Kiwiburger

Absolutely true. Many plugins are designed to model analog, and get crunchier the hotter you go, and may totally clip at 0dBFS. That is purely by design - it's not a limitation of the internal digital stream.

However - I personally believe when Mixing (as opposed to Pre-Mastering) you shouldn't have any plugins on the master bus. Otherwise you have limited choices when mastering. And because you should be mixing with sufficient headroom for mastering, especially if you intend to output via D/A converters. So you don't even need a limiter, because your master bus output is going to be well under 0dBFS.

I'm definately not recommending that anyone " slam the {censored} out of your {censored}". That is Where's willful ignorance on display.

The results of abusing analog converters are immediately obvious - yes. Nothing to do with the "masters". That is analog thinking - it doesn't apply to digital.

That's a lazy mans method, and guaranteed to ensure under-recording, with resulting noise and loss of bit depth.

The only area of contention in my mind is concerning the misconception that you can't use the master fader to lower your mix to the appropriate output level. I see no justified reason to believe that - my ears can't hear any reason to be scared of doing this.

This thinking affects what Where calls "keeping proper fader levels ". I believe he is suggesting that to alter the output level of your mix, you must alter the level of all individual channels. That seems an incredibly restricted way of working.

Personally - I think analog VU meter thinking doesn't work that well with digital. Its fine for outside the box, but inside the box it's peak dBFS that matters most (which is why DAW makers give us peak meters). Possibly the exceptions would be certain plugins, but they often have their own meters for the specific task.

My experience of recent years is mainly with Cubase SX. I haven't used Protools for a while, and can't comment on how it sounds - but Digidesign literature suggests the same principles apply.

Seems to me that people with major issues with digital summing quality tend to be Protools users for some reason - there might well be something to it that I don't know about.

Yes - if you track correctly, it would be hard to screw up.

I wasn't ever suggesting tracking too hot. I was simply trying to explain the reason why you don't want to track to hot.

It's not because you can clip a digital summing bus. It's for reasons in the analog realm.

Protools HD uses 48 bits fixed (newer native DAWs use 64 bits, either fixed or floating).

But from the Protools website I got this fact:

48 bits fixed allows enough headroom to sum 128 tracks of phase aligned sinewaves all peaking at +12dbFS!

For practical purposes, you just aren't going to clip a summing buss - not even in Protools HD, which is arguably the worst.

You're right - I won't labour the point. It's fairly academic anyway - but I like to understand 'why' I have do something. I'm never satisfied I know what i'm doing until I know 'why' i'm doing.

Where is 100% correct about aiming for the sweet spot of a converter, which is typically around 18dB below full scale. This is absolutely spot on advice, compared to the general recommendation to track as hot as you can.

It was just his recurring statement that you do this to avoid clipping the digital summing bus that irked me. In my view you do this to avoid unwanted coloration in the analog circuit of the converter. And maybe unintentional coloration in the preamp and any other analog circuitry in the chain.

Analog generally gets dirtier the hotter it gets.

Pure digital audio (numbers, not voltage) get cleaner the hotter it gets - right up until the point it suddenly clips. But floating point math never clips.

The same principles apply when coming out of the box - but in practice, because of the loudness wars, this whole thing gets thrown out the window.

Sorry to be so pedantic - no hard feelings.

OK - I give up. You are willfully ignorant about what I am saying - and I can't say it any clearer so I give up. You are intent on misunderstanding and misquoting me, so I just give up.

There is absolutely no way that I am "insisting that you can go beyond 0dBFS" within the converter. I made this point very clear by saying "i'm not talking about digital clipping, running out of numbers."

I have agreed all along that you should not record too hot in the analog realm - at least, not unless you like the distorted sound. And this distortion is waaaay before 0dBFS clipping.

Why are you so intent on misunderstanding, and presenting bogus information - such as saying that you can clip a digital summing bus by exceeding 0dBFS inside the computer? This is false, misleading information.

Nobody is arguing about tracking too hot. I was taking issue with your bogus information concerning digital summing. Summing involves "summing" many digital tracks together. So even if you record 24 tracks at nominal 0dBVU - when you summ them together, they can exceed 0dBFS internally.

Using your advice - which I disagree with - you would say that you should lower the track faders, so the don't sum to anything higher than 0dBFS. You are saying it's possible to overload the digital summing bus - and I am disagreeing because this is not true.

Have you ever done the simple sine wave test I described above? I'm think you haven't, because it would become clear that you have egg on your face.

Anyway - I give up. It will still annoy me when I see you telling newbies this misleading information, but I guess they can work it out for themselves.

No - a digital meter is obviously post-a/d converter. It's just your analog stuff that needs to be set up around nominal 0dBVU. Up to, and including, your A/D converter.

Some of this advice is more academic than useful. As long as you don't clip your a/d converters, or seriously under-record, you can't go too wrong.

If you understand that a/d converters are an analog device - the more you push them, the more distortion you get (within it's analog electronic circuit - not talking about digital clipping, running our of numbers.

So the purpose of lowering your gain structure and not hitting the a/d so hard is simply to avoid unwanted analog distortion.

Consider that many people find that DAW recordings are waay too clean, and apply saturation plugins and all sorts of tricks to dirty up the sound and make it more analog ...

So forget your meters and trust your ears. Maybe you will like the analog distortion of your converters (not talking about clipping at all).

Also, your preamps and compressor (if used) have similar analog distortion characteristics. Don't blame your converters for distortion that might be occuring before it gets to them.

Ultimately - there is a wide range of taste in sounds. Some people buy certain gear to get clinical, sterile, accurate sounds. Others buy certain gear to get warm, distorted, character sounds.

I think it's very useful to know exactly what each piece of gear you have is capable of. Abuse it, make it distort, back off, see how clean it gets. Then you will know it's "sweet spot", or at least know it's weaknesses.

Understanding each link in the chain let's you make decisions that help you get the sound you want.

Frankly, I think VU meters are fairly useless. Sometimes to get the sound I like, I'm pegging a VU meter and I feel sorry for it. Othertimes, to get the clean sounds I want, the VU meter is hardly moving. Fairly irrelevant in my opinion.

It's a new digital world - peak meters are more important. That probably offends the old school, but let it be.

slow server - double post.

No - a digital meter is obviously post-a/d converter. It's just your analog stuff that needs to be set up around nominal 0dBVU. Up to, and including, your A/D converter.

Some of this advice is more academic than useful. As long as you don't clip your a/d converters, or seriously under-record, you can't go too wrong.

If you understand that a/d converters are an analog device - the more you push them, the more distortion you get (within it's analog electronic circuit - not talking about digital clipping, running our of numbers.

So the purpose of lowering your gain structure and not hitting the a/d so hard is simply to avoid unwanted analog distortion.

Consider that many people find that DAW recordings are waay too clean, and apply saturation plugins and all sorts of tricks to dirty up the sound and make it more analog ...

So forget your meters and trust your ears. Maybe you will like the analog distortion of your converters (not talking about clipping at all).

Also, your preamps and compressor (if used) have similar analog distortion characteristics. Don't blame your converters for distortion that might be occuring before it gets to them.

Ultimately - there is a wide range of taste in sounds. Some people buy certain gear to get clinical, sterile, accurate sounds. Others buy certain gear to get warm, distorted, character sounds.

I think it's very useful to know exactly what each piece of gear you have is capable of. Abuse it, make it distort, back off, see how clean it gets. Then you will know it's "sweet spot", or at least know it's weaknesses.

Understanding each link in the chain let's you make decisions that help you get the sound you want.

Frankly, I think VU meters are fairly useless. Sometimes to get the sound I like, I'm pegging a VU meter and I feel sorry for it. Othertimes, to get the clean sounds I want, the VU meter is hardly moving. Fairly irrelevant in my opinion.

It's a new digital world - peak meters are more important. That probably offends the old school, but let it be.

I absolutely agree 1000%. Because whenever we are talking about converters, we are talking about the analog domain.

My comments about the digital domain were simply because I take issue with the idea brought up that you can "overload a digital summing buss". You can't.

Sure - if you go waay over 0dBFS you get 'rounding errors', as the floating point moves. But that's no worse than throwing away bit depth resolution - which is a similar loss.

Neither will be audible for practical purposes.

I just think it's useful to know exactly where the distortion is occuring. Filling people's heads with the nonsense idea that the distortion is occuring in the digital summing bus doesn't help the understanding of the problem.

This is so frustrating ...

What part of "you cannot clip a 32 bit floating point file" don't you understand?

Anyone can prove this - not that there is any need to prove this. It's a simple digital audio fact.

Create a sinewave that peaks at 0dBFS. Apply a 12dB boost.

YES - you will hear square wave clipping, as you are overdriving the FIXED point 24 bit converters.

Export that 'clipped' wave to a 32 bit floating point wave file.

View it in a wave editor - and it will probably look like a square wave.

Import this 'clipped' wavefile into your DAW.

Lower the master fader by 12dB - the clipping goes away. You are left with a perfect sinewave. This is because the wave file was never clipped - you only thought it was because of what your converter was doing.

I'm not defending overly hot, squashed mixes. I'm just saying you can't clip a floating point file. And therefore there is no reason to waste bit depth, or be worried about lowering the master fader as required.

Like I've said numerous times - once your audio has to go through a converter, you then have to start apply analog reasoning again - and leave sufficient headroom.

You still don't want to understand this, do you?

Within the digital domain, you can safely exceed 0dB Full Scale, with zero damage to your audio wave.

I'm talking about any semi-decent DAW, like Cubase SX3. Read the makers literature about 32 bit floating point files.

The damage only occurs when you reach an analog converter and clip that. Or, if you use a plugin or other software application that uses fixed point data, or is designed to clip to emulate analog.

This is so easy to prove, that i'm surprised at experienced engineers who try to state otherwise.

The headroom isn't infinite, but it's so huge as to never present a practical problem.

I'm not saying abuse it. I'm simply saying don't be afraid to use all the available resolution of your tracks, and there is no reason to be alarmed if your master bus goes above 0dbFS prefader.

There is no need to panic and upset your mix by trying to lower all your track faders. Simply lower the master fader so your external analog converter does not clip. Or so your rendered file doesn't clip.

If you are rendering to 32 bit float (which I recommend) you can leave it 'clipping' (because it's not really clipping at all).

The reason I strongly believe this is,

A - because it's true

B - because i've proved it

C - because with digital audio, you never want to throw away bit depth if you don't have to

Everthing changes once you decide to hit an analog converter. That is when you want to optimise the sound.

(Or not, if your a friggin Mastering Engineer, you will probably want to clip your D/A converter on purpose).

How do you think the top mastering engineers manage to clip their converters? You have to exceed 0dBFS to achieve that. You don't want to be apply 12dB of useless empty zero digital makeup gain to achieve that. You want the full resolution available.

And before i'm flamed for promoting squashed dynamics - this is not the issue at all. I'm talking peak levels (dbFS) - not average levels. So this could be highly dynamic material i'm talking about.

I'm just wanting to expose what I see as faulty rhetoric about digital. Read the makers manual - and run some tests to prove it to yourself.

I realise what I said could be misunderstood ...

FWIW - i'm not recommending tracking too hot ... if you understand what I mean about the A/D converters being analog, you will understand that I am recommending tracking at moderate levels (nominal 0dBU for the converter, typically around -18dBFS).

If you also understand that the D/A converter is analog, you won't want to smash your mix into the top 6dB range either ... even though most mastering houses seem to do this on purpose these days.

So i'm really saying the same this as Where and Massive. But where I slightly disagree is the continuation of analog thinking once inside the box. To the point that many people think they can't use the full range of their track faders - which is daft.

There is no reason to be afraid of using the master fader as required.

There is no reason to worry about digital channels exeeding 0dB, not until you reach the D/A converter.

Although the exception would be certain plugins that are designed to model analog. You will hear the clipping if that is the case.

My basic principle with digital stuff is to only attenuate, and try to avoid boosting, at all costs. That's because when bit depth is lost, it's lost for good.

With analog, you can attenuate then boost, attenuate and boost, numerous times, and get away with maybe some extra noise.

With digital, if you attenuate - you throw away apparent bit depth. If you boost, you don't regain the lost bit depth. So if you attenuate and boost too many times, you could end up with very grainy waves.

That's why I think that it's healthy to have strong levels on each track, which will be attenuated later. And for the sake of processing, I think the attenuation is best done after the processing. I see (and hear) no problem with lowering the master fader.

It's just the expression "overloading the summing amps" that bugs me when talking about internal digital audio paths. I don't believe it's possible.

(Donning flame proof suit ...)

OK - here's my take on it:

Digital converters (A/D and D/A) are primarily Analog devices. Think about that for a minute. Analog.

All Analog devices are designed with good headroom above 0dBU. That is because VU meters display a slow average level, and there are plenty of transients in audio that temporarily exceed 0dB but don't move a VU meter.

So in the analog realm, right up to, and including, the a/d converter, you must maintain headroom. The same applies coming out of the digital realm - although this is flaunted for the purpose of loudness (at the expense of distortion) even by mastering engineers these days.

I disagree with some of the terminology in some comments above:

Most daws don't have summing amps that can be overloaded. It's just mathematical calculations - so this statement is a little like saying you can overload a pocket calculator. Sure - you can exceed the significant number of bits, and end up with some small rounding errors, but this is not a practical concern.

Any damage/distortion that you might hear from running digital too hot is usually ocurring in the analog realm of your converters.

It is completely possible to make audio files that exceed 0dB. These will sound extremely clipped because you are hearing them through fixed bit converters. You can save the file, reimport it, drop the master fader, and the distortion goes away again. This is because the file was not actually clipped at all - you just thought it was, because of what your d/a converter was doing.

The statement that "all bits are used at 0dBU" is true and misleading.

If you record total silence at 24 bits - all 24 bits are used.

If you record total clipping at 24 bits - all 24 bits are used.

If you record at perfect 0dBU at 24 bits - all 24 bits are used.

That's not very helpful information.

The important factor is called the "Apparent Bit Depth". For example - you can convert a 16 bit file to 24 bits. You haven't improved the sound quality - you've just bulked it up with extra bits that do nothing. Some software can analyse a file and tell you the apparent bit depth. The more bits that are "used", the better the sound quality.

This is academic hair spitting. It should be obvious to any sensible person that the deeper the apparant bit depth, the better the audio quality (I like to just say "resolution" for short, but that upsets the academics).

I think it's a case of old school vs new school thinking. Both are correct, in there correct realms.

If you apply analog reasoning to digital, you screw up.

If you apply digital reasoning to analog, you screw up.

If you fully understand that digital converters are analog devices, you shouldn't screw up.

And don't forget that some DSP algorithmns are designed to model analog, and therefore you may need to apply analog reasoning in those cases. (E.g. saturatation plugins).

Flame away ...

What's wrong with motorised faders on a control surface? The Behringer one is cheap as chips.

I doubt it. Who wants to maul their keyboard with their sticky paws. Or a pointy stick. Ugh. Unhygenic, not very accurate, expensive and highly breakable. I would think.

Apple will probably make some expensive disposable toys with it.

The big thing is - being an Apple patent, it will require Apple software.

No for me.

Ah yes - that essential studio item. My lava lamp has just faded over time

From what I understand, it's wax and water. The wax is dyed with oil-soluble dye, and the water is dyed with water soluble dye. Otherwise they would just mix and end up a uniform color.

So basically, it's molten wax embeded in your carpet ... I think a steam cleaner that has some real heat, and water and detergents should be able to get this out.

Ain't that the truth.

Sorry, I don't mean to bag newegg - I don't know them, wrong country and all that.

It seems to me that this guy wants free computer building.

It's a bit like getting the cheapest prices for all the materials for making a house. And then getting quotes from builders to build a house, and accusing them of being greedy for charging for their labour.

Not really different.

And not helping him get into computer music.

Edit: ooops - forgot this is Phil's column and I should be more polite and less . Sorry everyone - forget this.

As your were.

It's not just CPU speed. Things change all the time. Firewire 400 will be replaced by Firewire 800, etc, etc, etc,

My point is that you can wait, and wait and study and shop around and wait and study and shop around ... it's a moving target, and you will never get the 'best' machine for more than 5 minutes.

You just have to buy the best you can right now, otherwise you won't buy anything and still be asking the same questions in 6 months time.

But I think the original issue is that this guy wanted bargain basement prices, with lots of custom assembly and support thrown in. Ain't going to happen.

It's one thing to accuse corporations of being greedy. It's another thing to want to screw everyone for the cheapest possible bargain, and then expect a high level of service. It's two sides of the same greedy coin.

Shangrila - please learn how to use paragraphs. What you said is bloody hard on the eyes, and you will be alienating a lot of people who might be able to help you.

See - it's not hard to do.

Anyway - you seem to be picking on new egg because they are cheap. Buying from the cheapest vendor is usually a BIG mistake. They are cheap for a reason - they will be buying the cheapest deals they can, and offering the least service they possibly can. You run a big risk that they are equally cheap with their warranty and after-sales support.

Ain't no way they are going to employ more skilled people to build custom PC's - and NOT PUT UP THE PRICES like everyone else.

Get over it. Get another job if you have to - otherwise you will be asking the same questions in 18 months time.

I didn't suggest your weren't a musician. But since you bring it up, many great musicians never touched a computer in their lifes.

The point is - if you really want to be a computer musician, don't wait 18 months. Jump in - buy something, anything.

Whatever you buy today would be a gleam in the eyes of somebody 18 months ago, and a boat anchor in 18 months time.

It's not worth putting excess energy into disposable PC's. You said you wanted to be musician - or a computer tech?

I have a bunch of IT certificates, which is helpful, but if it's music you are interested in, make some music.

That's crazy. If you keep waiting for a better PC, the one coming out next month will always be a better PC.

And there are plenty of people already building PC's to your specs. If the eggy boys don't wanna, they don't wanna, so don't pester them. They would have to raise their prices if they went there.

In my experience, the small companies run by young people who build fast gaming machines are your cheapest and best people to deal with.

Just order a large case with the biggest, quietest power supply.

Order the exact parts you've decided on.

In 6 months time it will be obsolete anyway - but do you want to record now or what?

Jump in - upgrade later. There is far more to recording than the PC - it's the cheapest, disposable part of your whole system.

Well it depends on the particular compressor. With software compressors, it's possible to be extremely accurate. You can precisely specify that if the signal exceeds the threshold, it will be precisely multiplied by the exact ratio specified.

Older analog stuff wasn't this accurate - you can imagine glowing lightbulbs and tubes and transformers and stuff. The parameters where more of a rough statement of intention.

Some of these analog compressors sounded very cool, so software models try to emulate their quirks.

I am fairly sure that in between the time the Threshold is exceeded, the envelope circuitry starts pulling the level down - towards the maximum reduction allowed by the Ratio parameter. This takes some time (specified by Attack) and therefore I don'

t believe an analog compressor is ever exactly multiplying all of the signal above the threshold exactly by the ratio the whole time.

Compression design is obviously an art - there's more going on that you would think.

With digital audio, each bit causes a doubling in volume, which is a 6dB boost. So 64 bit internal processing = 64 x 6 = 384dB.

But you don't use these big numbers for improved dynamic range, it's more about avoiding rounding errors when you process the signal.

The dynamic range of most audio sources is usually too great and needs controlling. Tape and vinyl was around 60 - 70 dB range, and many people think that was the ultimate. Digital allows for extreme dynamic range, suitable for classical music, but generally we like to squash music more.

The problem is you can squash it too much.

The thing about a compressor is that it's gain (or reduction) is constantly floating up and down - described as the Envelope. The parameters like threshold, ratio, attack, release define the limits that this envelope complies with, but it's constantly floating in between. There are other parameters that can be used to - like Hard Knee, Soft Knee or variations in between.

A compressor can't make abrupt changes, otherwise the signal would be distorted. That's why they use a Sidechain signal to create a smoother envelope that varies the gain.

One version of hardware compressor that was/is very effective is the optical compressor. That's where the signal lights up a light bulb or LED, and that in turn varies a photo-resistor that acts like a volume control. They aren't very fast - but that's part of why they sound so good. The slowness of heating and cooling the lightbulb allows for a smooth envelope.

Unfair to compare the tracking situation against modern mastered CD's. Totally different things.

Most modern CD's have been smashed with digital limiters with lookahead - and yes, those transients have been well and truely smashed off. But - you are also looking at the end result of probably thousands of processes and edits. Various analog and digital saturation effects have probably been used to tame a lot of transients.

The history of tracking with compressors comes from the days of magnetic tape. Magnetic tape smashed your dynamic range and your transients, and you didn't have to care about clipping A/D converters. You cared about Average levels, hence VU meters and tracking with slow (e.g. optical) compressors was never a worry.

With digital - Average doesn't cut it. Instantaneous peaks are the worry, as they will clip your converters. Very few compressors are fast enough to reduce a transient peak.

Or - if you have a very expensive compressor that can be set fast enough, I wouldn't like to use it as a matter of course since it would probably kill sound quality if set too agressively.

I doubt that anybody asking questions about compressors would own one of these compressors that are fast enough.

Software with lookahead is a different story. Anyone can own a fast compressor/limiter because they are free. But that's the wrong side of the A/D converters to be of any use when tracking.

Magnetic tape was about the best

Transformers

Tubes

Digital compressors and limiters can have the advantage of 'look ahead' that hardware compressors never had. That means they have see the audio stream coming, and take evasive action before the transient arrives.

But - be aware that when a limiter responds to a transient, it ducks the level and then whatever follows straight afterwards gets ducked as well. That's been described as 'punching holes in your sound'.

You can redraw some sharp spikes in your audio.

You can automate some fader movements