Kiwiburger

Digital DAWs can behave in unexpected ways. In theory, there is no reason why you should not use your Master fader for adjust the mix level. Why would the DAW maker give you a Master fader if you aren't supposed to use it? I have done experiments with mixing sinewaves in Cubase SX, and I can prove that it is possible to grossley exceed 0dBFS on all tracks, and export a massively "clipped" 32 bit floating waveform. If you play this back through 24 bit fixed converters, it is distorted to hell. But - if your re-import this waveform, and lower your master fader way down to the point where the waveform is below 0dBFS, the waveform cleans right up and there is absolutely no damage to the sinewave. You can phase invert against the original sinewave, and they cancel - zero damage done. In the real world, you can't get away with this. For a start, your A/D and D/A are analog devices with a definate sweet spot (typically around -18dbFS). But the real problem area is plugins. Some of these have serious internal clipping issues. Even when the input and output levels are way below 0dBFS, some plugins can distort badly. I would advise testing every plugin you use with sweeping sinewaves, and noting at what level they distort. Some will surprise you at how bad they are. So do as Where suggests - there are far too many problems with tracking and mixing too hot. The damage occurs before the master fader, so simply lowering it doesn't undo the damage.

May have misunderstood your question. The bundled plugins in most DAWs are usually fairly average. The good stuff is made by third party developers, and there are far more VST developers than DX developers. In my opinion, the best plugins exist as VST, not as DX. Start downloading the freebies and demos, and you should find out for yourself. The big advantage of VST, is that if you don't like it, it's no problem just to delete or move the .dll file to a another folder.

Automation. I would only buy a DX version if there was absolutely no VST version available, and there was no other way to get that particular effect. I also like the way VSTs are managed. DX has to be installed as a program. A VST .dll file is simply placed into your designated VST folder and it's good to go. This makes it really easy to manage - or at least I find this to be the case.

'Good tone' is a very subjective thing. PODs have been used in studio recordings - and possibly in some stuff that I like, I don't know for sure. Craig Anderton likes his Line6 stuff. And as he says, you only need to find one or two sounds that work for you to make it useful. All I know is that i've experimented with a lot of cheap guitar gear, and not so cheap guitar gear. I have a PodXT myself, because I wanted to see what the fuss was all about, and there is a ton of stuff out there that is clearly worse. Roland COSM stuff for a start. But I really don't like their AIR modeling - I have to disable that. It's a cheesy slap back echo and poor room reverb. You can do much better with a convolution reverb or any decent reverb. I find the effects are pretty average - I can happily replace them all with freeware VST plugins in a recording situation. The thing that annoys me the most is the wolf tones when you try to do solo stuff. Like a bad pitch shifter doing harmonies in the wrong key. I'm still on the lookout for the perfect guitar preamp - not a lot available in New Zealand, and i'm not sure I want to spend $1700 to get Marshal JMP-1 ... I am getting much better results using analog stomp boxes, together with something to give a basic clean amp eq curve. I think the Line6 stuff is tolerable, mainly because there isn't a lot of choice out there at the moment. Not in new stuff that I can purchase today from my country using VISA anyway ... There is some interesting preamps available if you live in the states.

However, I noticed that if I run a plugin on the master bus, it can cause problem if the pre-fader signal is too hot. I guess not all plugins like to receive a signal hotter than 0dB.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. Slamming into the masters will not sound as good as keeping proper fader levels ... the sonic difference is immediately obvious. 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. Anyway, I read somewhere that you should track with the level just high enough that with every fader at unity, you'd have your desired mixThat'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.

About the 32bit floating thingy...isn't that whole argument pointless if you track correctly, or am I missing something here? 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.

AD are not just analog. AD stands for analog to digital, the input is analog, the output(into your daw) is digital. That digital end has a finite limit, 0dbfs. If you want to keep insisting that you can go beyond that and still have audio that is musical, please post an example. Otherwise stop giving this bogus advise. It's well known fact that over0dbfs is digital cliping, unlike analog, it is anything but musical, 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.

Originally posted by wooden since i am going directrly into my 1616m i dont have good metering options. Can i insert this voxengo's free virtual VU meter in the track to measure it? is there any other better way? Originally posted by wooden since i am going directrly into my 1616m i dont have good metering options. Can i insert this voxengo's free virtual VU meter in the track to measure it? is there any other better way? 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.

Originally posted by wooden since i am going directrly into my 1616m i dont have good metering options. Can i insert this voxengo's free virtual VU meter in the track to measure it? is there any other better way? 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.