I'm confused: Zero Latency Monitoring

  • Thread starter Thread starter Dolemite
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Dolemite

Dolemite

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I'm a little confused by the issues of latency, specifically in regard to monitoring. Ideally I want a system that will allow me to multitrack everything but vocals, then be able to record my vocals separately while monitoring them mixed with the recorded tracks. As I understand it, with a standard 4 in 4 out card like the Delta 44 or the Echos this can be handled by the card's internal mixer (correct me if I'm wrong), but can it be done with zero latency? I.E. no delay between what I sing and what I hear in the headphones?

I just got a sweet deal on a used Midiman 2044, but I was previously looking at the M Audio Delta Studio and the Seasound SoloEX, which both handle this issue externally. I'd really like to know if the 2044 is really a bargain and can do what I need it to or if I need one of these other systems to do my version of "zero latency monitoring."

Thanks
 
You'll be able to do "zero latence monitoring" with that card. As long as any card is full-duplex it shoudl be able to do ti with zero latency.

Jake
 
Ah but the question is, what are you monitoring?

I have the same question.

I ran a line out of my Delta44 back into an input and played/recorded a high hat click. The recorded click was perfectly on time (visually) with the original. Therefore I concluded that there was zero latency when recording.

HOWEVER, what am I monitoring? I use a 2700 sample ASIO buffer which is like a 40ms delay at 48Khz. That would be too much latency if it was audible. In fact I know it's too much latency after trying the live input feature of n-Track.

As I see it there are a couple things that could be happening.

- The signal could simply be routed via the internal mixer on the soundcard to the output without ever touching an A/D converter.

- The signal I'm hearing was indeed digitized, but is being routed via the internal mixer to an output before it's *buffered*.

Sorry to bust into your thread Dolemite, but maybe my question here will help elaborate on things a bit.

Slackmaster 2000
 
I think there's some confusion over the term "latency" here. Even the cheapo soundcards allow you to monitor a .wav file while you record a new .wav file without timing problems. "Latency" refers to the response time (time between changing a control and hearing the change) when using software effects, from simple volume and pan adjustments to more complex stuff like EQ, compression and reverb.
 
Thanks for the responses guys.

I guess I was using the term "latency" incorrectly. Now that I understand it, the whole thing makes much more sense. Anyway, its good to hear that my new card will do what I want it to.

Hey Slack,

Feel free to commandeer my thread at any time, you bastard! (j/k)

Actually I have an idea of what might be going on there. From what I've read, all the current sequencing/recording programs have a setting that compensates for the timing differences that result from latency. Whatever program you're using probably just detects the sample buffer your drivers are set to and compensates for the delay.
 
That's pretty much true doc, but consider this:

Let's say you're going to add a track to an existing mix. You're going to be monitoring that mix while recording, say, a guitar.

You hit record. The software buffers up 40ms of audio and begins to play. You then pluck your guitar and the incoming signal is placed into a recording buffer of whatever size.

Now my issue isn't whether the recorded track will line up with the playback track, for the software can easily compensate.

My issue is when do you hear your guitar? You certianly don't hear it after it's been committed, or you'd be hearing it 40ms in the past....just like using live input processing.

So the signal must be routed from the hardware input directly to the hardware output immediately. My question is...does the signal go through the converters? In this case what you hear while recording is most certainly what you will hear when you play it back.

However, with cheaper soundcards, I think the signal is routed pre-converter, because what you hear when monitoring your live signal is NOT what you hear when you play it back. This used to drive me nuts with my old AudioPCI card...I had to record and play back repeatedly to find the right sound.

According to this article at RME, their products route the signal after the converters. I'm not sure what the delta does. http://www.rme-audio.com/english/techinfo/lola/lomo.htm

Slackmaster 2000
 
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