Can you be more specific than "crazy low"? What are your peak levels?

The +4/-10 switch controls sensitivity, -10 being the higher sensitivity setting. Most likely the -10 position is giving you more level. But if the drum module is putting out "instrument" level it may still be too low for a -10 line input. It may be that you need a dual channel direct box to connect your module to the mic inputs of the interface. If there were an instrument/guitar input I'd suggest using that, but I don't see one in the Google pics.

The -10dB / +4dB switch is not a limiter. It is to set your input level to the appropriate signal strength. Typically an unamplifed instrument would use the -10dB setting. As far as I know the outputs are unbalanced ¼" and you should be using mono instrument cables.

Check out post #9 here:

https://www.vdrums…

Which 1/4" jack output of the TD-11 are you using?

If you want a mono input via your Mbox, you should use a TS-TS (mono) jack cable from the L/mono output of the TD-11 into one channel of the Mbox. If you want stereo recording, simply use a second TS-TS cable from the R output of the TD-11 into another channel of the Mbox. The (lesser) alternative is to use an insert cable from the stereo headphone output of the TD-11 going to the jack inputs of two channels on the MBox.

What you should not do is connect a TRS-TRS (balanced) jack cable from the TD-11 headphone output to an input of the Mbox. This would result in a weak and thin recorded signal, as the Mbox will record the difference between the L and R signals.

dvdhawk, post: 440930, member: 36047 wrote: The -10dB / +4dB switch is not a limiter. It is to set your input level to the appropriate signal strength. Typically an unamplifed instrument would use the -10dB setting. As far as I know the outputs are unbalanced ¼" and you should be using mono instrument cables.

Does it matter if the 1/4" are stereo? They are both stereo...(the ones I'm using). I thought you could use stereo for mono, but not mono for stereo (for obvious reasons)?

bouldersound, post: 440929, member: 38959 wrote: Can you be more specific than "crazy low"? What are your peak levels?

The +4/-10 switch controls sensitivity, -10 being the higher sensitivity setting. Most likely the -10 position is giving you more level. But if the drum module is putting out "instrument" level it may still be too low for a -10 line input. It may be that you need a dual channel direct box to connect your module to the mic inputs of the interface. If there were an instrument/guitar input I'd suggest using that, but I don't see one in the Google pics.

I'd have to look, which I will do. But at full tilt on the drums, I'd get 3 green bars in the PT meter.

That #9 post looks helpful, I will try that and report back. The volumes are all the way up, but I never noticed "sensitivity" settings.

Someone from Roland got back to me and said I may need to up the firmware, and I'm on 1.14 and current is 1.15

Boswell, post: 440932, member: 29034 wrote: If you want a mono input via your Mbox, you should use a TS-TS (mono) jack cable from the L/mono output of the TD-11 into one channel of the Mbox. If you want stereo recording, simply use a second TS-TS cable from the R output of the TD-11 into another channel of the Mbox. The (lesser) alternative is to use an insert cable from the stereo headphone output of the TD-11 going to the jack inputs of two channels on the MBox.

What you should not do is connect a TRS-TRS (balanced) jack cable from the TD-11 headphone output to an input of the Mbox. This would result in a weak and thin recorded signal, as the Mbox will record the difference between the L and R signals.

Between all the above advice I'm sure I will find the problem. Thank you so much!

Oh, and just fyi, I tried jumping the 1/8" headphone out, to RCA to the aux input of the Mbox Pro. And it was considerably louder than using the standard line out.

Ill let you know what it turns out to be.

Have you gone to the site I linked to? It seems to have the definitive solution that renders all solutions offered here, including mine about using DIs, irrelevant.

I may be wrong, but I'm wondering if referring to the cables as "mono" and "stereo" might be causing some confusion. All stereo ¼" phone plugs are TRS, but not all ¼" TRS phone plugs are used in stereo applications. In pro-audio the TRS connection (which stands for Tip/Ring/Sleeve) can be used in lieu of an XLR in many cases, and similarly uses the 3rd conductor (in this case the Ring) for balancing a (mono) signal. Balanced signals are vastly superior over distance, and used in most pro applications, because of their ability to reject noise. The Tip is the normal signal, the Ring is the inverted version of the same signal, and Sleeve connects to the shield /ground to help reject noise. If you need a more thorough explanation, you'll have to ask.

As to your question, yes it can matter a lot, and may explain at least part of your problem. Your MBox uses 2 "XLR combo-jacks" on the front, and what are sometimes described as "auto-sensing" ¼ inputs on the back. Both can accommodate either TRS ¼" balanced connections, or ¼" TS unbalanced TS connections. When the MBox input 'senses' that the Ring and Sleeve of the TRS are not shorted together, as they would be on a TS plug, it's probably expecting a balanced (typically line level) signal. The auto-sensing circuitry then often automatically reduces its sensitivity expecting something that's already been amplified to some extent.

The TD-11 manual is woefully uninformative, it clearly illustrates using normal unbalanced instrument cables to connect to an amplifier, but in the specs only states the output jacks are "(L/MONO, R) Stereo ¼-inch phone type", which I honestly believe is a typo. I've worked with several Roland kits including TD-8, TD-12, and TD-20 (with DIs) in live sound and recording situations, and to the best of my knowledge the outputs on your TD-11 are unbalanced, so I'd use 2 normal TS instrument cables from the drum-brain L/R to the Mbox inputs of your choice. I'd set the sensitivity switches on the corresponding MBox channels to -10dB. And I'd make sure I read the link Bouldersound provided and check the pad sensitivity settings, to make sure the output levels aren't getting choked before they even get a chance to get out of the drum module. Luckily, there's metering built-in for that purpose.

I'm not surprised the headphone output was hotter signal, it's way more amplified. It's intended to be powerful enough to drive speakers, albeit small ones. It's not recommended in terms of audio quality, and could theoretically damage either the headphone amp or the input, but rarely does. However, your RCA inputs are definitely -10dB, and probably don't really appreciate having an undisclosed number of milliwatts rammed up their snout. (to paraphrase Zappa)

Best of luck!

bouldersound, post: 440963, member: 38959 wrote: Have you gone to the site I linked to? It seems to have the definitive solution that renders all solutions offered here, including mine about using DIs, irrelevant.

I did buddy, I just haven't gotten around to trying it yet. Holiday weekend and all. But I'm turning the lights on right now. Thank you for that. I did search for anything I could find. It never occurred to me to go to a Roland Drum forum and search there. So thank you for alleviating my stupidity.

dvdhawk, post: 440975, member: 36047 wrote: I may be wrong, but I'm wondering if referring to the cables as "mono" and "stereo" might be causing some confusion. All stereo ¼" phone plugs are TRS, but not all ¼" TRS phone plugs are used in stereo applications. In pro-audio the TRS connection (which stands for Tip/Ring/Sleeve) can be used in lieu of an XLR in many cases, and similarly uses the 3rd conductor (in this case the Ring) for balancing a (mono) signal. Balanced signals are vastly superior over distance, and used in most pro applications, because of their ability to reject noise. The Tip is the normal signal, the Ring is the inverted version of the same signal, and Sleeve connects to the shield /ground to help reject noise. If you need a more thorough explanation, you'll have to ask.

As to your question, yes it can matter a lot, and may explain at least part of your problem. Your MBox uses 2 "XLR combo-jacks" on the front, and what are sometimes described as "auto-sensing" ¼ inputs on the back. Both can accommodate either TRS ¼" balanced connections, or ¼" TS unbalanced TS connections. When the MBox input 'senses' that the Ring and Sleeve of the TRS are not shorted together, as they would be on a TS plug, it's probably expecting a balanced (typically line level) signal. The auto-sensing circuitry then often automatically reduces its sensitivity expecting something that's already been amplified to some extent.

The TD-11 manual is woefully uninformative, it clearly illustrates using normal unbalanced instrument cables to connect to an amplifier, but in the specs only states the output jacks are "(L/MONO, R) Stereo ¼-inch phone type", which I honestly believe is a typo. I've worked with several Roland kits including TD-8, TD-12, and TD-20 (with DIs) in live sound and recording situations, and to the best of my knowledge the outputs on your TD-11 are unbalanced, so I'd use 2 normal TS instrument cables from the drum-brain L/R to the Mbox inputs of your choice. I'd set the sensitivity switches on the corresponding MBox channels to -10dB. And I'd make sure I read the link Bouldersound provided and check the pad sensitivity settings, to make sure the output levels aren't getting choked before they even get a chance to get out of the drum module. Luckily, there's metering built-in for that purpose.

I'm not surprised the headphone output was hotter signal, it's way more amplified. It's intended to be powerful enough to drive speakers, albeit small ones. It's not recommended in terms of audio quality, and could theoretically damage either the headphone amp or the input, but rarely does. However, your RCA inputs are definitely -10dB, and probably don't really appreciate having an undisclosed number of milliwatts rammed up their snout. (to paraphrase Zappa)

Best of luck!

Thanks a lot. That was super informative. I always thought they were interchangeable going from S to M, just not from M to S.

I actually don't have any mono cables, so switching them out will take me a couple days.

I use the back jacks on the Mbox Pro, but it seems it will take whatever you give it. I've literally just never plugged a mono cable into it.

Thanks!

Brother Junk, post: 441062, member: 49944 wrote: I always thought they were interchangeable going from S to M, just not from M to S.

And that's the worst of it, sometimes they ARE interchangeable, and sometimes they aren't. It completely depends on what you're connecting together.

Brother Junk, post: 440959, member: 49944 wrote: Does it matter if the 1/4" are stereo? They are both stereo...(the ones I'm using). I thought you could use stereo for mono, but not mono for stereo (for obvious reasons)?

Yes it mathers, if an input or output is unbalanced, using a balanced cable (stereo as you say) will cancel a part of the signal...

Lower your volumes and gains and use a ''mono'' cable.

So I messed with it yesterday and here are some things I noticed. Keep in mind, I have not owned the Mbox Pro for very long (nor do I have as much experience/knowledge as you guys)

Plugging into the rear jacks, and using the sensitivity switch, they are very quiet. The switch does make a difference, but it's slight. Just enough to be noticeable. No matter how you set the inputs on the front (to be line input in the back or the front jacks) the front knobs will not adjust levels when the jacks are plugged into the rear. Maybe that's just how it works (it would make sense).

Plugged into the front 1/2 the knobs now work, and I can get more volume out of it, but it's still very low. To get anything that would be considered normal volume, I have to crank it so high, all I can hear is the noise floor.

bouldersound, post: 440929, member: 38959 wrote: Can you be more specific than "crazy low"? What are your peak levels?

Ok, on the drum module, I peak out at 4 bars...and extremely occasionally, I'll hit 5 (peak). In PT, Im barely making 3 green bars. I recorded something yesterday, and with the volume levels set to "reasonably audible" the recorded track barely looks like a line, not the typical large waveform, with peaks and valleys. Just a tiny squiggly line.

I found the sensitivity setting for the pads like you said. Unfortunately, if I put them much above 10 the module shows me clipping. I can also hear the clipping. It was helpful to find out those were there though.

dvdhawk, post: 441063, member: 36047 wrote: And that's the worst of it, sometimes they ARE interchangeable, and sometimes they aren't. It completely depends on what you're connecting together.

That would explain my success with having them be interchangeable up until now.

pcrecord, post: 441070, member: 46460 wrote: Yes it mathers, if an input or output is unbalanced, using a balanced cable (stereo as you say) will cancel a part of the signal...

Lower your volumes and gains and use a ''mono'' cable.

Ok, so based on everything (and thank you all!!!!!) it seems like the cables are indeed the problem. And I forget who said it, but I think someone said that the back line inputs are different than the front inputs. Aside from one being the -10/4 choice in line level adjustment vs a knob, I think someone said that the front ones will sense what type of jack you put in? I'm unsure if that is what causes the notable volume difference (Though either one is way too low).

The front inputs are louder, but I would prefer to use the rear inputs. So, I guess I need two mono cables to proceed from here. Which kind of blows, bc I have every cable you can imagine...except mono's. But isn't that how it always works....

But thank you all again, I will keep you apprised of what I figure out.

GO AND BUY UNBALANCED CABLES !!! 90% of chances, that's your problem!
It is normal that the rear and the front don't have the same levels. The back are line inputs and fronts are Mic inputs. Not at all the same level !
No need to pick your nose further, see the page on the manual ; it seems to show unbalanced cables doesn't it ? (sorry I hate repeating myself) ;)

BTW guitar cables are unbalanced (mono)

Brother Junk, post: 441099, member: 49944 wrote: Ok, on the drum module, I peak out at 4 bars...and extremely occasionally, I'll hit 5 (peak). In PT, Im barely making 3 green bars. I recorded something yesterday, and with the volume levels set to "reasonably audible" the recorded track barely looks like a line, not the typical large waveform, with peaks and valleys. Just a tiny squiggly line.

What are "bars"? Give us the levels of your recorded tracks using numbers in dBFS. And, yes, get unbalanced cables. Though I'm not convinced that's the problem it would be very useful to know for sure one way or the other.

And have you tried the suggestions in the link I posted? I found that by searching "TD-11" and "low volume". A bunch of posters seemed to have exactly your problem and the fix was something in the module's settings. Most of them seemed to have the problem with TD-15 modules but I think it was the same fix for the TD-11.

bouldersound
pcrecord
dvdhawk
Boswell

First of all, thank you all for all the suggestions and sites. I'm poor right now, so I had to borrow the cables.

I borrowed some mono cables. Both speaker and shielded. And the volume increased x10. Tracks in PT just fine etc. It turns out the volume knob that I thought just controlled the headphone output, also controls those line outs. With stereo cables, the volume was so low, that messing with that knob made no noticeable difference.

Lesson learned, thank you all. Stereo cables don't always work in mono situations. I turned the pad sensitivity up to 12 iirc, and I had to turn them back down to 8.

bouldersound, post: 441104, member: 38959 wrote: What are "bars"?

The module has an output meter. 4 solid bars, and a final hollowed out bar that indicates clipping (or close to it). It's like a typical output meter. When I turn the pad sensitivity up, it clips.

bouldersound, post: 441104, member: 38959 wrote: And have you tried the suggestions in the link I posted? I found that by searching "TD-11" and "low volume". A bunch of posters seemed to have exactly your problem and the fix was something in the module's settings. Most of them seemed to have the problem with TD-15 modules but I think it was the same fix for the TD-11.

Yeah, it just took me a bit bud. House renovations are being finished up, in-laws are coming to stay, had to pick up the cables from my father's band's practice spot. [side note, I'm not sure if you can stack possessives like that, father's band's? Not sure if that's correct or not]

But that's what the problem was, simply the mono cables. In previous situations, I've always been able to use stereo cables in mono situations w/o problem. I see now that it matters, sometimes.

I tried mono speaker cable, 1/4" to 1/4", also shielded of the same. There was no difference. In researching a little further, I might try them again and try to listen for background or ground hum.

I borrowed the cables...I wanted to try them first before I spent money...in case it wasn't the problem. But mono works. So, is there any reason for me to get shielded vs un-shielded?

***Also, I would prefer to use the inputs on the back. Long story short, I didn't try the rear inputs, but I'm assuming they will work correctly too now. Any reason I should use one vs the other? And with the Mbox Pro, if I use the rear inputs, can I also use the front inputs at the same time and just use the switch? Meaning can I keep stuff plugged into both and it will be fine?

Basically, I want to keep the drums plugged in - all the time - to rear 1/2, the guitar mic is on 3, the vocal mic is on 4. The KB is usb. If I have a bass player or someone else come, can they just plug into the front jacks and I hit the switch to move it to the front input? Or would I (for some reason, I just want to make sure) have to unplug the drums?

I'd like to keep the Mbox Pro as my DA for a while, before upgrading to something with more inputs, so I just want to make sure I don't break it.

I'm glad you got it to work.
And thank you for coming back and keep us informed ;)

pcrecord, post: 441625, member: 46460 wrote: I'm glad you got it to work.
And thank you for coming back and keep us informed ;)

What can I say, I enjoy it when other people learn from my stupidity. And I appreciate all your help. I read everything that everyone has posted, I just have a lot going on right now so I have to slip these studio setup activities in the times I have between.

You can see it in the diagram you posted, but L is marked as mono. As a drummer yourself, would you do L/R? Or mono? It works the same, just stereo vs mono...

I realize it can be done either way, I'm curious as to how you would choose to do it?

I'm glad you got it sorted out, but hope you didn't end up keeping any speaker cables connected between the Roland and the MBox.

It would be OK for the sake of experiment, but a shielded TS instrument cable is the correct cable for the job. The shielding that an instrument cable would provide would be beneficial with all EMI and RFI your computer and other devices emit. A 20ft. speaker cable (which should be UNshielded), connected to a sensitive input like a mic-pre acts essentially like a 20ft radio antenna (aerial for our British friends).

dvdhawk, post: 441631, member: 36047 wrote: I'm glad you got it sorted out, but hope you didn't end up keeping any speaker cables connected between the Roland and the MBox.

It would be OK for the sake of experiment, but a shielded TS instrument cable is the correct cable for the job. The shielding that an instrument cable would provide would be beneficial with all EMI and RFI your computer and other devices emit. A 20ft. speaker cable (which should be UNshielded), connected to a sensitive input like a mic-pre acts essentially like a 20ft radio antenna (aerial for our British friends).

Thank you, you just saved me all sorts of headache...

Brother Junk, post: 441627, member: 49944 wrote:
You can see it in the diagram you posted, but L is marked as mono. As a drummer yourself, would you do L/R? Or mono? It works the same, just stereo vs mono...

I realize it can be done either way, I'm curious as to how you would choose to do it?

Actually, 2 connectors one left and one right meens they are both mono signals. If you want you could send them to 2 seperate self-powered speakers. IN MONO.
In the DAW, you can use 2 mono tracks or combine 2 mono input to make a stereo input. But understand that if you leave the TD11 with 2 cables, they stay mono signals until combine in the DAW.. Clear ?

pcrecord, post: 441635, member: 46460 wrote: Actually, 2 connectors one left and one right meens they are both mono signals. If you want you could send them to 2 seperate self-powered speakers. IN MONO.
In the DAW, you can use 2 mono tracks or combine 2 mono input to make a stereo input. But understand that if you leave the TD11 with 2 cables, they stay mono signals until combine in the DAW.. Clear ?

I understand that stereo is the result of two unique mono signals. I see why you were clarifying when I read my post, and I see what you mean, it would only be stereo if it was a single jack. What I have is dual mono. I meant running them essentially in stereo by using the L/R. How do I do that? Or is that even what I should do?

I think what I should have asked is - Would you run it as a single mono signal into a single track?
Or run left and right, two separate channels, two separate tracks? I've never worked with an instrument like this in a daw...it's always usb or singularly mono like a mic or guitar. It seems like, for recording this instrument, I want the stereo effect recorded in there...yes?

pcrecord, post: 441635, member: 46460 wrote: combine 2 mono input to make a stereo input.

I think that's essentially what want to do in PT10. Combine channels 1&2 to make a single stereo track...? Maybe lol? I guess what I'm trying to ask bc I understand so little about it, is just how would you do it?

I usually send stereo instruments 1 stereo track unless I fear the could unbalanced somehow.. Even in mono tracks I can link them in Sonar and the faders will follow each other ; easier to manage ;)
So yes, 1/2 to make L/R seems like a good Idea !

pcrecord, post: 441642, member: 46460 wrote: I usually send stereo instruments 1 stereo track unless I fear the could unbalanced somehow.. Even in mono tracks I can link them in Sonar and the faders will follow each other ; easier to manage ;)
So yes, 1/2 to make L/R seems like a good Idea !

I've never even seen Sonar, I hear a lot about it.

In order to get inputs 1 and 2 to combine to a single track in Pro Tools, do I have to mess with the flow setup in the Mbox?

I am very dumb with this kind of stuff, so I may just be missing it. But I didn't see a way in PT to just combine the channels in the edit or mix window. But I think that if I go into the Mbox program I can do it.

I will look it up when I get home in a few hours, but if anyone just happens to know, I'd appreciate it.

Most DAW including Protools will let you choose your inputs as mono or stereo tracks. The only limitation is that you have to chose consecutive inputs pair : 1/2, 3/4 etc... you can't choose 1 and 5 or 2 and 3...

Create 1 stereo track and then assign a pair of inputs to make stereo :

You might want to consider checking some tutorial like in this play list :

pcrecord, post: 441668, member: 46460 wrote: Create 1 stereo track and then assign a pair of inputs to make stereo

I'm using the shortcuts too quickly and assigning it as a mono track :( That's why it was only giving me the option of 1 or 2, instead of 1/2.

I feel incredibly stupid.

You'll get there.. Protools is not the most simplified DAW.. it needs some Learning and getting used to

pcrecord, post: 441688, member: 46460 wrote: You'll get there.. Protools is not the most simplified DAW.. it needs some Learning and getting used to

I appreciate the soft let down....

But that was just plain stupid on my part.

I agree, it's not the easiest DAW. I started with Ableton Live for Dj-ing, then Logic just goofing around for myself, but when I started going to studio's they all use PT, so, I figured I'd best learn it.

Thanks again to all. The drums are working perfectly now.

Except for the USB connection and PT instrument plug-ins. E.g. I can't choose a drum kit from PT and use my drums. 1/16th hits or higher it can't keep up. However, I've looked up all I can about it, and it seems it's just a problem no one has managed to solve. The included kits are pretty good though, so, it's not bad.

These things are a boat-load of fun. The toms are hard to arrange the way I want, but I suppose I'll get used to 'em

Brother Junk, post: 441761, member: 49944 wrote: I can't choose a drum kit from PT and use my drums. 1/16th hits or higher it can't keep up.

Drums are so time critical, I always make the player hear the TD module and record only the midi.. Once it's in the box, I activate a VSTi in my DAW.
Playing directly through a vsti work well when there isn't a lot of processing in the project, the more stuff going on the more latency you get..
Usually I don't take any chances...

pcrecord, post: 441762, member: 46460 wrote: Drums are so time critical, I always make the player hear the TD module and record only the midi.. Once it's in the box, I activate a VSTi in my DAW.
Playing directly through a vsti work well when there isn't a lot of processing in the project, the more stuff going on the more latency you get..
Usually I don't take any chances...

If I understand you correctly, you let the person listen to themselves, on a TD-11 kit, playing through headphones or whatever, while the daw is recording the midi...bc the latency is caused by the plug-in being active during the midi input?

So if you record the midi without the vst, and then apply the vst, there is no latency?

I do a lot of back reading here, and I read that expression, "in the box" a lot. Is that fundamentally what you guys mean by ITB? That you put the midi in first, then apply the vst?

That would be awesome if I'm understanding you correctly. I didn't even know you could do that. I'm still getting used to the professional vernacular you guys use, so sometimes when I read, it's with the intention of understanding later on.

I love this place!

If that is what ITB means, I think I understand it now.

**Edit, what do you typically set your latency to?

Brother Junk, post: 441763, member: 49944 wrote: If I understand you correctly, you let the person listen to themselves, on a TD-11 kit, playing through headphones or whatever, while the daw is recording the midi...bc the latency is caused by the plug-in being active during the midi input?

Yes, I take the audio output of the TD and plug it to my audio interface then I send the computer and the TD signal to the player headphones (my interface has a realtime mixer, so should yours) At the same time I record the midi data to a track in my daw

Thing is, it takes more time to the computer to receive midi information and send it to a VSTi and produce sound and send it back to the player headphones.
Only recording midi and do the monitoring in realtime makes it easier on the computer.
Some will be able to play through a VST while recording, but this means having very low latency. (Latency being the time the computer takes to grab a signal, process it and playing it back.)

ITB, in the box, means you do in within the computer instead of going outside to external gear. Ex : if I mix ITB, no signal is going to outboard gear, I'm using just the DAW and plugins.. OTB would be the out of the box version where external gear process the audio.

Brother Junk, post: 441763, member: 49944 wrote: **Edit, what do you typically set your latency to?

Usually buffer at 256 is the starting point but it depends on your computer performances and how busy the computer is with other activities. If you have an antivirus in memory you make have more latency and therefor risk of crashing the playback if the buffer is set too low.

I can go in more details but it's basicaly it...

Brother Junk, post: 441763, member: 49944 wrote: f that is what ITB means, I think I understand it now.

As Marco mentioned, "ITB" (In The Box) means that - with the exception of getting your initial signal into the DAW - all of your mixing, processing, editing, etc., is all done within the DAW program.
Those who are strictly ITB aren't using any external hardware for processing ( compressors, limiters, EQ's, special FX, etc.)
Now, they may "front load" the input signal with some type of particular analog processing on the way into the DAW( using Outboard, or "OB" gear) such as channel strips, gain reduction, EQ, etc., but once the signal is recorded and resides on a track within the DAW, any further processing of that audio for the final mix is done within the DAW itself.

To take the explanation a bit further...
Accordingly, "OTB" ( Outside The Box) means that external outboard processing gear is actively being used during the mix. This might be an external analog console, or maybe OB compressors, limiters or EQ... This workflow is often referred to as a "hybrid" mixing set up, because it's using both digital and analog processing...

There are those who feel that - while plugs have done a very admirable job of simulating certain outboard hardware pieces, that only the real hardware can truly give you "the" sound that a particular piece is known for.
Some examples of these external pieces would be ( but not limited to) processors like Urei 1176's ( black, silver, and blue stripe models), or Teletronix LA2A's, LA3's, etc., Focusrite Reds, and even particular channel modules from classic consoles like Neve, Harrison, Focusrite, Universal Audio, SSL, Trident, etc.

It really all depends on whom you ask, and their own perception of the accuracy of the digitally simulated models of these pieces, and, it also depends on the particular hardware vs. plug-in version that is being discussed.
For example, there are some who claim that the Waves, T-Racks, Slate and UA SSL collections are as close as you can get to the real thing - without actually using the real thing... and, these opinions are coming from people who actually spent time on real SSL E and G Series desks in the past.
(BTW, On a personal note, I happen to be one of these people who share this opinion, having had experience on both E and G Series desks. I've been very impressed with the response and sound of these various SSL VST simulations, and honestly, in my experience with both, I was knocked-out by how accurate they sounded. I'd be hard-pressed to be able to tell the real SSL gear from the sims).

OTOH... there are other some plugs that, while still sounding good and proving useful, don't quite measure-up to their famous hardware predecessors. I've heard from Pultec fans, most notably our very own Chris ( audiokid ) who happens to have extensive experience on the real Pultec pieces, (both the EQP-1A and the MEQ-5), along with other engineers I know who have also spent quite a bit of time working with these real Pulse-Technologies units, and the general consensus, at least of this writing, is that even the best sims out there for the Pultecs don't quite measure up. This isn't to say that the various sims out there aren't still useful, it's just that to some ( or maybe I should I say many), it's not the same thing... that even the best Pultec VST's out there ( I think that most would likely agree that Softube has come the closest so far) still falls short in the response and sonics of the real Pultec hardware.

Another example would be the Fairchild 660/670 hardware. Considered now to be the "Holy Grail" of gain reduction devices, the Fairchild 660/670 ( The 660 was mono, the 670 was stereo) are famous for their use on hundreds ( if not thousands) of hit recordings over the years, since it's inception in the late 1950's...it's been used on songs by everyone from The Beatles, Stones, and Pink Floyd, to The Beach Boys, Alan Parsons, Deep Purple, Bob Seger, Peter Gabriel, The Police, and other "classic rock" songs, to modern sessions happening right now in 2016... it's become a highly collectable piece, because it reacts and sounds like no other limiter does. With a pretty fast attack time all the way around, but with adjustable release times from as as long as 5 seconds to lightning-fast milliseconds, it initially became popular because it alleviated the "pumping" that other compressors suffered from at the time that it was first introduced... this being just one of the advantages.
(If you're interested, you can check out some of the history of the Fairchild here: http://recording.org/threads/fairchild-670-history-and-how-it-works.58168/)

Now, as a disclaimer, I have no idea whether the various Fairchild 660/670 sims available are truly accurate or not, because my own experience in working with the real thing was so sparse and for such short moments... LOL... and by "experience", I'm broadening that definition to mean that I was in the same room with one a few times... LOL... back in my days as an AE ( assistant engineer). But I have no actual personal work time on the real thing, or at least none to speak of, so I can't say whether the various VST's available are accurate or not. I have to take the word of people whom I respect, who have used actual Fairchilds.
And, in the end, with the 670 commanding prices up into the $40k, $50k, (and even $70,000, yikes!) ranges these days, it's highly doubtful that any of us here will get enough actual time on one to form an opinion one way or the other.

Once again, though, this doesn't necessarily mean that the various plug-in versions available aren't still useful.
If a plug you are using happens to work for what you are doing at the time, and by using it, you're getting a sound or vibe that you like - regardless of the plug you are using, or if it's accurate to the hardware or not - then that's all that really matters., no? ;)

-d.

DonnyThompson, post: 441788, member: 46114 wrote: As Marco mentioned, "ITB" (In The Box) means that - with the exception of getting your initial signal into the DAW - all of your mixing, processing, editing, etc., is all done within the DAW program.
Those who are strictly ITB aren't using any external hardware for processing ( compressors, limiters, EQ's, special FX, etc.)
Now, they may "front load" the input signal with some type of particular analog processing on the way into the DAW( using Outboard, or "OB" gear) such as channel strips, gain reduction, EQ, etc., but once the signal is recorded and resides on a track within the DAW, any further processing of that audio for the final mix is done within the DAW itself.

To take the explanation a bit further...
Accordingly, "OTB" ( Outside The Box) means that external outboard processing gear is actively being used during the mix. This might be an external analog console, or maybe OB compressors, limiters or EQ... This workflow is often referred to as a "hybrid" mixing set up, because it's using both digital and analog processing...

There are those who feel that - while plugs have done a very admirable job of simulating certain outboard hardware pieces, that only the real hardware can truly give you "the" sound that a particular piece is known for.
Some examples of these external pieces would be ( but not limited to) processors like Urei 1176's ( black, silver, and blue stripe models), or Teletronix LA2A's, LA3's, etc., Focusrite Reds, and even particular channel modules from classic consoles like Neve, Harrison, Focusrite, Universal Audio, SSL, Trident, etc.

It really all depends on whom you ask, and their own perception of the accuracy of the digitally simulated models of these pieces, and, it also depends on the particular hardware vs. plug-in version that is being discussed.
For example, there are some who claim that the Waves, T-Racks, Slate and UA SSL collections are as close as you can get to the real thing - without actually using the real thing... and, these opinions are coming from people who actually spent time on real SSL E and G Series desks in the past.
(BTW, On a personal note, I happen to be one of these people who share this opinion, having had experience on both E and G Series desks. I've been very impressed with the response and sound of these various SSL VST simulations, and honestly, in my experience with both, I was knocked-out by how accurate they sounded. I'd be hard-pressed to be able to tell the real SSL gear from the sims).

OTOH... there are other some plugs that, while still sounding good and proving useful, don't quite measure-up to their famous hardware predecessors. I've heard from Pultec fans, most notably our very own Chris ( audiokid ) who happens to have extensive experience on the real Pultec pieces, (both the EQP-1A and the MEQ-5), along with other engineers I know who have also spent quite a bit of time working with these real Pulse-Technologies units, and the general consensus, at least of this writing, is that even the best sims out there for the Pultecs don't quite measure up. This isn't to say that the various sims out there aren't still useful, it's just that to some ( or maybe I should I say many), it's not the same thing... that even the best Pultec VST's out there ( I think that most would likely agree that Softube has come the closest so far) still falls short in the response and sonics of the real Pultec hardware.

Another example would be the Fairchild 660/670 hardware. Considered now to be the "Holy Grail" of gain reduction devices, the Fairchild 660/670 ( The 660 was mono, the 670 was stereo) are famous for their use on hundreds ( if not thousands) of hit recordings over the years, since it's inception in the late 1950's...it's been used on songs by everyone from The Beatles, Stones, and Pink Floyd, to The Beach Boys, Alan Parsons, Deep Purple, Bob Seger, Peter Gabriel, The Police, and other "classic rock" songs, to modern sessions happening right now in 2016... it's become a highly collectable piece, because it reacts and sounds like no other limiter does. With a pretty fast attack time all the way around, but with adjustable release times from as as long as 5 seconds to lightning-fast milliseconds, it initially became popular because it alleviated the "pumping" that other compressors suffered from at the time that it was first introduced... this being just one of the advantages.
(If you're interested, you can check out some of the history of the Fairchild here: http://recording.org/threads/fairchild-670-history-and-how-it-works.58168/)

Now, as a disclaimer, I have no idea whether the various Fairchild 660/670 sims available are truly accurate or not, because my own experience in working with the real thing was so sparse and for such short moments... LOL... and by "experience", I'm broadening that definition to mean that I was in the same room with one a few times... LOL... back in my days as an AE ( assistant engineer). But I have no actual personal work time on the real thing, or at least none to speak of, so I can't say whether the various VST's available are accurate or not. I have to take the word of people whom I respect, who have used actual Fairchilds.
And, in the end, with the 670 commanding prices up into the $40k, $50k, (and even $70,000, yikes!) ranges these days, it's highly doubtful that any of us here will get enough actual time on one to form an opinion one way or the other.

Once again, though, this doesn't necessarily mean that the various plug-in versions available aren't still useful.
If a plug you are using happens to work for what you are doing at the time, and by using it, you're getting a sound or vibe that you like - regardless of the plug you are using, or if it's accurate to the hardware or not - then that's all that really matters., no? ;)

-d.

Thanks for the thorough explanation. I feel like I get it now.

Some of the old posts I've read made me think that it was how a person recorded....meaning you were either an ITB guy, or an OTB guy. But the way I record, I think, is a little of both?

Your explanation cleared it all up for me.

Learning another language (I speak 3) is nothing compared to learning all of you guy's lingo.

I'm studying the two daw thing now.

Aside form the Youtube tutorials on work flow, this place has been the singular greatest asset to learning my way around this stuff. At least better than I currently do. I'm surprised there aren't more people here. Maybe a lot of lurkers, like I was for a while.

Brother Junk, post: 441799, member: 49944 wrote: Aside form the Youtube tutorials on work flow, this place has been the singular greatest asset to learning my way around this stuff. At least better than I currently do. I'm surprised there aren't more people here. Maybe a lot of lurkers, like I was for a while.

We appreciate you saying this.

Unlike other recording forums that are founded mostly on gear and its "related" status, or sales, or support based on just one platform, or... based on how "loud" you can make your mixes... ( insert eye rolling emoticon here) ...

RO is an audio recording and production forum that is, at its core, largely Problem-Based Learning. Our main goal is to help those who seek it, not by fixing things for people, but by helping them to understand how to solve issues themselves.
As these members learn, many of them stay on here and give back by assisting other new members... and the cycle continues.
There's nothing wrong with lurking, I think everyone here - with perhaps the exception of our fearless leader, site owner (and fishing guru), Chris ( audiokid) - has probably done a little bit of of lurking at first, before they eventually became members, when they initially found this forum. Those who eventually become members likely find RO to be a very cool place; helpful, accurate; and eventually, a select few of these new sign-ups become active contributors over time themselves as well; continuing the cycle by teaching others as they had been taught.

For those who end up not becoming members; my suspicion is that they are probably people who are just looking for quick "here's how you fix this" answers; who don't want to take the time to learn how to solve issues on their own.
That's just a guess, though. I could be wrong. I've been wrong before ... once ... uhmmm... in July of 1983. ;)

For me, RO has become my online home, a community of kindred spirits, a sort of virtual reality coffee shop / studio lounge where the members pop in, between sessions and gigs; to say hello, give input and suggestions as needed, help each other to solve problems, or to ask for help themselves... a place where, even though I'm a 35 year veteran in this biz, I continue to learn new things ALL the time.
This place is a treasure to me; its members are some of the most talented, knowledgeable, helpful, unselfish and friendly people I've ever had the privilege to know, and to be a part of.

We've had a few wackadoo's here over time, as any online forum is vulnerable to having ( cough-cough REMY cough-cough LOL); but we've always managed to handle these people, and to keep the focus of RO on track ( no pun intended... no, wait ... pun definitely intended. LOL)
We've got a great group of people here who are friendly, respectful, and who like being here. One of the indicators of this, is that our Moderators have had to do very little actual "moderating", and AFAIC, that's the way it should be.

I am honored to have these guys ( and the occasional girl.. not nearly enough, though... LOL) as my peers and colleagues.

The talent and knowledge bank here is immeasurable. Within our ranks, we have experts in:
Acoustics, Microphones/Mic Techniques, Electronics Repair and Design ( Tube/Solid State), Computers, Online Technology, Music, Arrangement, Theory, Music and Studio Business, Recording and Production ( Digital, Analog, ITB, OTB, and Hybrid), Processing, Midi and VSTi's, Audio Recording History... and that's just to name a few things... ;)

It's my thought that if someone has an audio problem that they are trying to solve, in any of these various fields of study, and they can't get it solved here ?
Then it's almost guaranteed that they're not gonna get it solved anywhere. ;)

FWIW
-donny

DonnyThompson, post: 441800, member: 46114 wrote: Our main goal is to help those who seek it, not by fixing things for people, but by helping them to understand how to solve issues themselves.

Amen to that.

I wish I could contribute more in that endeavor. I should start a thread somewhere that says, "Dog Training Advice" ...so I can pay my part of the rent lol. The current guy is a handsome one, so I'm a post a pic. I think the thread is pretty much done anyway.

Here's the dam

And the sire ;^)

Hey, even Remy had a lot to contribute. That thing about recording an ensemble using speakers to monitor the playback, then recording just the speaker bleed and using that to cancel out the bleed in the actual takes was brilliant.

bouldersound, post: 441802, member: 38959 wrote: Hey, even Remy had a lot to contribute. That thing about recording an ensemble using speakers to monitor the playback, then recording just the speaker bleed and using that to cancel out the bleed in the actual takes was brilliant.

It took me a minute to wrap my mind around it, but if it works, it is brilliant!

"Remy"...no idea what/who that was...that was some interesting reading lol.

Brother Junk, post: 441801, member: 49944 wrote: Amen to that.

I wish I could contribute more in that endeavor. I should start a thread somewhere that says, "Dog Training Advice" ...so I can pay my part of the rent lol. The current guy is a handsome one, so I'm a post a pic. I think the thread is pretty much done anyway.

Here's the dam

And the sire ;^)

Poor Harambe...he died for our memes....

bouldersound, post: 441802, member: 38959 wrote: Hey, even Remy had a lot to contribute.

RemyRad was really something !! She had a thing for sm57 if I remember !! ;)
She suffered brain damage at somepoint and was babling alot.. but she had her good days..

pcrecord, post: 441805, member: 46460 wrote: RemyRad was really something !! She had a thing for sm57 if I remember !! ;)
She suffered brain damage at somepoint and was babling alot.. but she had her good days..

Yes...some of those old posts are very tangential...even savant-like...but there was a method to the madness.

Very idiosyncratic to say the least.