-of ones such as waves and nebula).

So you can either try to read through the documentation and see if there is clear info on every plugin you own, or you can conduct detailed tests of every single plugin, or you can just keep your levels low and never worry about it. Totally up to you.
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Old 11-09-2010, 11:04 AM   #49
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...I have always thought that modern audio interfaces (even inexpensive ones) can record at very high quality also at -1 dBFS levels. Can you show some real measurement data that show how recording at -20 dBFS is better than recording at -1 dBFS?...
Yes, see reply below to Fabian.

Last edited by yep; 11-09-2010 at 11:27 AM.
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Old 11-09-2010, 11:11 AM   #50
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Originally Posted by Lawrence View Post
...Some cheap pres will start to clip internally at +14dBu...
To clarify, I'm not talking about preamp distortion (which might even be a good thing, if done deliberately), I'm talking about the analog stage between the preamp and the converters. And that's not just a straight wire. See next post for an illustration...
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Old 11-09-2010, 11:26 AM   #51
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Quote:
Originally Posted by Fabian View Post
...could someone please explain what the "Maximum Input Level" and output levels below really mean in relation to Reapers metering? Thanks.

Code:
Microphone Inputs	(2) Female XLR Pin 2 Hot
Input Impedance		600 ohms balanced
Phantom Power		+48 Volt
GAIN			+44 dBu
EIN			-115dB A-weighted @44dB gain (150 Ohm source impedance)
Maximum Input Level	+6.5dBu
...
Excellent post.

dBu is typically calibrated so that 0dBu equals -18dBFS on a digital meter.

That means that +6.5dBu is equivalent to about -11.5dBFS on a digital meter.

IOW, your interface is designed to operate within spec at up to -11.5dB on REAPER's input meters.(see note*)

So what does that really mean for your studio practice? Who knows. The manufacturer is telling you that the device is meant to perform within spec at levels up to around -11.5dBFS. Beyond that, it doesn't say. Maybe it will hard-clip everything. More likely, it will handle short peaks and transients just fine with a slight worsening of THD performance that becomes more egregious the more sustained the signal above threshold is. Performance losses may be somewhat frequency- or dynamics-dependent above threshold.

The point of the post that spawned this thread was not to tell you good or bad, just to suggest that you can side-step all this stuff by simply recording at lower levels. Alternately, everyone can post all their specs and argue over whether which levels should be good enough and we can all speculate about what probably happens above that 6.5dBu threshold...

*Note, AGAIN, that analog calibration is not an exact science, and there are sometimes slight differences between American and European definitions, etc. So, AGAIN, we're talking about blurry areas, not bright-line definitions. Which is exactly the problem, that all the digital literature and discussion ignores the messier realities of analog input and output stages, and "analog"-sounding plugins and processors, and also the potential undetected problems of inter-sample clipping caused by digital processing such as steep filters etc. Not to mention the generally iffy quality of digital peak detection.

EDIT: BTW, the really telling one on your spec above is the +13bDu max LINE INPUT level. That means, that even using an outboard preamp and feeding signal "straight to the converters", the interface is STILL only spec'd to handle up to -5dBFS or so, which is EXACTLY what I've been getting at. Even if you fed a signal straight into that interface with no gain, and your meters showed no clipping, the input stage itself is distorting above -5dBFS. Note also that even the output stage shows a max line level of +16dBu (-2dBFS). So, again, you have distortion even if the digital meters show zero clipping. There's no such thing as all-digital!!!

Last edited by yep; 11-09-2010 at 11:33 AM.
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Old 11-09-2010, 12:09 PM   #52
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To expand on the above, and to try and keep this thread from tuning into an autistic spec-examination...

All those numbers and measures in Fabian's spec above sound very technical and precise, but they should be regarded as essentially meaningless. Or at least, no more meaningful than a guitar amp that goes up to eleven. Here's why:

Let's say I'm a gear manufacturer making audio interfaces, and I want to publish a spec for max input level. Here are some of the ways I might go about coming up with that spec, short of outright making it up (and there's really nothing to prevent me from doing so, it's not like these are government-regulated measurements of anything):

- Look at the component list and design drawings and calculate the nominal tolerances.

- Take 10 sample units and measure the performance of each, and then set the spec level at either the average, or best-performing one (more to the point, what is the measure of "max"? Where it blows up? Where THD exceeds some arbitrary threshold? Where I think it starts to sound bad? On top of that, what am I even "testing"? A 1kHz sine wave? White noise? Swept sine waves? A Beatles CD?)

- Test the original hand-built design prototype that I made on a breadboard in my design lab (hmm... that couldn't possibly perform differently than real-world production units stamped out in a factory in Thailand made with the cheapest available components, could it?)

- Simply publish the specs of the existing hardware whose design I am attempting to copy (phew! that was easy! I was afraid I might have to learn how to use an oscilloscope!)

- Publish the design spec that the unit was SUPPOSED to meet, according to the product design criteria (IOW, marketing/management says we want a device that meets X spec. I design a device and say it should meet that spec, so that's the spec we publish. Saves a lot of trouble that way).

- Guess.

Maybe your device manufacturer wouldn't possibly do those things. Okay... so, where do their specs come from? Actual tests of your individual production unit? If so, what did they test? What does "max" mean?

My point is, don't go rushing for your manual to see if your numbers are bigger than Fabian's numbers, because those numbers are meaningless unless they are measured results of your specific unit in your studio with your real-world program material. It's like comparing how high the numbers on guitar amplifier volume knobs go.
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Old 11-09-2010, 01:39 PM   #53
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Thanks yep, great info.

Maybe I'm drifting off-topic here, if so I apologize for that, but just so I understand this...

There is a peak LED on the device that according to the manual lights up "when the input signal is within 5dB of analog clipping". I understand that we cannot be sure what they mean exactly by "analog clipping", but lets assume that "analog clipping" occurs at "max line input level", which is specified to be +13dBu. Then the peak LED would light up around +8dBu, right?

So I hooked up a HW synth to the device line in and adjusted the input level so I just barely got a weak flickering on the peak LED. This input signal Reaper measured as -5.7 dBFS peak.

Is this a valid test? Is it meaningful? What does it tell us? Does it mean that 0dBu is around -14dBFS?
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Old 11-09-2010, 02:51 PM   #54
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Originally Posted by Fabian View Post
...Is this a valid test?
Well, it does prove that there is an LED, and that it lights up.

Quote:
Is it meaningful? What does it tell us?
It tells us that the manufacturer has provided, as a courtesy, a little red bulb that lights up to warn us that we may be pushing the design limits of the system. It's like "redline" on a car tachometer-- it's not really a scientific measure of anything. The car doesn't blow up if you go over the redline, nor does instantly and constantly accelerating to one RPM below the redline necessarily prolong its life beyond ordinary sensible driving that might occasionally include "redlining" the engine.

In short, if we have a sort of autistic need for everything to indicate a precise boundary and meaning, then no. The light doesn't have any meaning. If, on the other hand, we are better-equipped to make use of vague and "messy" information, then it could be very useful visual cue to help alert you when the singer is succumbing to "volume creep".

Quote:
Does it mean that 0dBu is around -14dBFS?
I don't think that has been conclusively proven by your test. There are a lot of ways that a "clip" LED might work, and they don't all relate precisely to individual sample values-- suppose the clip LED takes 1ms to respond, which is a perfectly reasonable time to detect audible analog clipping: that means that, at an 88.1k sample rate, 88 samples will have gone through the converter before the clip LED detects "analog" clipping.

So if you have an OCD need to know exactly what everything "means", then you're going to need much better test equipment. Otherwise, you could just record lower levels and not worry so much about what it "means."

Last edited by yep; 11-09-2010 at 02:57 PM.
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Old 11-09-2010, 02:58 PM   #55
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Yep, your patience and ability to consistently bring issues back to the same conclusion is awesome! Seriously!

I think I began this thread looking for a black/white answer. The answer is gray and the reasoning is sound (pun-intended).

Thanks!
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Old 11-09-2010, 04:10 PM   #56
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Thanks you! I think this has also answered the question I asked earlier today in this thread about input levels on my soundcard and the output on my pre.

I think the answer is that I shouldn't worry about it - just track at an appropriate level and record. If so, that's what I have been doing, but just wanted to be sure.
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Old 11-09-2010, 08:36 PM   #57
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Yep is indeed amazing with his patience and his willingness to share knowledge - thanks! I'm still confuse about my post shown below...any answers on this?

Thanks again!!

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Originally Posted by Serenitynow View Post
I'm also following and learning a lot from this thread. This first page seems to sum it up nicely for me ; especially if you scroll down to the bottom of the page where it compares an analog hardware meter vs. a DAW digital meter.

http://www.homestudioguide.com/HowTo...ingLevels.aspx

But, how does the +28db analog hardware meter relate to this PSP Vintage meter plug-in below? It only goes to plus 3 (or is that supposed to mean 30 or something)?

http://www.pspaudioware.com/plugins/...vintagemeter/#
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Old 11-09-2010, 09:18 PM   #58
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Yep is indeed amazing with his patience and his willingness to share knowledge - thanks! I'm still confuse about my post shown below...any answers on this?

Thanks again!!
Um, you are linking to a document on your C drive, which nobody else can access. The second question about PSP vintage meter is completely unclear, but I'm pretty sure you can adjust the meter's range, if that's what you're asking.

It might help if you can clarify the question.
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Old 11-09-2010, 09:51 PM   #59
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On the subject of PSP Vintage Meter, mind if I bump Post 39 in this thread? It's actually related to the Naiant meter.
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Old 11-09-2010, 10:09 PM   #60
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On the subject of PSP Vintage Meter, mind if I bump Post 39 in this thread? It's actually related to the Naiant meter.
Once again, I can't really tell what the question is.

Peak meters display the level of each sample. VU or RMS meters display a sort of weighted rolling "average".*

If the "average" level is much lower than the instantaneous peak levels (say with drums, or hand claps), then the VU meter will read much lower than the peak meter. If the "average" level is almost as loud as the maximum peak level (say, with string pads of heavily-saturated electric guitar), then they might read about the same.

Is that what you're asking?

*note that "average" is a very imprecise and fuzzy term when it comes to audio metering. Classical "VU" meters are traditionally set to try and approximate the apparent "loudness" of typical human speech. Human speech is often different from, say, electric bass or string sections, so usability can vary. Once again, it is/was up to the equipment designers and manufacturers to tailor the equipment to what they thought people would find most useful for the intended application. Also once again, where and how they set the meter's "target" could, in itself, have a significant difference on the "sound" of the piece of kit, by encouraging different target levels of operation.
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Old 11-10-2010, 06:12 AM   #61
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Yep should be given a LIFETIME Reaper license for his considerable efforts here. No bull-pucky.

For prosumer digital with relatively poor metering, -10 is the new 0. If you browse through some digital console manuals you'll probably find a reference to that as a general "maximum peak in all cases" target.

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Old 11-10-2010, 06:13 AM   #62
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Sorry, it was related to metering that shows 0dBu when we currently see -18dBFS in REAPER. Gizzmo suggested using the Naiant meter and its -18 calibration setting.

The question was, "Can I use this meter with this setting to properly assess my recorded and internal levels?" Well...

It's been said that to be on the safe side, and to avoid detailed analyses of all pieces of input gear specs and plug in programming, one should set averages around -20 or peaks around -10. I was blindly applying that here, thinking that 0 on a VU meter means the same thing in terms of loudness and peak for a kick drum as it does a droning violin note.

I feel like I'm searching for a rule to be applied to all tracks regardless of content. Is that realistic? Getting them to 0 on a VU meter isn't the answer because it isn't telling the whole headroom story. Sure, it's built to protect headroom, but it doesn't exactly show it. And that brings us right back to peak metering and my OP. If all levels obey a peak (and it seems like that peak could be anywhere between -20 and -10 dBFS) that should lead one to a cleaner mixbus, clinically speaking.

So the $10,000 question is, "If I track all my inputs to peak around -20dBFS on REAPER's meters, or trim (pre-FX volume) any tracks NOT recorded that way to -20dBFS on REAPER's meters, will I end up with a better sounding mix than if I had run all those same levels to just below 0dBFS."

It's a big question that belies the depth of understanding to answer it, but from this thread I'm hearing YES as the answer.
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Last edited by kelp; 11-10-2010 at 07:00 AM. Reason: Changed "0dBVU" to "0dBu".
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Old 11-10-2010, 06:39 AM   #63
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"Joe Meek" says it (again) ...

Quote:
Many people assume that digital recorders and DAWs should be driven up to maximum - so that all 24 bits of sample data are used - in order to get the maximum dynamic range and signal-to-noise ratio from the digital system. On a digital recorder or DAW, the top of the bargraph meter represents "FSD" or "Full Scale Digital" - in other words, the onset of digital clipping! These meters are typically referenced such that -18dBFS = 0dBu, so in analogue terms, FSD is actually equivalent to +18dBu, which is a very high level and which in most analogue gear is within a few dB of clipping.

Signal-to-noise ratio and distortion in most analogue gear are optimum at around +4dBu, which is why most meters are calibrated such that "0" VU corresponds to +4dBu.

Now whereas digital users like to see their bargraphs bouncing up to within a few dB of the top, most analogue users work at more restrained levels and analogue meters generally cover a smaller range. In other words, while a DAW meter may cover the range from -40dBu to +18dBu, analogue meters may only cover the range from -24dBu to +12dBu. This problem is compounded when (as with the Joemeek twinQ and oneQ ) mechanical VU meters are fitted that only cover the range from about -16dBu to +8dBu.
Other than that, get good monitors and use your ears.
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Old 11-10-2010, 10:05 AM   #64
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Hi everyone:

I did an experiment concerning the topic.

I had a mix with some tracks touching almost 0.0 in reaper. All of them had their respective FXs still online.

What I did was Item Properties>Normalize> -16.0db (this was done to all the Items)

Some compressors had to be adjusted for obvious reasons but I didn't notice any other change apart of the lower volume.

Did I do it in the correct way or I just did something without any sense at all?

Thanks and sorry for the newbieness

Last edited by Behind; 11-10-2010 at 10:21 AM.
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Old 11-10-2010, 04:05 PM   #65
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...I feel like I'm searching for a rule to be applied to all tracks regardless of content. Is that realistic? Getting them to 0 on a VU meter isn't the answer because...
Shooting for max peak levels of -10dBFS (digital) is probably safe for most things.

Of course, everything always comes back to "if it sounds good, it is good" and vice-versa. VU-style metering can make it a little easier to focus on a "target" range, since peaks can be wild and jumpy, and since an occasional instantaneous over on a loud drum hit or whatever probably isn't going to wreck your track. In that case, -20 or -24 AVERAGE will usually produce peaks of around -10dBFS, so it's kinda six/half-a-dozen.
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Old 11-10-2010, 04:09 PM   #66
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Originally Posted by Behind View Post
Hi everyone:

I did an experiment concerning the topic.

I had a mix with some tracks touching almost 0.0 in reaper. All of them had their respective FXs still online.

What I did was Item Properties>Normalize> -16.0db (this was done to all the Items)

Some compressors had to be adjusted for obvious reasons but I didn't notice any other change apart of the lower volume.

Did I do it in the correct way or I just did something without any sense at all?

Thanks and sorry for the newbieness
Yeah, that's fine. You'll probably get better results in the future by TRACKING at a lower level initially, since some of what we're talking about is distortion that can get embedded in audio that is recorded too close to 0dBFS in the first place.

Also, -16 peak might be a little bit lower than you really need to go, and might start to make it hard for some compressors to find enough signal to grab onto.

And of course, all of this is a suggestion, not a rule. The suggestion is basically to keep your tracks at the same approximate level that an analog recordist would: somewhere around -20dB average, -10 peak.
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Old 11-10-2010, 05:30 PM   #67
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Doh, I'm so sorry! I'm really not a moron! I just started using a new add-on "Scrapbook" feature of Firefox. It saves a copy of the page on your hard drive for keeps (instead of later going to a 404 page). So, when I was looking at it I just copied what I thought was the URL into the post!!

Anyway, here's how it should have read w/ the correct link;

I'm also following and learning a lot from this thread. This first page seems to sum it up nicely for me ; especially if you scroll down to the bottom of the page where it compares an analog hardware meter vs. a DAW digital meter.

http://www.homestudioguide.com/HowTo...ingLevels.aspx

But, how does the +28db analog hardware meter relate to this PSP Vintage meter plug-in below? It only goes to plus 3 (or is that supposed to mean 30 or something)?

http://www.pspaudioware.com/plugins/...vintagemeter/#


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Originally Posted by yep View Post
Um, you are linking to a document on your C drive, which nobody else can access. The second question about PSP vintage meter is completely unclear, but I'm pretty sure you can adjust the meter's range, if that's what you're asking.

It might help if you can clarify the question.
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Old 11-10-2010, 06:27 PM   #68
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...But, how does the +28db analog hardware meter relate to this PSP Vintage meter plug-in below?...
What are you actually trying to accomplish?

You're right that the first link explains exactly what we're talking about. I don't get what the PSP meter has to do with it. Have you read the manual for the PSP meter? What are you trying to do with it? You can set the "zero" level on the PSP meter to whatever threshold you want, if that's what you're asking...
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Old 11-10-2010, 09:19 PM   #69
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I'm just saying the first site demonstrates to me what you guys are talking about - I get all that - it's been very helpful to me.

But, all I'm wondering is why do the normal old school VU meters like in the 2nd link (the PsP vintagemeters, as well as the kind we've all seen on cassette decks etc. all these years)only go to +3 (while the digital analog hardware meters in the first link go to +28)? How do they fit into the equation and compare to each other? Is it a totally different scale? ...just curious.

Thanks for anyone's help!


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What are you actually trying to accomplish?

You're right that the first link explains exactly what we're talking about. I don't get what the PSP meter has to do with it. Have you read the manual for the PSP meter? What are you trying to do with it? You can set the "zero" level on the PSP meter to whatever threshold you want, if that's what you're asking...
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Old 11-10-2010, 09:58 PM   #70
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Quote:
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...But, all I'm wondering is why do the normal old school VU meters like in the 2nd link (the PsP vintagemeters, as well as the kind we've all seen on cassette decks etc. all these years)only go to +3 (while the digital analog hardware meters in the first link go to +28)? How do they fit into the equation and compare to each other? Is it a totally different scale? ...just curious...
Some guitar amps go to eleven. people make meters differently. that's really all there is to it.

With a VU meter, it really doesn't matter how headroom you show, since the user is reading average level.
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Old 11-11-2010, 10:45 AM   #71
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I just wanted to show this input meter from Studio One. The "good" range for tracking is nicely setup there, the last "fat" spacing, between -24 and -12. Keeping your input signals averaging in the middle of that range would be a no brainer so I like the way they marked the scale. It could still use an RMS meter there also though.

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Old 11-11-2010, 12:28 PM   #72
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Does this guideline to record at -24...-12 dBFS level apply only to 24-bit recording?
What is the recommended level in 16-bit recording?

Another thing that still puzzles me is the input level calibration. Based on the Homestudioguide article linked in post #67, it seems that after you have calibrated your mic/instrument inputs in the analog mixer you are safe to use the analog headroom (don't adjust input gain but use faders). And when you use all that headroom and record the mixer's otput to DAW you will get -2 dBFS peak level in your DAW. According to the Homestudioguide article that is perfectly fine.
But reading this thread I understand that there can be potential distortion problems in the DAW input (MOTU 828mkii in the article) if recorded at -2 dBFS level.
So, the Homestudioguide article did not tell the whole story and following the advice in that article could lead to bad recording quality in the DAW.
Did I understand this one right?


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Old 11-11-2010, 01:25 PM   #73
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Does this guideline to record at -24...-12 dBFS level apply only to 24-bit recording?
What is the recommended level in 16-bit recording?
The recommended level is to switch to 24-bit recording.

More seriously, with 16 bit you have a much smaller needle to thread. The same principles apply, but you're closer to the noise floor. With 16-bit it's time to start considering applying analog compression, filtering, etc, before converting to digital, in order to make the best use of the "space" you have. More sensibly, though, it's time to switch to 24-bit recording.

Quote:
Another thing that still puzzles me is the input level calibration. Based on the Homestudioguide article linked in post #67, it seems that after you have calibrated your mic/instrument inputs in the analog mixer you are safe to use the analog headroom (don't adjust input gain but use faders). And when you use all that headroom and record the mixer's otput to DAW you will get -2 dBFS peak level in your DAW. According to the Homestudioguide article that is perfectly fine.
But reading this thread I understand that there can be potential distortion problems in the DAW input (MOTU 828mkii in the article) if recorded at -2 dBFS level.
So, the Homestudioguide article did not tell the whole story and following the advice in that article could lead to bad recording quality in the DAW.
Did I understand this one right?
Once again, the more obsessive-compulsive and autistic we start to get about rules and recipes and "correct" levels, the more all this stuff disintegrates into "if it sounds good, it is good" (and vice versa). Which is absolutely correct, but of mitigated usefulness when you're trying to record 12 tracks of drums in the same room the drummer is playing in, while also trying to monitor for subtle distortion on the bottom snare mic. Or when you're trying to detect a little bit of edginess on an acoustic guitar track while simultaneously playing the guitar and signing to a headphone mix of backing tracks with your voice resonating in your skull and the guitar vibrating your chest cavity and so on.

The point of the suggestion that prompted this thread is not to tell you at what input level you will start to get analog clipping or "inter-sample" digital overs... I have no idea, because I don't know what you're recording or what kind of equipment it's going through or anything like that.

With respect to the specific question about whether the home recording guy might still be experiencing clipping, there's no way to know from here, but a couple of points are relevant:

1. Very brief, instantaneously clipping of peak transients is basically inaudible and irrelevant. One or two samples one track in a 48-track project that stray into overs are not going to ruin a take.

2. Analog systems frequently handle such instantaneous overs quite elegantly, without distorting. Think of analog as like rubber, digital as like glass-- it IS possible to break or permanently distort analog by trying to pass something too-large through it, but you can get away with a lot more if you don't push your luck too much. So even if he occasionally gets a transient past the "optimum performance" range of the system, it's not necessarily going to ruin the take (at least not within the analog realm-- converting to digital is another story).

3. Very few kinds of source tracks in typical pop/rock production are likely to have peaks that are much more than 14 or so decibels above the steady-state "average" (we're speaking in pretty fuzzy terms here, so take with a grain of salt). IOW, not very many people who have an average daily checking account balance of, say, $10,000 are likely to have to deal with recurring periods where they unexpectedly have $100,000 in their account for a day or two. So if he's recording at a max "VU meter" level of zero dB, where zero on the VU corresponds to -20dBFS or whatever, the problem of what happens to +18dB peaks is likely to be an infrequent, if not entirely theoretical one.

This suggestion is not a "rule for good sound".

If what you care about is your own recordings, the best thing you can do is to suss out your own gain-staging with a day or two of careful experimentation. Put on a pair of good headphones, Record different instruments as hot as you can and then at softer levels and see whether and where the "edginess" or "brittleness" starts to recede, and then record at levels slightly lower than that. And then do the same thing with all your plugins and processors, and every other piece of equipment.

Or, make life easier and just record a little bit lower than common sense tells you is "safe", and chances are you won't have many problems, and that, if you do, they will be infrequent and not too severe.
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Old 11-11-2010, 01:34 PM   #74
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Yep, thanks once again. I've learned a ton on this thread!
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Old 11-11-2010, 01:55 PM   #75
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Here's that same meter as a plug, but with RMS metering. I'd be comfortable recording that source signal right where it is. It peaks a little less than -10, no biggie, but the average level is right where it should be. The synth attack has a bit of a transient there.

Why we don't have RMS on the actual input meters, no idea. If I were tracking this through an analog console it would land just about there in the daw. Consider that white line the "VU needle". It's hovering right around a common semi-pro converter calibration.


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Old 11-12-2010, 03:54 PM   #76
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A pretty interesting read btw is found here:
http://en.wikipedia.org/wiki/DBFS

If you look at the differing reference levels, sorry: "alignment levels", in different countries or even within the same country (Germany ), I'm not surprised anymore that I couldn't give you a better description in my first reply
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Old 11-12-2010, 05:06 PM   #77
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So tracking levels of -18 on average are ok...and my song sounds quite nice...but then everyone talks about how advantageous it is to leave yourself so much headroom...I think because you then have room to boost your song if you want to....to raise it's volume. But then when I try to give it a 3-6db boost to lift rms from say -18 to -12....the song starts to sound terrible and hot and ugly. So I must be doing something wrong...if I've left all that sweet sweet headroom...but when I start to put my foot in the door of that headroom it starts to sound ugly almost right away. The acoustic guitar stops sounding like a breathing acoustic guitar and starts sounding like some mechanical thing that breathes hot air and fire. I'm recording at 24bit

What would you all say is my problem?
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Old 11-12-2010, 05:35 PM   #78
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----- But then when I try to give it a 3-6db boost to lift rms from say -18 to -12....the song starts to sound terrible and hot and ugly.-----
And the master is NOT clipping???
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Old 11-12-2010, 07:16 PM   #79
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So tracking levels of -18 on average are ok...and my song sounds quite nice...but then everyone talks about how advantageous it is to leave yourself so much headroom...I think because you then have room to boost your song if you want to....to raise it's volume. But then when I try to give it a 3-6db boost to lift rms from say -18 to -12....the song starts to sound terrible and hot and ugly. So I must be doing something wrong...if I've left all that sweet sweet headroom...but when I start to put my foot in the door of that headroom it starts to sound ugly almost right away. The acoustic guitar stops sounding like a breathing acoustic guitar and starts sounding like some mechanical thing that breathes hot air and fire. I'm recording at 24bit

What would you all say is my problem?
I think you've hit upon exactly what this thread has been getting at-- that it is very difficult to know exactly where something might be going wrong when you are relying solely on digital peak metering.

In a high-end analog world, if you noticed that kind of problem, you could just glance at all the meters and chances are you'd see one of them "in the red", indicating that it was operating outside the range where the designer thought it sounded best (maybe on the console, maybe on a compressor, maybe on an outboard effects box, whatever-- the point is that they all had "target" meters)

The simplest advice in your case is just to keep your levels low and your headroom clean. If you feel the need to maximize levels for final distribution, shoot it to a mastering engineer who has the gear and setup to do that properly.

The longer answer is suss out every single thing that is happening in every track, signal chain, aux bus, and even your digital-to-analog output, speaker amp, speakers, etc. You'll find it eventually.
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Old 11-14-2010, 04:31 AM   #80
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I don't know if I am relying on digital 'peak' metering....although maybe I am.

I thought I was concentrating on the rms...the average level.

Maybe I am looking at the average level to the detriment of awareness of peak level when all summed.

I will have to look into the things you outlined more carefully
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