Different ways to use Reverb pre-CPU

[rook takes knight]

Hi,
my PCI Audio Interface has this for I/O:

Two 1/4" Balanced Inputs
Two 1/4" Balanced Outputs
24-bit/192kHz ADAT In/Out (switchable to S/PDIF)
24-bit/96kHz S/PDIF In/Out (switchable to AES/EBU)


I wanted to know how to monitor with Reverb outside of the box.
I looked through the old TweakHeadz Lab website and found that you could use Reverb at:

The Hardware Mixer
The Preamp
The Audio interface

Are there any more ways to do it? I need real world examples on how it works with units that are available right now. I'm a little shaky with some terms such as "auxes, buses, sends and returns", so please bare with me.

[lfm]

If you have a mixer of some sort with aux send/return - might be my choice.
Thinking you have a reverb hardware rack unit then.

Reverb input connected to Aux send on mixer, and having any possible mix dry/wet knob on reverb set to 100% wet.
Out from reverb back to mixer aux return.
So on mic input channel in mixer - you increase the aux send signal to adjust the amount of reverb you want.

Mackie 802VLZ4 is an excellent mixer that also has an aux bus(about $200 or so).
http://mackie.com/products/vlz4-series

Onyx preamps are really nice as well, and handle 60 dB gain and doing just about any mike out there.
Three of those in 802 model, and five line level.

But also talking about audio interface you've got - are you running a daw as well - works just fine that way too.
You have the aux bus with a plugin handled in the same way as external gear. Output from audio interface can have reverb mixed in - but still record dry signal to finalize the amount for a mix later.

[rook takes knight]

But also talking about audio interface you've got - are you running a daw as well - works just fine that way too.
You have the aux bus with a plugin handled in the same way as external gear. Output from audio interface can have reverb mixed in - but still record dry signal to finalize the amount for a mix later.

How does this work? Is it being done out of the box? Would this be a form of direct monitoring?
I only ask because I'd like to operate with 0 latency. I read that can only be achieved in the analog domain.

Right now my setup consists of:

PCI Audio Interface
Preamp (connected to 1 of the 2 balanced inputs)
Monitor Controller (connected to the L/R balanced outputs of the interface)
Studio Monitors

I'd like to add a hardware reverb to that list.
Analog or Digital (what's the difference?)

If I get a hardware mixer, will it replace the preamp and monitor controller or can I somehow use all three?

I'm really interested in learning on how to utilize the 4 different sets of I/O that's on my interface.
Other than using a mixer, what else is available to me?

[BertKoor]

Right now my setup consists of:

PCI Audio Interface
Preamp (connected to 1 of the 2 balanced inputs)
Monitor Controller (connected to the L/R balanced outputs of the interface)
Studio Monitors

If I get a hardware mixer, will it replace the preamp and monitor controller or can I somehow use all three?

There are mixers with USB connectivity, so then it indeed it replaces the audio interface, the preamp and monitor controller. And then there are also mixers with builtin effects (like reverb, delay, etc)

The thing is though, a $200 mixer with USB is very likely of lesser quality than a $200 interface. So you have to do some proper research before jumping on the "add to cart" button. Some mixers can only send the Master Mix Out to USB to be recorded, and that's no good for flexibility.

I'd like to add a hardware reverb to that list.
Analog or Digital (what's the difference?)

It's going to be digital, trust me. There are only two types of analog reverb I know of. One is a physical chamber (occupies your cellar or attic) which is only good if you want to sound like it came out of the old Motown studios. The other is a spring with transducers as found on the Fender Twin Reverb amp, which only is good if you want to sound like Hank B. Marvin of The Shadows.

I'm really interested in learning on how to utilize the 4 different sets of I/O that's on my interface.

First find something that accepts the digital outputs as inputs. Otherwise there's no use for that at all.

[lfm]

But also talking about audio interface you've got - are you running a daw as well - works just fine that way too.
You have the aux bus with a plugin handled in the same way as external gear. Output from audio interface can have reverb mixed in - but still record dry signal to finalize the amount for a mix later.

How does this work? Is it being done out of the box? Would this be a form of direct monitoring?
I only ask because I'd like to operate with 0 latency. I read that can only be achieved in the analog domain.

Many interfaces have options like - either direct monitor button - or knob to mix in level of a direct signal - or support ASIO Direct Monitoring meaning A/D converters data is reversed and sent right out again through D/A converters(this latter then is if daw support it).

Absolut zero - is that absolutely necessary?

#1. reverb itself is a lot of delays to signal and hardly dependent on zero delay
#2. direct signal then - yes - but how short is still ok?
If it's about recording - it's just while singing or playing an acoustic instrument that may cause some tiny phasing. It's not on the recorded material itself.

So how short is ok and not be disturbed by serious delay - like slap delay or tape delay?

PCI interface itself should allow for ASIO buffers 32-64 samples one way - if of a brand like RME. You are in single digit milliseconds here.

There are shades of gray for delays, you know!

So take case - ASIO buffers 64 samples(2ms) - would hardly inhibit playing or singing.

I do with eDrum engine midi input and listen to SuperiorDrummer with that and feels like it's instant as I hit a part of drumkit.

Advantage of accepting something of a delay - is that you need not invest in external hardware - just your interface+daw+reverb plugin. It's usually vocals that is really hard to do all dry - so you want some reverb. Any other instrument is hardly dependent on that while playing.

When inserting a plugin in daw - you can do that on a bus where you send dry vocal signal to - and out comes the mix of those. But only dry vocals is recorded - and you can mix as you please later. You just give vocalist enough so they can sing full heartedly - without feeling like in a closet.

[rook takes knight]

There are mixers with USB connectivity, so then it indeed it replaces the audio interface, the preamp and monitor controller. And then there are also mixers with builtin effects (like reverb, delay, etc)

The thing is though, a $200 mixer with USB is very likely of lesser quality than a $200 interface. So you have to do some proper research before jumping on the "add to cart" button. Some mixers can only send the Master Mix Out to USB to be recorded, and that's no good for flexibility.

Will a USB Mixer allow me to monitor FX while tracking with 0 latency?

First find something that accepts the digital outputs as inputs. Otherwise there's no use for that at all.

I found a hardware reverb with S/PDIF in and out.
https://www.sweetwater.com/store/detail/M350
The thing is it's $200. I'd prefer something much cheaper.
I believe you just need to connect 2 S/PDIF cables from the reverb unit to the interface and enable the soundcard to act as a master clock.

Many interfaces have options like - either direct monitor button - or knob to mix in level of a direct signal - or support ASIO Direct Monitoring meaning A/D converters data is reversed and sent right out again through D/A converters(this latter then is if daw support it).

Correct me if I'm wrong but don't all those applications have some sort of latency?

Absolut zero - is that absolutely necessary?

Professional studios, to my knowledge, track with 0 latency.
As a consumer, I don't see why I can't have what they have. Recording studios have been around since the 1800s. It's not like only a handful of people have the technology or that it's out of reach financially. It's become common and widespread. 0 latency is the standard. Why would I settle for something less.

Also, it may be hard for some of you to believe, but I can still feel and hear the difference between playing the guitar with an amp and playing it through the PC. Even more so when DSP FX is added.

[RichieWitch]

A lot of information to sift through here...

Given the equipment you already have, the lowest cost solution (since you're a guitar player), is buy a reverb pedal and call it a day.

But let's say you don't really like that solution. The next cheapest solution is to buy a small format mixer (without a USB interface inside) and connect the inputs and outputs of your audio interface to the mixer. The mixer will need at least one Aux send (or even better, a send and return). You plug your guitar, preamp, and monitors into the mixer as well. Monitoring directly from the mixer will get you zero latency. However, if you're doing any overdub work, like recording just the guitar and then singing to a playback track, some tiny amount of latency is unavoidable unless you have a LOT of money for top-shelf, high-definition converters (professional studios) or you're recording to tape (1880's). Recording to tape does have its charm.

The turn-around time to the audio interface, then the computer, back through the audio interface, to your headphones, will always introduce some latency (hey, even electrons have a speed limit). Zero latency only works at the consumer level with direct monitoring, that is, before the sound gets to the computer. Everything else is marketing hype.

Correctly adjusting buffer sizes should get the latency down to imperceptible levels, but if you can't live with that, then really your only choice at consumer prices is to record everything all at once through the mixer, direct monitor from the mixer, and use your computer as a high-tech recording device.

Next, about the reverb unit... A reverb unit with SPDIF connections is going to cost more because it must contain its own A/D/A converters, and those cost extra. A reverb/FX unit with analog connections is going to be cheaper. You can get a decent Lexicon unit for around $100 right now. Personally, I would spend $200 to get something that sounds good, but we all have to live within our budgets.

The mixer would make the monitor controller unnecessary, so you could sell that and offset some of your costs.

I own a fairly extensive home studio with lots of hardware and outboard gear, including a couple of mixers that form the central audio bus of my studio. If you want specific connection advice, I can draft something up for you.

[rook takes knight]

Monitoring directly from the mixer will get you zero latency. However, if you're doing any overdub work, like recording just the guitar and then singing to a playback track, some tiny amount of latency is unavoidable unless you have a LOT of money for top-shelf, high-definition converters (professional studios) or you're recording to tape (1880's). Recording to tape does have its charm.

How come there's latency introduced when overdubbing?
like say recording a single track over other tracks in the DAW that have already been laid down?

How would high-definition converters change that?
you mean like an external DAC unit right?

Correctly adjusting buffer sizes should get the latency down to imperceptible levels, but if you can't live with that, then really your only choice at consumer prices is to record everything all at once through the mixer, direct monitor from the mixer, and use your computer as a high-tech recording device.

What do you mean by "all at once"?

Next, about the reverb unit... A reverb unit with SPDIF connections is going to cost more because it must contain its own A/D/A converters, and those cost extra.

About using S/PDIF connections...
I was reading an old SOS article on it and this is what it had to say:

Using a digital connection will avoid passing the audio through two sets of A-D and D-A converters, with their combined delays, but the soundcard buffers and other much smaller internal delays on the soundcard will still need to be compensated for

So I guess it's still not 0 latency. Would there be a difference if you set the external effect as the master clock and vice versa?

The mixer would make the monitor controller unnecessary, so you could sell that and offset some of your costs.

I'm using it to connect 2 sets of speakers to the output of my interface. Can a hardware mixer switch between 2 sets of speakers on the fly?

sorry if this is a lot

[ghettosynth]

You are way overthinking the latency on reverb issue. Put it on a send and you will not be able to perceive the reverb delay at all. I get what you're saying about perceiving the delay of the dry signal that passes through a chain, but as long as the dry/distorted signal is at zero latency, the reverb does not matter as much as you think that it does.

Further, using SPDIF in and out with your sound card and a reverb may or may not put you in clocking hell.

With a zero latency monitor setup there is no latency while overdubbing. The performer sings/plays as he hears the pre-recorded tracks in his monitors. It requires returning the monitored output back to the analog monitor.

Here, watch this, then we can answer more questions. You will save yourself a LOT of headache by getting a useful small mixer for this. It's really worth the effort to setup comfort reverb, compression, and analog monitoring for vocals and guitar.

https://www.youtube.com/watch?v=4l0BsPKpi-Y

[rook takes knight]

I think that may be the golden ticket.

I had a few questions on the 2 different methods that he shows:

Hookup Mixer with Inserts/Direct Outs
http://tinyurl.com/qfr8tzf
Hookup Mixer with Aux Busses
http://tinyurl.com/mwct8zy

With the Insert/Direct Outs method, since there are only 2 inputs on the audio interface, can you only record 2 at a time? What if you want to record 4 tracks at a time?

On the Aux Busses method, the "Mon Send" and "FX Send" go into the inputs of the audio interface.
Does this mean if you use a mixer with more than 2 channels for instruments/mics that you can use all of them at once?

How would you go about adding a hardware reverb to these 2 setups?
If I want to use an external preamp instead of the ones that come with the mixer, where would I need to plug it in?
------------
According to the TweakHeadz Lab page, you can also use the Alt 3-4 Bus that are available in some mixers.

You need a mixer with an ALT 3-4 output (sometimes called a subgroup out, or group out) You can also use direct outs from two channels.

The subgroup (ALT 3-4) connects to the soundcard line in. The soundcard line out connects to any two available line inputs. The can be aux returns, tape ins, or channel line inputs (preferred).

The active monitors go to the CONTROL ROOM OUTS.
NOTE: Though it defies newbie common sense wisdom, you do not have the main outs connect to anything. The Control Room outputs act as the main outs and acts as a volume control for the whole system.

You simply press the ALT 3-4 button (often the "mute" button) to send the channel to the soundcard.

I've googled how to connect 2 sets of speakers but I'm not really getting anything. I looked at a few different mixers online, and most if not all, have a Main Out and a Control Room out. But the thing is, the level control for the Control Room out is also tied with the Headphone out.

Anyway, thanks guys for all the help.

[lfm]

Many interfaces have options like - either direct monitor button - or knob to mix in level of a direct signal - or support ASIO Direct Monitoring meaning A/D converters data is reversed and sent right out again through D/A converters(this latter then is if daw support it).

Correct me if I'm wrong but don't all those applications have some sort of latency?

Yes, and so is analog gear like consoles in professional studios.
The point I tried to make is - there are shades of gray to delays.

Analog gear has something in the range microseconds propagation delay each unit, whatever it is a mixer of each stage of a analog console.

A/D and D/A converters are in the range 10-50 samples. Later technology give the same quality with less delay. Some tend to think a converter is a converter - but quality is very different. Sometimes a brand shines, like RME, doing excellent drivers as well - keeping cpu load low and keeping latency low.

Since professional studios are doing 24/96k - meaning 24-bit 96000 Hz - so doing everything like professional studios do it - that is what you should go for.

50 samples converter delay = 521 microseconds.
10 samples converter delay = 104 microseconds.

Absolut zero - is that absolutely necessary?

Professional studios, to my knowledge, track with 0 latency.
As a consumer, I don't see why I can't have what they have. Recording studios have been around since the 1800s. It's not like only a handful of people have the technology or that it's out of reach financially. It's become common and widespread. 0 latency is the standard. Why would I settle for something less.

Also, it may be hard for some of you to believe, but I can still feel and hear the difference between playing the guitar with an amp and playing it through the PC. Even more so when DSP FX is added.

There a lot of good fx plugins that is zero latency - reverbs too.
Amp sim plugins a known for rather high latency - so playing through will add roundtrip latency quite a bit.

Look at studio work as too separate operations:
a) tracking - keep delays so it does not affect performance in any way
b) mixing - a correctly setup daw has removed all lateny for you from tracking part.

So in the end - latency is not something you live with for the rest of the mix.

Audio interface setup in daw have a basic setup for buffer settings, like for ASIO as one common protocol. So buffer + converter delay is what you live with. A buffer of 64 samples is what any modern computer and good audio interface do these days. Especially having internal cards, and not usb which is more hard to do really low.

And there are different ways daws are affected by these settings. Reaper is probably the worst in this scenario - always adding a full buffer latency.

Let's say you have a clean project from plugins - and add the first plugin with 16 samples delay. Reaper then add a full buffer of 64 samples to already 64 out + 64 in on audio interface to roundtrip - so you get 192 samples adding just this plugin. Reaper compensate final position of audio clip correct - 64+64+16=146 samples - but monitor through get 192 samples - earlier adding a perceived latency while playing/singing.

Other daws add just the extra 16 samples for roundtrip latency. Reaper add full multiples of buffers as needed.

Using the ADAT interfaces and external preamps and converters - this info must be calibrated and told to daw settings if to automatically compensate these delays - at least if you want sample accuracy on final result you mix with. Some daws have settings each track and what delay to use for that track to line up properly. To calibrate this you do analog loopbacks out+in again - and see how things line up using a particular route. ASIO drivers has no clue how long those delays are - you have to do this manually. You can run metronome and record and see if recorded material line up with grid in daw. I keep a little postit note with delays for each input.

I found that even using addon board on RME it was 46 samples different delay compared to regular converters on card - I noticed when doing both lined guitar as well as through amps to different inputs - and got this boxed sound when starting to add some lined guitar with amps stuff - frequency cancellations. But normal input with separate instrument on same converter board I don't bother compensate for this - I hardly play 0.5-1.0 ms accurate anyway, it would be robotic.

Best overall control I find in ProTools - there you have this kind of module in mixer:


You see for each track how much it is delayed by plugins, and how much that track is delayed in the mix - and also field to compensate and set you own delay you know is relevant.

Cubase is pretty good having a field on each track to set delay - but not clue how much roundtrip delay you are in for - if doing a late recording in mix process. You sort of have to know which tracks to mute and listen to a minimum of tracks to do a good performance. Full mix is never needed, really.

Most daws I tested does not have that ability for track delay at all(I only know PC, not Mac). I really wish for everybody to go ProTools route on that.

Anyway, some more nitty gritty on things when recording in daws - there are delays everywhere. So deal with it.

[BertKoor]

I found a hardware reverb with S/PDIF in and out.
https://www.sweetwater.com/store/detail/M350
The thing is it's $200. I'd prefer something much cheaper.

Sorry, but that IS cheap. Imho. I've got a ART DSP Pro unit (also dual stereo engine) and that costed new about $1100, although that was 15 years ago. If cheap is what you want, go for the second-hand market.

A reverb unit with SPDIF connections is going to cost more because it must contain its own A/D/A converters, and those cost extra. A reverb/FX unit with analog connections is going to be cheaper.

Since these units are internally digital already, I disagree with your reasoning. The analog I/O already requires A/D/A converters, S-PDIF should just bypass that. Although it does add complexity: more sockets, potentially sample rate conversions...

[ghettosynth]

I think that may be the golden ticket.

I had a few questions on the 2 different methods that he shows:

Hookup Mixer with Inserts/Direct Outs
http://tinyurl.com/qfr8tzf
Hookup Mixer with Aux Busses
http://tinyurl.com/mwct8zy

With the Insert/Direct Outs method, since there are only 2 inputs on the audio interface, can you only record 2 at a time? What if you want to record 4 tracks at a time?

On the Aux Busses method, the "Mon Send" and "FX Send" go into the inputs of the audio interface.
Does this mean if you use a mixer with more than 2 channels for instruments/mics that you can use all of them at once?

How would you go about adding a hardware reverb to these 2 setups?
If I want to use an external preamp instead of the ones that come with the mixer, where would I need to plug it in?
------------
According to the TweakHeadz Lab page, you can also use the Alt 3-4 Bus that are available in some mixers.

You need a mixer with an ALT 3-4 output (sometimes called a subgroup out, or group out) You can also use direct outs from two channels.

The subgroup (ALT 3-4) connects to the soundcard line in. The soundcard line out connects to any two available line inputs. The can be aux returns, tape ins, or channel line inputs (preferred).

The active monitors go to the CONTROL ROOM OUTS.
NOTE: Though it defies newbie common sense wisdom, you do not have the main outs connect to anything. The Control Room outputs act as the main outs and acts as a volume control for the whole system.

You simply press the ALT 3-4 button (often the "mute" button) to send the channel to the soundcard.

I've googled how to connect 2 sets of speakers but I'm not really getting anything. I looked at a few different mixers online, and most if not all, have a Main Out and a Control Room out. But the thing is, the level control for the Control Room out is also tied with the Headphone out.

Anyway, thanks guys for all the help.

You have to decide how much effort and expense you want to put into this. You are correct in that with only two inputs on your interface that you can only record two tracks at a time with the insert method. In reality though, if you use the aux method you can still only record two tracks at a time. You can record more performers, but you will have to premix them in the analog domain and you will not be able to separate them (practically speaking) later.

So, the bottom line here is rather simple. If you need to track more unique sources at the same time you MUST have more inputs on your interface. There's no magic sauce here, that's how it is. For each distinct source that you want to track separately you either need a channel with an insert of a separate aux send as well as a distinct analog input channel on your interface.

I use both methods insert or aux, depending on what I'm doing. I have a Mackie DFX12 that has some built in effects that is the core of my direct monitoring setup. It also is my main audio mixer for my modular. So if I'm mixing a bunch of signals from the modular, I might not care about tracking them independently so I just use the aux busses for that. There are only two aux busses on that mixer, but I only record myself or an occasional guest. Generally, unless I'm mixing analog synths and I want to just capture them all together, I use the channel inserts for tracking. If I were tracking drums or an entire band, I would have to use a much more serious mixer.

The second component that I think is just something that you have to bite the bullet on is a patch bay. All inserts, sends, and returns go to a patch bay with my rack mount effects unit so that my configuration can be flexible. I also route the inputs from my interface here. I don't route the outputs, I'll get to monitoring in a minute.

As far as effects for tracking. My DFX has a crap reverb built in, but, it gets the job done for "comfort verb", I can patch in a better reverb if I'm feeling fussy, but usually it's fine. Although I think that this is clear to you, I'll say it anyway, you don't need to record this verb, in fact you don't want to. That's the value of using the aux sends or the inserts, you want to record a dry unprocessed signal. The only thing being processed is the monitoring and the only thing that reverb gets used for is for monitoring. When I'm recording vocals I feed the direct out into both the input channel on my interface and a compressor. Then I return the compressor via the channel insert return. This gives me a compressed and EQed vocal with reverb and or delay in the monitor but a dry uncompressed signal goes to my DAW.

Ok, monitoring. I don't monitor through my mixer. I spent a few hundred dollars on a presonus monitor station and it was worth every penny to me. It's not necessary, but I find it simpler to separate my "monitor speaker selection" from my "tracking monitoring." The Mackie tape outs route to one of the inputs on my monitor station so I can choose to listen to it on any set of speakers. If I wasn't doing that, I would just route my DAW output back to the "tape return" of the mixer. You don't need a fancy monitoring section. Using the tap input from your daw and the tape or main output to your monitors will get the job done for basic tracking. What I don't like about this is that the mixer would be in the monitoring path all the time. The monitor station is much quieter than my Mackie mixer. Of course, you can always spend MUCH more for a mixer, or, you can just ignore the minor amount of extra noise.

I use headphones when I'm tracking vocals, of course. But I sometimes use speakers when I'm tracking non-miced sources. This is where the monitor station comes in handy. I NEVER use my studio monitors as tracking monitors, they simply aren't designed for the dynamic range of live / uncompressed signals at volume. I have a small PA in my studio that I use for tracking so I can get as loud as I want.

Hope this helps. Let me know if you have any specific questions about how I have it setup.

[rook takes knight]

Yes, and so is analog gear like consoles in professional studios.
The point I tried to make is - there are shades of gray to delays.

Analog gear has something in the range microseconds propagation delay each unit, whatever it is a mixer of each stage of a analog console.

Anyway, some more nitty gritty on things when recording in daws - there are delays everywhere. So deal with it.

I'm fine with that. Just as long as the technology that I'm paying for and using is up to par with what has been and currently possible.

For each distinct source that you want to track separately you either need a channel with an insert of a separate aux send as well as a distinct analog input channel on your interface.

Sorry, but I've never used a mixer before so I'm learning as I go.
I'm looking at a picture of the DFX12 for reference .
What do you mean by "an insert of a seperate aux send"?

It also is my main audio mixer for my modular. So if I'm mixing a bunch of signals from the modular, I might not care about tracking them independently so I just use the aux busses for that.

By modular, do you mean modular synth? Where on the DFX12 are the 2 aux busses?

When I'm recording vocals I feed the direct out into both the input channel on my interface and a compressor. Then I return the compressor via the channel insert return.

Which is the direct out? Does it have 2 outs to it(1 going into the interface, the other going into the compressor)?

The Mackie tape outs route to one of the inputs on my monitor station so I can choose to listen to it on any set of speakers.

So the L/R tape out from the mixer goes into the aux in L/R of the Monitor Station?

So if I'm seeing this right, you have:
Audio Interface > Mixer > Monitor Station > Monitors
and the patch bay somewhere in the picture. Does it matter that the signal is being sent through more than a few units?

The second component that I think is just something that you have to bite the bullet on is a patch bay. All inserts, sends, and returns go to a patch bay with my rack mount effects unit so that my configuration can be flexible. I also route the inputs from my interface here. I don't route the outputs

I'm not entirely sure on how a patch bay works. Do you have the mixer go into the patch bay then have that go into the interface?

If I wasn't doing that, I would just route my DAW output back to the "tape return" of the mixer.

Where on the DFX12 is the tape return?

Using the tap input from your daw and the tape or main output to your monitors will get the job done for basic tracking.

So
DAW output > tape input
Mixer tape out/main out > monitors ?

1 last thing, can all mixers bypass it's built in pre-amps?
I read that you just need to connect the external preamp to the Line or Insert on the mixer.

[BertKoor]

I'm looking at a picture of the DFX12 for reference .
What do you mean by "an insert of a seperate aux send"?

There are a couple of ways to tap off one single channel for recording purposes:

1. Make sure it is the only channel with sound, then you can record from Main Mix. No good if you are playing along with some other tracks.
2. Make sure it is the only channel on a specific sub bus. A proper mixer doesn't only have multiple inputs but also multiple outputs. There's usually two buses (which carry the sum of all channels switched to that bus) : the Main Mix bus and an alternate bus Alt-34. You could for example put all tracks on the Main bus and put the instrument you are recording on the Alt34 bus. It has separate outputs you can use for recording.
Note: the DFX12 only has a Main bus, no Alt34. Plenty other mixers of similar size do have "alt" buses though.
3. Each channel has a "send" knob to Aux buses. These also go to dedicated outputs, usually to "send" it to external (or saometimes internal) effects, but can also be used for recording. Since an output is just an output, and the name label does not dictate where you send it to.
4. Bigger mixers have DirectOut jacks for each channel, so you can hook up a multitrack recorder directly and record all the incoming channels in one go.
5. Cannel Insert jacks can be used for the same purpose. On the DFX12 you see these at the top, just below the Line Inputs for each channel. These are TRS jacks that carry both Send and Return signals, to be used with Y-cables for an external effect (or chain of effects) placed after the mic preamp & gain trim stage. If you insert a guitar cable half way into an insert, it can be used to tap off that channel. For recording for instance...
6. Also a hack: put the channel on Solo or PFL (Pre Fader Listening) and record whatever comes out of the ControlRoom outputs, while Main Mix is just that: the whole mix (with monitors playing that)

Where on the DFX12 are the 2 aux busses?

On each channel there's the red pots to set the level to send to AUX1 & AUX2. Above the red faders are extra pots to set the total level to send. On the top right just left of the Main outs are the Aux Send outputs.

Where on the DFX12 is the tape return?

Those are the RCA inputs labelled "CD/TAPE IN", connected to the white fader inbetween the red & blue faders.

Can all mixers bypass it's built in pre-amps?

Sure. Just use the 1/4" TRS inputs with Line level instead of sticking it into the XLR which is for mic level. But in my experience there's no need to use an external preamp anymore if your mixer is of a certain quality level. Bad preamps on a mixer usually means the whole mixer is to be avoided. Because the line inputs go through the exact same circuitry, but only with less gain than microphone signals.