Direct monitoring and latency

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JonFoo

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I'm confused about direct monitoring and latenzy...

Direct monitoring means that when overdubbing you will hear what you play without delay, because the signal you hear in your headphones isn't digitilized, right? But what is recorded is of course digitilized, and thus delayed(?) How do you solve that? By moving the overdubbed track a few milisecs "back"?

I'm using Tracktion and is has a function which auto-detects the delay (by looping the input into the output), and (I belive) compensates for latency by moving the recorded track back the necessary milisecs. This solves part of the latency problem. However if I don't use direct monitoring, what I play will be delayed in the monitor, and things are screwed up anyway..?
 
Yep hardware or direct monitoring means you hear the signal being recorded 'on the way in' to the computer, so there's no delay between playing a chord or note and hearing it in the headphones.

When you overdub, you listen for what's been recorded, then play along. The only latency involved when monitoring direct is the time it takes for the recorded signal to fill the buffers and travel through the D/A convertor into your ear, plus the time it takes for your guitar chord or whatever to pass through the A/D convertor, fill the buffers & hit the hard disk. Depending on how many samples you've specified for your buffers to hold, this will only be a few milliseconds. If that's too long you can manually drag the recorded audio back in your daw. It really isn't a big deal.

It can become a big deal if you're recording a soft synth that's streaming samples from the hard disk, or you're using Input Monitoring where the recorded signal passes through the daw and any effects you've applied before you hear it in the headphones
 
Bulls Hit said:
The only latency involved when monitoring direct is the time it takes for the recorded signal to fill the buffers and travel through the D/A convertor into your ear, plus the time it takes for your guitar chord or whatever to pass through the A/D convertor, fill the buffers & hit the hard disk. Depending on how many samples you've specified for your buffers to hold, this will only be a few milliseconds.

Isn't what you describe here non-direct monitoring?? (What I play is digitalized before I hear it and thus delayed.)

The buffer-size can be set manually and will typically be what, 5-50ms? Is this so little you can't hear the difference?
 
JonFoo said:
Isn't what you describe here non-direct monitoring?? (What I play is digitalized before I hear it and thus delayed.)

The buffer-size can be set manually and will typically be what, 5-50ms? Is this so little you can't hear the difference?

No, direct monitoring means direct: what you play goes straight from the inputs to the headphones. There's no latency there. The latency arises from the time taken to fill the record & playback buffers.

When you press record, the daw starts recording immediately. But the there's no audio to record yet because the record buffers are still empty. So the sequence of events is
Press record
Playback buffer fills with audio
Audio passed through DA convertor to your headphones
You play your guitar along with the headphone audio (no delay here - this is direct monitoring)
Guitar audio passed through AD into Record buffers
Record buffers fill with guitar audio
Record buffers written to hard disk

The time taken to go through these steps depends on the number & size of the playback and record buffers. Anything around 10ms is acceptable given that you'll play either ahead or behing the beat anyway. Once you get below 5ms you're dependant on your daw to be able to refresh the buffers fast enough to keep up without audible glitches
 
Ah! Thanks.. I think I've got it now.
The problem with latency and delays is solved by using a fast DAW which can handle a small buffersize - without noticable delay. With my new fast Macbook I will try to reduce the buffersize (and hope it can handle it..), use direct monitoring, and hopefully won't be bothered with latency again!

Btw, how will I experience problems if the buffer size is too small? Lagging, etc..?
 
You'll hear pops & glitches in your recorded audio.

Don't worry, you'll recognise it if it happens
 
Anything up to about 20 ms with ASIO monitoring shouldn't really be noticeable.. from there you'll notice sluggishness, for example when using a guitar plugin and playing a fast riff.
 
But what is recorded is of course digitilized, and thus delayed(?) How do you solve that? By moving the overdubbed track a few milisecs "back"?

The software knows how big the buffer is, so it compensates automatically on playback. So when you listen back to tracks that are all overdubbed, and if you used direct monitoring, you should have no problem at all...
 
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