The only way to get true zero latency monitoring is not to go through the DAW.

If you were tracking through the board this would be easy to achieve: send the monitors from the channel input paths rather than from the returns from the DAW.

If you have external preamps you wish to record through you could y-split their outputs to feed both your DAW inputs and your monitor board at the same time. Or you could use duplicate outs if they are provided, eg: feed your octopre into the DAW via its digital lightpipe outs while running its analog outs into the desk. That way the desk is not part of your recording chain at all and is only used for monitoring.

Where you might run into difficulties is if your preamps are built into your interface. In this case you might be able to use the inserts to tap off analog monitor signals before they hit the ADC.

Hi IIRs. Thanks for such a quick reply. OK, yeah I don't want to track through the pre-amps on the board. I want to use the saffire pro as an interface, have the octopre run through the saffire into the daw through an A/d converter, and use the pre-amps on both the saffire and the octo-pre. It would be for an overdub situiation so the saffire would be use to get 8 out of 16 guides down and act as an interface, everything else overdubbed through an octopre.

So I would be looking at scenario two you described, recording through external preamps and the desk is not really part of the recording chain, but using it for monitoring.

Sorry to be such a dummy on this as you have kindly covered all eventualities here, but when I use a y-splitter cable (presumably into the insert points on the desk), does the fact that the tip ring sleeve setup mean that the DAW inputs and the desk are fed exactly at the same time, thereby achieveing no latency? If so, does this mean that I still can't go through the DAW to hear signal as its being printed to disk? (the reason I am asking is to not have the ability to monitor through the software to hear it actually going down to the hard disk worries me a little, not to be able to hear any little giltches and only hearing the playbacki n the presence of the band where it could throw up problems). Kind of like being able to hear back from tape in a mix B scenario whilst tracking to tape. Or can i be confident that with a high buffer sizes what I hear going in wil be what I hear in playback? Sorry for so many questions,thanks again for your help it's hugely appreciated.

Obviously you are still a little confused!

The object is to split the signal out of the pre-amps so that they go into the DAW for recording while also going into the desk for monitoring. The best way to approach that depends on the pre-amp outputs: if each pre-amp had just a single XLR output you could make (or buy) y-split cables so that each preamp could be plugged into a line input on the audio interface and also a line input on the desk. We don't need to use inserts for this method, and the y-split cable is not wired the same as an unbalanced insert type y-cable.

I'm pretty sure the octo-pres have both digital and analog outputs however, so in this case you would not need any y-split cables: assuming you are already using the digital outs to feed the Saffire interface, you could simply plug the unused analog outputs into your desk's line inputs (not inserts!) and you're good to go.

The problem you may find is with the Saffire's built-in preamps. I don't know the units so I don't know if there is any way to tap off the analog signal after the preamp... this was why I mentioned inserts: if the Saffire provides inserts for its own preamps (I don't know, does it?) then these may provide a solution.

As far as your monitoring is concerned... assuming you have a seperate control room (or the talent is on headphones) there is no reason for you to listen to the same mix as the band. The simplest way to achieve that would be to patch your studio monitors directly from your audio interface. That way the band can listen to latency-free heaphones mixes via the desk while you monitor the results back from the DAW. A more elegant solution would be to patch your monitors from the control room outs of the desk, and to bring the DAW outs back into the desk via 2-track returns or spare channels, depending on the specifics of the desk you are using. That would allow you to monitor the musician's foldback sends while setting up their mixes, then quickly and easily switch to monitoring the DAW returns once they are happy.

Using the Sapphire as an interface means that the TRS line outs on the back could be used to send signal to the monitor board. They just have to be routed in the ASIO GUI 1 for 1.

Ok. Firstly thanks again for all this great help. I quite clearly needed this clarified! Yeah, I have a separate control room. I think the penny has dropped here.... but let's see....

So I'm splitting the signal before it reaches the DAW and taking that out to the desk, where I will make the headphone mixes from. This means that the performers don't have to hear any latency because they are hearing the signal before it goes into the DAW.

As I am listening in the control room, I can monitor directly from the audio interface, allowing me to hear what is going on as the audio goes to disk, because there is no reason for me to hear the track in sync with the performer,(given that they are hearing a desk based monitor mix through headphones.) I can also use external reverb units patched from the desk to send to singers, therefore bypassing the need for the performer to hear any plug-ins inside the DAW at all, asides from the playback which will route out into the desk. So the DAW becomes a 'tape' I can monitor rather than a monitoring system in itself. Is this roughly correct?

Also in this scenario (which is great thank you!), I imagine I can't have a perfromer in the control room with me to lay down parts , ( I sometimes do this for communication with the musician, miking their amp in the live room and just running the lead through from amp to their instrument so they are playing in the control room but through a miked amp if you see what I mean. I take it with this method I just have to accept the performers will always be in the live room monitoring through headphones, if I want to hear what is going on as the signal goes into the software, short of monitoring the signal myself through headphones on the back of the audio interface. (If so, it's hardly a big sacrifice for the benefits involved.)

Yes I think you've got it!

Obviously if the performer is in the control room listening to the same monitors as you then one of you will have to compromise. Either you will need to listen to his analog foldback mix, or he will have to listen to your delayed DAW returns... if you go with the first option it might be wise to make a short test recording first, "to set levels", so you can reassure yourself that everything is working as it should be before you start cutting takes. If that sounds ok you are probably safe to go ahead and record your session without monitoring the DAW returns: chances are that in this scenario the talent will be overdubbing parts on top of existing backing tracks, so as long as these are still playing back glitch-free at the correct speed all is probably well!

One last little point: when setting up monitor mixes you will indeed need external hardware for stuff like compression (if that is needed) but reverbs or delays will probably still work fine with a bit of latency as its just like increasing the pre-delay by a few ms. So if you are short of hardware reverbs there is no reason not to use a plug-in patched to a spare pair of DAW outs and thence to your monitor desk.

Thanks again for your help clearing that up for me. Also thanks for the tips on the using reverb with latency, i can see what you mean there. I can see what you are saying now also about the concern of using the preamps on the saffire (i think) i.e. because this is the interface, to monitor out of that effectively means you are monitoring out of the DAW. (Although you have both posted possible methods around that- thanks!) I

think you are saying the octo-pre is different becasue it is a channel strip, so by using the outputs here to the line ins on the board, it can be split before it reaches the DAW, much more easily than if having to use the preamps on the saffire. I suppose in the ideal world I would have two octo-pre running into the saffire and just usie the saffire soley as in interface, allowing me to split 16 channels immediately at all times.

Dundee1 wrote: I can see what you are saying now also about the concern of using the preamps on the saffire (i think) i.e. because this is the interface, to monitor out of that effectively means you are monitoring out of the DAW. (Although you have both posted possible methods around that- thanks!)

Yes, almost. If I understood TheJackAttack correctly he seems to be saying that the Saffire offers internal routing facilities that would allow the mic inputs to be routed directly to the corresponding line outputs as well as appearing in the DAW for recording (a fairly common feature for this type of interface). If so this means you are not actually going through the DAW, but you are still going through the AD and DA converters, with some processing in between. ie: the latency will be less than you would get if monitoring through the DAW and would not be affected by your buffer size settings, but it would not be zero.

This might be a workable solution as the latency should be pretty small, but you might need to be careful in multi-mic situations as it will be enough to cause phase problems, eg; for a multi mic'ed drum kit. If you made sure that all the drum mics were plugged into the octo-pre and used the saffire for the rest of the band you would probably be ok. If you needed to squeeze in a bottom snare mic and had used all 8 ins on the octo-pre you would be ok plugging it into the saffire so long as you didn't send any of it to the drummers cans...

Another possible solution would be to y-split the mics plugged into the saffire's mic inputs so you can also plug them into mic inputs on the monitor desk. This method has the disadvantage that the mics will be seeing a lower impedance and might not sound quite as good, but if its just for a guide vocal, or for a hi-hat mic that you aren't even sure you will use, it should be a workable solution. (I y-split mics all the time in a live context and it never seems to cause problems. Condenser mics probably won't sound any different with a y-split, but make a decision as to which preamp will send phantom power and stick to it: probably not a good idea to power a mic twice! Dynamic mics might change in sound a little, but probably not enough to be a problem for most sources. Ribbon mics probably shouldn't be y-split, unless they are one of the modern active types, but it probably won't hurt to try. Just remember to check BOTH preamps have phantom turned off before you plug it in...)

A possible step up from y-splits would be active splits. These would allow you to do the same thing but without changing the impedance that the microphones sees, and would also protect the mics from any interference on the monitor signal path. Good quality active splits can be expensive, but you might get away with cheap models for your scenario: just make sure the isolated outputs are sent to the monitor board while the DAW inputs get the direct feeds. Hopefully that way any degradation from cheap components will only affect the foldback mixes.

The latency of AD DA in the Sapphire would not be audible if routed directly internally. That's why we can use DSP boxes and outboard eq's and comps etc in live sound. If we want to pick nits, there is latency in the wiring of an analog board too but it is also negligible.

I am quite willing to be wrong on this but that is my experience.

Also, the pre's in the Sapphire Pro are likely to better than many small format mixers though certainly not all.

TheJackAttack wrote: The latency of AD DA in the Sapphire would not be audible if routed directly internally. That's why we can use DSP boxes and outboard eq's and comps etc in live sound.

A couple of ms of latency would not be audible in isolation, but if it is mixed with another correlated but undelayed signal the difference will be audible as comb filtering. That's why I can't use a digital comp such as an XTA C2 for parallel compression in a live context, and why I have to be very careful when setting up parallel compression on digital FoH consoles.

TheJackAttack wrote:
If we want to pick nits, there is latency in the wiring of an analog board too but it is also negligible.

I agree that the latency of an analog board is negligable! The latency of a typical set of AD/DA converters will be many times greater than that. Remember: even just 1ms of delay will cause comb filtering artifacts as low as 500Hz when mixed with an undelayed signal. 0.1 ms of delay will still cause a major cancellation at 5KHz. 0.05ms would put the first dip up at 10KHz... you might get away with that, but 1/20th of a ms is not very much at all: just over 2 samples at 44.1KHz

Feel free to correct my maths of course: I skipped too many maths lessons as a youth. (Does anyone feel like calculating the latency of a typical analog desk? :wink: )

corrected my own maths :roll:

TheJackAttack wrote:
Also, the pre's in the Sapphire Pro are likely to better than many small format mixers though certainly not all.

He doesn't actually specify its a small format mixer. Maybe he's thinking of purchasing a vintage Neve? ;-)

Anyway, I was only suggesting using the mixer pres for the monitor mixes: the recordings would still be made from the Saffire's preamps.

OK. I concede the point. I would suspect the latency on the Sapphire for straight through to be less than 1ms but then again I use the FF and I'm not in a position to measure it. 64 samples buffer is about 1.4ms and that's in and out of a pc.

True about the mixer. Wish I had one of Remy's Neve consoles.

Digital PA boards hadn't made the scene except for the biggest guys when I was running live sound regularly. I got my first digital studio board about six months before leaving the Corps.

If he wants to split prior to the Sapphire then an ART S-8 would seem to fit the ticket.

Hi both. The desk I will be using in this scenario is a soundcraft ghost 24:8

If the latency on the saffire is 1ms or so straight through, that is certainly good news/aceptable, by going straight through. So you mention earlier if using the adat option to connect the octopre to the saffire I may be able to use just the analog outs to the line ins in the desk and you mention this may negate the need for a splitter, presumably just using a loom with the approriate input/output connections. Would this defintely work? (it would be great to avoid the immedaite purchase of a splitter if at all possible.)

Also, (this is very base question)but if I started running myself into real problems, I take it the optical out is all in need to send the signal from the octopre to ADAT.....( can I even handle going back to a tape based solution???) For example if I decide to run the octopre straight to ADAT for drums and simply use the pre's on the desk to capture the guides, then record perhaps vocals into the DAW where I need more tracks than 16. Obviously I realsie this negates the saffire from the initial drum tracks. (Sorry to introduce this scenario I thought I might explore this eventuality whilst you are giving me such great help)

What should I look for in a splitter in this situation, on top of the one mentioned, are there any other ones which you would recommend that would work for this setup?

Thanks in advance.

I'm pretty sure you won't need splits for the octopre, if its the same as the ones I have used and not some cut-down "LE" version or something? If it has digital and analog outputs then it will work. I have used 3 octopres to feed a pair of HD24s for live recording in the past: I had the optical digital outputs feeding the first HD24 and the analog outs feeding the second as a backup. Worked perfectly.

As far as ADAT is concerned: the optical light-pipe format you are using was originally developed for ADAT machines, so I can't see any problems there. Don't know why you would want to go back to using those things though!

IIRs wrote:
As far as ADAT is concerned: the optical light-pipe format you are using was originally developed for ADAT machines, so I can't see any problems there. Don't know why you would want to go back to using those things though!

No, quite!! It's an unbearable thought. I am just thinking of possibilities if I had to change tact due to unforseen problems with a DAW during a session.

The octopre would be the full version. OK, so the feed should work just fine from what you have said, so wil try that. And the splitter mentioned would do the job if I decide to go down that route later on? So I just need one if I decided to go down the splitter route?

The ART S8 looks ideal. 8 channels, so yes you would just need one.

That way your monitor board would have 8 channels of line inputs from the octo-pre plus 8 channels of mic inputs from the isolated outputs of the S8.

Alternatively you could start off with simple y-splits, then gradually substitute single or dual channel active splitters when your budget permits.

That's great! Thanks for all your excellent help. It's genuinely very much appreciated. I'll let you know how it all goes!