Fixed-Point vs. Floating-Point Digital Signal Processing (MOTU 2408 and 828?)

[part12studios]

I'm just curious if anyone knows if 828 mk1 or 2408 mk2 are fixed or floating.

I'm not sure what year exactly this became a thing.. original fixed was normal but floating was introduced at some point. i'm not clear when.

Thanks,
Caleb

[mikehalloran]

I'm just curious if anyone knows if 828 mk1 or 2408 mk2 are fixed or floating.

I'm not sure what year exactly this became a thing.. original fixed was normal but floating was introduced at some point. i'm not clear when.

Thanks,
Caleb

It isn’t a thing and never was. Internal processing is 24 bit. 32 bit floating is handled in the DAW. Both manuals say they are compatible with the 32 bit floating in AudioDesk.

The 828mk3 introduced 32 bit floating processing effects in CueMix except for the graphic EQ which uses 64 bit floating processing. Again, internal processing is still 24 bit.

If you look up each manual online and search (Control f or Command f) for floating, you will see this for yourself.

[part12studios]

Hi there, to be more clear this is the thing i'm talking about:

https://www.soundonsound.com/sound-advi ... nt-systems

This article is from 2004.. the MOTU 2408mk2 came out circa 2000 and the 828 Mk1 as well.. according to this article it seems that high end stuff uses floating while home studio / less expensive stuff uses fixed. So I would imagine that fixed is likely what MOTU gear of this era / price point was.

I was just noticing that i was running into a mix that was sounded really distorted yet nothing was above the 0db range in the audio spectrum.. but i'm thinking that i need to back down everything a little bit to give more headroom. So things are not just not hitting the red.. but actually a bit more below that to insure say a pronounced bassline doesn't end up garbling everything else as it comes in.. yet nothing ever actually clips..

just the first time i've run into this when recording something..

Thanks!
Caleb

[bayswater]

Hi there, to be more clear this is the thing i'm talking about:

https://www.soundonsound.com/sound-advi ... nt-systems

This article is from 2004.. the MOTU 2408mk2 came out circa 2000 and the 828 Mk1 as well.. according to this article it seems that high end stuff uses floating while home studio / less expensive stuff uses fixed. So I would imagine that fixed is likely what MOTU gear of this era / price point was.

The article is referring to DAWs and perhaps internal DSPs, not converters. As Mike says, any floating point processing would be going on in the processing of sound not the ADA conversions. I have the 2408-2 and 828-3 and it would be a great surprise to me to learn they were generating floating point digital signal for recording into a DAW.

References in the manual for the 828 mentioning floating point processing are referring to the built in mixer and effects, not the conversion. All of these are 32 bit floating point except the Eq which is 64 bit. These things are laid out in the 828-3 manual on page 13.

There are 32 bit converters, handy for recording gamma radiation and background noise from the Big Bang.

[mikehalloran]

I was just noticing that i was running into a mix that was sounded really distorted yet nothing was above the 0db range in the audio spectrum..

Happens to me all the time… but I'm now mixing 20–30 voice church choirs. If none of the meters in DP go above 50%, I have a chance at a clean mix but only a chance. I then need to tweak the individual tracks till the mix has the headroom I need. My interfaces have nothing to do with this—it's all in the DAW.

[part12studios]

that's good to know. i sometimes found myself wondering if the digital summing of DP 3.11 might have issues. Wondering if maybe taking the mixes into an analog board would sound better / offer more headroom, but i do believe just bringing down the collective volumes was all.. i'm just concerned of signal loss..

I take it that it's normal to take a final mix and run that through a pre-amp to boost a signal back up again? obviously not drastic boosting but just getting the signal back closer to but below unity gain.

[mikehalloran]

that's good to know. i sometimes found myself wondering if the digital summing of DP 3.11 might have issues. Wondering if maybe taking the mixes into an analog board would sound better / offer more headroom, but i do believe just bringing down the collective volumes was all.. i'm just concerned of signal loss..

I take it that it's normal to take a final mix and run that through a pre-amp to boost a signal back up again? obviously not drastic boosting but just getting the signal back closer to but below unity gain.

It all depends on what you're trying to do.

In this case, what looked like normal levels sounded like a wall of distortion. To save time, I cheated a little. I used the Declip plugin from RX8 on the Master. Then I lowered faders on the various tracks till it showed no more audio clips being repaired. I then put a DeEsser and my favorite reverb — again on the Master. I was done. Since the piece is just over a minute, this was fine.

https://youtu.be/BFEzXDaWkIU

The final result

https://youtu.be/8fY4LLtMoOM

[part12studios]

wow that's cool. so that's interesting that clipping could be detected yet the meter not hit the red. So in general though when recording we should still always record as "hot" as possible, but obviously no clipping. the headroom of 24bit is more in the mixing of of a song than the recording?

I'm always weary of noise / hiss creeping into a song, so the idea of not recording strong worries me, but maybe recording hot isn't always good for some things? i realize if you have an instrument that varies wildly in its volume / attack / etc.. compression could come into play to help or even limiters..

but are there times when something isn't recorded hot that isn't even itself something super dynamic?

[part12studios]

Also that's cool how you did that recording. i can definitely see how you have everything surprisingly low but the the master out has a nice strong signal. it's almost like there should be a meter that detects how "Wide" the mix is not just how tall it is.. like things are getting saturated but not clipping.. sounds like that plugin you mentioned is kinda like that, just does it in the background.

[FMiguelez]

wow that's cool. so that's interesting that clipping could be detected yet the meter not hit the red. So in general though when recording we should still always record as "hot" as possible, but obviously no clipping. the headroom of 24bit is more in the mixing of of a song than the recording?

I'm always weary of noise / hiss creeping into a song, so the idea of not recording strong worries me, but maybe recording hot isn't always good for some things? i realize if you have an instrument that varies wildly in its volume / attack / etc.. compression could come into play to help or even limiters..

but are there times when something isn't recorded hot that isn't even itself something super dynamic?

With 24 bits you have a ridiculous amount of headroom. The point of recording in 24 bits is so you can take advantage of that and not having to cram your levels all the way up like we used to with 16 bits.

When I record my individual orchestral instruments from the VIs, they usually hover between -20 to -10 dBFS. By the time they play together as an orchestra, levels are fine in the master and I never have to worry about clipping or distortion, even in full ff tutti.
For mastering, all there is left to do is to use a limiter to gain a couple of extra dB if necessary, but all the gain-staging was set correctly at the track level (otherwise you must pull all your faders down anyway, to the area where they don't have as much visual/mechanincal resolution; it's no big deal, but it's best to use good gain-staging from the start).

So don't cram your recording levels all the way up if you use 24 bits. Depending on your music and instrumentation, aim for between -20 to -8 dBFS for each track

[HCMarkus]

With 24 bits you have a ridiculous amount of headroom. The point of recording in 24 bits is so you can take advantage of that and not having to cram your levels all the way up like we used to with 16 bits.

Exactly. The noise to worry about is in your analog signal chain before the A to D.

The mix bus in DP operates at 32 bit floating point. The dynamic range that can be represented by a 32-bit (floating point) file is 1528 dB; compared to a 24-bit WAV file, the 32-bit float WAV file has 770 dB more headroom.

The distortion Mike is reporting has me wondering whether a plugin somewhere in his chain is inducing distortion.

https://www.centerforlydteknik.dk/downl ... -audio.pdf

How to Avoid Overloading a Plug-in
The traditional solution is to use the output gain knob in the previous plug-in to lower the signal being passed on to the next point plug-in. This is similar to the workflow of analog mixing. There is no quality loss with this method.

A similar option is to attenuate the input of the overloading plug-in. However, not all plug-ins have an input gain knob but most have an output knob, which is why the above solution is usually better. Another option is to insert a dedicated gain or trim plug-in before the overloading plug-in. Use the gain plug-in to lower the level of the signal before it reaches the next plug-in. There is no quality loss with this method either.

How to Avoid Overloading a Bus or Master
If you are not using any plug-ins on the bus or master channel you can simply lower the relevant bus or master fader to remove any overloads without problems.

If you are using plug-ins then the best option is to use a gain or trim plug-in as the first insert on the bus, and use that plug-in to lower the level until it is safe again.

Levels when Processing with Plug-ins
Aim for a peak of around -6 dBFS on individual channels to be on the safe side, though you are likely not to experience any problems using higher values. This value is not necessarily the peak value shown on the output meter of the channel since that value can be offset and hidden by a lowered channel fader, as explained earlier. Rather it is the internal level of the signal chain which you can see by setting the channel fader at unity or using a pre-fader metering option in your DAW.

[mikehalloran]

The distortion Mike is reporting has me wondering whether a plugin somewhere in his chain is inducing distortion.

Nope. Nada. Zip. Zed. No plugins were used on this except RX8A Declip on the Master. When it showed me 0 clipped intervals repaired and my ears agreed, I removed Declip, inserted MOTU De-Esser followed by Eventide's SP2016 Reverb — again on the Master.

The quality of the tracks I receive ranges from mediocre to pretty bad. All volunteer old voices, mostly singing into their Androids and iPhones, then uploading the resulting mp3 and mp4a. A couple are using GarageBand over iOS but that's it. Some tracks are noisy while others are less so but none are what I'd call good. There are a couple of singers whose tracks need to be Normalized before I pull them back -20dB.

I'm not against plugins. On some songs, a few singers get the MW Limiter to tame peaks but I didn't need that here. Overuse of the Limiter—aggressive settings, too many tracks or both—will make the mix sound like I ran it through my 50 year old Big Muff π stomp box. Every once in awhile, I need to use RX8 Spectral De-noise or De-click on one or more tracks but I'll bounce the results to Mono -3dB and delete the original tracks to keep the overhead lower.

With RX8 Advanced, I can use De-clip, De-noise or De-click as a stand alone plugins instead of calling up the entire suite and using them from within. If in a hurry, I sometimes leave De-clip in; it's quite good at what it does but I prefer it as a diagnostic tool.

The only pitch correction I ever use is DP — and then, only on my basses if they have a sustained Eb or lower. I correct individual notes, never a track. Besides the sheer number of extra hours it would take, what happens when these choirs sing live again after I've spent all that time making them perfect in my studio? Nope, not going there.

Now the above only applies to my choirs. With my praise bands, I'm producing records — nothing is off the table.