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WARNING!!!! Ignorant question ahead....


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I am in the middle of doing a major upgrade to my studio. Upgrading to DP4, purchasing a new dual 2.5gig G5, and other various items. Anyway, I need to be reminded about monitoring latency when recording. Here comes the "ignorant" question. Is it true that using a hardware mixer is the cure for latency-free monitoring? If so, could someone explain how this works in detail? I know this is very basic but I am going back to understanding the basics because my new upgraded studio will be much more scrutinized than before. thanks!
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You can get sound cards which offer direct monitoring so that the ins are wired directly to the outs for zero latency. Obviously, if you wanted to track with reverb in the cans then you have to use an external reverb unit.

 

Having said that, with a dual G5 you should be able to get very low latency as long as you're prepared to use a crappy reverb plugin. If I use a Waves IR-1 I have to go up to a 256 sample buffer, if I use Nuendo's reverb I can record at much smaller buffer settings.

"That's what the internet is for. Slandering others anonymously." - Banky Edwards.
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Well, I'm not Phil but I'll give it a go.

 

For my old rig, I had to monitor through a hardware mixer to get around my latency problem.

 

here's how I had it configured:

 

say I wanted to overdub a guitar track, I had the guitar mic plugged into the mixer's preamp with the direct out of that channel going into the DAW's input. Then, I had the DAW's stereo output also going into that mixer. This way, I have a mix of the mic and the DAW's stereo mix.

Latency free monitoring.

 

I hope this helped a little...

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You should be able to get latency of around 3 ms or less with the dual G5, which is probably not going to mess up your overdubs.

 

Also many interfaces (including the ones by MOTU), and most that work with ASIO, will let you essentially patch the sound card input to the sound card output, again giving zero latency monitoring. However, note that if you want to record through plug-ins (e.g., reamping guitar) this will NOT work for you.

 

What I strongly suggest is seeing if you can get the latency low enough to be acceptable, and if it is, then live with it. Being able to monitor through plug-ins and such is very helpful.

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As said above, the MOTU gear gives you the option to monitor straight through without latency. I use this feature and get reverb and such from my mixer. Are you using a mixer? If so, no problem. However, I do not have experience with the G5 so you may not have any probs and as Craig said, 3ms is really not a problem. You won`t hear it.

 

Ernest

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Originally posted by ryst:

hello?....phil?

Hello! Sorry i didn't see this before. It looks like you've already been given some great answers to your questions...

 

You should be able to get latency of around 3 ms or less with the dual G5, which is probably not going to mess up your overdubs.

 

I'd go with Craig on that one - he's much more "current" on Macs than I am. But I see no reason why Macs shouldn't be getting good results like that.

 

here's how I had it configured:

 

say I wanted to overdub a guitar track, I had the guitar mic plugged into the mixer's preamp with the direct out of that channel going into the DAW's input. Then, I had the DAW's stereo output also going into that mixer. This way, I have a mix of the mic and the DAW's stereo mix.

Latency free monitoring.

 

FoxTick's got the basics of how to set it up described very well here. :thu:

 

1. Connect the DAW monitor output into mixer,

 

2. Connect the mic or line recording source into mixer and then route it from there into a DAW track for actual recording.

 

The idea is that you want to hear the instrument(s) that are being recorded simultaneously (or as close to it as possible) with the DAW's outputs so that you can play in time with the track as you're doing your overdub.

 

However, if you're running large buffers on your DAW program and / or your audio interface's drivers, there will be a delay - latency - between what is coming into and what is going out of your DAW's audio interface. The larger the buffer, the worse the latency. DAW programs normally run a bit better, with a bit more "horsepower" in terms of processing, if they have a bit of "time" to think about - process - things / information. That's basically because the DAW doesn't have to dedicate as high of a priority to the throughput of the audio signals, freeing up CPU power and task priorities for other things.

 

Is it true that using a hardware mixer is the cure for latency-free monitoring?

 

It's one possible solution. Other ones include audio interfaces with the low latency "mixing" built into them, and DSP based DAW systems, which can get extremely low latency performance figures - near zero. One other way to go is throw horsepower at it... the faster computers get, the less of an issue it seems to be, because there's more horsepower to get multiple things done simultaneously. So getting a faster computer usually doesn't hurt things. :) However, there's still latency from the converters themselves, and if you're an extremely sensitive person / performer / musician, that might be noticeable to you. And small time shifts can definitely change the "feel" of things.

 

But in the real world, anything under 5 ms or so seems acceptable to the majority of musicians I've worked with - myself included; although I have noticed some instruments and vocal parts are more "forgiving" of slightly longer latency amounts than others are. I think of that in this way - sound travels at about 1 foot per millisecond, so you're looking at the analogy of a bass player standing five feet away from the drummer. Ever notice how bass players frequently want to be standing over by the drummer? My theory is that the closer to each other they are, the more simultaneously they're going to be hearing things, and thus the tighter they can play off of, and with each other. And that's their job as the rhythm section - to set that rhythmic / time / groove foundation. But can you play equally as tight if you're standing ten feet apart? It's probably not going to make a huge difference to you.

 

In a product review I just did for EQ, I got the lowest latency I've ever seen on a native DAW system - 1.1 ms. That was with Sonar 3 and a Yamaha 01X / i88X setup running on my Athlon 64 3400 PC system. Which got me thinking about what everyone else was getting on their setups, so I started this thread, where we're discussing that topic. You're welcome to come join in if you'd like. :)

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Here are a few quick tests on my Dual 2 GHz G5 with a Metric Halo 2882 + DSP:

 

16 tracks (2 stereo and 12 mono) all wired to different hardware inputs. One Nuendo reverb with every track sending different levels to it. Project set at 44.1/24.

 

I can't do 64 samples because of the occasional crackle but 128 is smooth. 128 is Input latency 4 ms an output latency is 5.2 ms

 

Similarly, @ 96/24 I get an input latency of 6 ms and output of 6.5

 

These are the conservative figures. You can push it harder and the recorded audio is fine but you'll hear an occasional glitch.

"That's what the internet is for. Slandering others anonymously." - Banky Edwards.
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Hey!

Sorry it took so long for me to respond. Thanks guys for all your help!. I think I am understanding this now. I guess if I want to go this route I would need to have a mixer with really good mic pre's, right? I have a lot of thinking to do. I will be using the original MOTU 828 until I decide to upgrade to the 828mII. I will have to wait and see how the new G5 handles things. Phil, I will come join that thread. Thanks for the invite! One last Question. How do you determine how many samples = how many milliseconds in latency? For instance, how do you know how many milliseconds 64 samples has and so forth? Is there some kind of chart I should have?

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AudioMaverick has posted the equation you need to calculate that over on the thread I linked to... Here's a quote:

 

Here's a raw equation...

- (Buffer Size in samples per second)/(Sample Frequency in Hertz)

. Let's put numbers in it...

.. 44,100 Hz

.. 64 buffers per second

> 64/44,100= approx. 0.00145 seconds, or 1.45mS

So, 96kHz would have a lower input latency with the same number of samples (0.667mS).

 

It's pretty simple actually - once you know how to do it. :)

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Originally posted by ryst:

Hey!

Sorry it took so long for me to respond. Thanks guys for all your help!. I think I am understanding this now. I guess if I want to go this route I would need to have a mixer with really good mic pre's, right?

 

Yes, or you can take another route and use outboard preamps... assuming you have the capability to handle them (via line inputs) on your mixer. I like the preamps in my board, but I also like outboard preamp "flavors" for some things. So what I do is plug a mic into the outboard preamp, and take the line out of the preamp and run it to a line in on my board. These line inputs have A/D (analog to digital) converters that convert that outboard preamp's analog output into digital format - my board is a digital mixer - and then I can digitally route it to my DAW program via a digital connection - ADAT lightpipe, S/PDIF - whatever.

 

If you're using the mic preamps on the board (analog or digital board), you want good sounding mic preamps. If you're doing what I described above (with external / outboard preamps and a digital board), the quality of the onboard mic preamps becomes a moot point; the quality of the board's A/D converters is the important issue. You can, of course, use external converters as part of the whole process too... plug in the external preamp's line output into an external A/D converter, plugging that device's digital output into a digital input on the board, and then routing the signal via a digital output on the mixer directly into the DAW for recording.

 

No matter which method you use, the idea is to get the DAW's monitor out signal into the board, and the mic signal into the board. You monitor those signals from the board while doing the overdubs, while simultaneously sending the mic signal out of the board and into the DAW for recording.

 

Phil, I will come join that thread. Thanks for the invite!

 

Cool - we'll see you there. :wave:

 

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Originally posted by ernest828@aol.com:

...and as Craig said, 3ms is really not a problem. You won`t hear it.

Ernest

But when you're waitin on your wife to get dressed 3 ms can seem like an eternity :D
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