which has more head room protools or cubase

[pedrogoya]

Its all about mixing it right mate. I dont think theres a different "headroom" from daw to daw. If it clips on pro tools...will clip on cubase also.

[joeebangs]

i figured that just had hope i guess lol

[Kiwi]

Cubase uses 32 bit floating point for internal maths. Depending on what version of Protools, it's 48 bit fixed or 32 or 64 bit floating. The point is - in the digital domain, it's practically impossible to clip audio ....

Pause for effect ...

The reason that your audio clips and sounds like shit is because your D/A converters are 24 bit fixed. So as soon as you exceed 0dbfs it clips. OR - if you bounce down to 24 bit fixed, it will clip the same way. But internally, the waveform never clips, becauses it is represented extremely accurately by those massively large numbers that are possibld with those formats.

You can prove this any time. Create a sine wave track, and then apply enough gain to force it to clip like a nasty square wave. Save that file as a 32 bit floating file. Do you think your sine wave has been destroyed forever? It's actually quite safe. Load that file into a new project ... notice how badly it clips and sounds destroyed. Now reduce the gain so it doesn't clip ... and you can see nd hear that you have your sine wave back with no damage. It's a miracle!

It's just how digital audio works. But you have to make it work with your analog converters - that's why need metering.

Also - a lot of plugins are design to work just like analog, so they re desgned to saturate and eventully clip, just like analog. That's why they clip - it's not a weakness of digital.

With intensive summing (which is really multiplication) it's possible for the internal maths to get really huge. 32 bit floating can handle thiis, but the thing about bit depth is that it defines your noise floor. 64 bit floating just keeps the noise floor much further away, but in practice it's fairly academic.

[chrysb93065]

Well, PT IS very pissy about clipping - requiring very careful gain-staging control - ride faders, limiters, etc. The others, including are more forgiving - when it hits red, it's not REALLY at clip - just very close.

[rumbletone]

If you are constantly clipping, perhaps your levels are too high? PT doesn't determine the input levels - your pre-converter analog levels do - which you determine, not PT. Once ITB, a properly recorded signal will clip only if you add processing or sum without properly adjusting your levels - again, it's the engineer who sets the levels and other parameters on the processing.

One useful tip I've seen is to default all channel faders to -12 (not 0, for example). The noise floor is so low on most pro set-ups these days that you needn't be anywhere near red on your channel meters - try to get them peaking just into yellow - that should be more than high enough.

Gain staging is a key element of audio recording. If you are not prepared to pay attention to it, then slap a brick wall limiter before and after every gain stage and then nothing will clip - it will sound like crap, but you won't see any red lights . . .

[projektk]

I tend to track at half way volumes coming into pro tools. No matter how many time I ask how loud is your loudest part in the song and test it, they always go louder LOL.

Sent from my LG-P925 using Gearslutz.com

[joeebangs]

nothing is hot waves start out fine, once i start mixing half way thru some sounds start to hit red i k now better gain staging an mixing will help but is there a software better at it im using digi oo3 pt 8.4 which i beleave is only 24 bit correct ? which version has 32 floating bit ?

[joeebangs]

anyone think a lynx aurora ad da into a dbox will help with this ?

[joeebangs]

another question how acurate are those stupid red lights ? i see red but dont hear clipping i dunno about this digi oo3 , i also cant bounce when i bounce i loose allll detail an panning gets weak total diff sound , i record into an audio track an save the file from audio list an much better sound

[Arksun]

Originally Posted by Kiwi

Cubase uses 32 bit floating point for internal maths. Depending on what version of Protools, it's 48 bit fixed or 32 or 64 bit floating. The point is - in the digital domain, it's practically impossible to clip audio ....

Just to add, whilst the 32-bit float engine in Cubase does allow you to overshoot individual channels and groups, if you run into the red on the main output bus, you ARE clipping realtime for Cubases output before hitting the audio interface mixer and D/A. You can test this by running into the red and turning down the volume in your audio interface mixer hitting the D/A quieter, that sine thats turned into a square will still sound like a square. Only if you render that offline to 32-bit float will the whole sine wave be stored.

[kalle74]

Probably a troll. Or total stupidity. I'll have a go anyway...

Gear is not to be blamed if you're not leaving enough headroom.

Learn to create and leave room to your mixes via sonics, mixing is not pushing everything louder than everything else. This has been discussed over zillion times before. Search.

Headroom is not an issue in 24-bit audio. User errors are.

Also, work on your PUNCTUATION and spelling.

[Dr sampson]

anybody else notice that in PT you can audibly hear a clip @ any gain stage in the software, whereas with cubase any overage just continues to the next sum. in cubase you actually have to clip an output channel to hear it

[joeebangs]

wow some of you douchbags get real touchy in this room dam geeks lol although yes skills have most to do with it you can bet ya ass off some equipment does make a diffrence,

[s.d.finley]

Originally Posted by joeebangs

anyone think a lynx aurora ad da into a dbox will help with this ?

Nope, but keeping your input levels lower while recording will help.

[Kiwi]

He is clearly a genius who has discovered a fault in Protools that nobody else has ever discovered ...

[Serpico]

One thing that might give the impression of different headroom betweens DAWs is the default panning law - a pan that sums to center at +6db will give higher buss levels for the same set of input levels as a 0db centre sum.

[stinkyfingers]

the problem is you are trying to use Pro-Tools and you are not a Pro...

[stakeoutstudios]

There is a difference in headroom between TDM systems and native - up until the release of HDX which put the DSP systems on the same 32 bit floating headroom as native.

Of course with proper gain staging it's possible to get great mixes, but it is true pro tools TDM was easier to clip.

In a 32bit float system, no matter how much your clip lights come on in tracks and auxes as long as your master is down and you're not upsetting any plugins, it won't be clipping.

You can actually switch the clipping indicators off in preferences if it makes you feel better. Those indicators are really there to warn you on TDM systems you would clip at the 24bit insert points. TDM plugins themselves are 48bit fixed point and so have internal headroom, but when audio leaves the plugin, it must be below 24 bit full scale.

In a 32 bit float native system, LE, Pro Tools 9/10, Cubase, HD native, HDX, Logic etc... These issues don't exist if the plugins are coded correctly. The effective headroom is over 900 db, which means even summing multiple tracks if you turn down the fader they are being sent to, it won't clip. Easy to test.

There is not as much headroom in TDM systems due to the nature of the DSP chips being fixed point. The same rules don't apply. If you send multiple hot tracks to an aux, and turn the aux down you will visibly see and obviously hear the transients have been destroyed. There is decent headroom in TDM so I'm talking a lot of tracks summing to one aux. Again, easy to test and prove, I've done it. On 32bit float systems the dynamics remain on an identical session set up.

This is not the same as setting your session to 32bit float as you can in Pro tools 10. This simply adds the headroom built into the mixer into the audio files, so if you exceed full scale with audiosuite or clip gain, the file itself is not destroyed.

It's always good to practise good gain staging, and slapping a brickwall limiter on every aux etc is not a good plan as you're effectively removing any headroom built into the DAW. Limiters are for the master buss only, unless for creative purposes. Before it hits your D/A which cannot exceed full scale without clipping. This doesn't mean you should always limit your master buss - if your track is going to be mastered afterwards you definitely shouldn't. Just make sure you always create and keep a close eye on your master fader.

There's so much myth and misinformation out there about all this, it's best to sit at your DAW, sum some incredibly loud signals, turn down the fader they are being sent to and prove it to yourselves. Check your plugins behave as you'd want with a hot input if you're planning on doing that too!