Jump to content

how do you deal with latency?


Recommended Posts

  • Members

I have a question about latency when overdub. with a DAW. I was planning on purchasing a good mic preamp like the great river mp2 or vintech 1272 as well as an Art Pro compressor, maybe an TC M-one-xl verb unit. I was informed on another forum (Akai dps24) that I would experience latency...is that true? How does one deal with latency? It seems that latency would make it near impossible to overdub leads, vocals...just wondering.

Link to comment
Share on other sites

  • Members

With any DAW there is latency - A/D/A converters, software processing, writing to disk, reading from disk - it's all numbers and it takes time. But with good hardware and software, this time can be only a few milliseconds. That's less time that it takes the sound to travel from your speaker to your ears in many cases.

 

In practice, it doesn't really matter how much latency, as long as your DAW compensates for it. For example - let's say you had a 1 second delay. You record your first track. When you play back, you would have to wait 1 seconds, but when the audio started you wouldn't really know or care. When you record your next overdub, as long as the DAW 'knew' that the incoming audio was 1 second late, and time-stamped it accordingly, it would be in perfect sync when you play them back.

 

So in practice - latency isn't a problem. Although a few ms delay isn't a big problem - sometimes it's easier on your PC to relax this out a bit to avoid overloads and clicks & pops. Beyond about 20ms, this delay can be annoying.

 

I recommend zero-latency monitoring. This simply means, when tracking you listen to the input of your soundcard, not the output. This can be done with preamps that have two outputs - send one to the A/D, the other to a monitoring mixer which combines this with the PC soundcard output. Many audio interfaces come with this zero latency mixing option installed - so you don't even need a mixer or headphone amp.

 

When tracking virtual instruments - the latency is unavoidable, but 20ms or so is ok. Some plugins introduce greater latencies - and the solution is simply save them for mix time - don't track with them on.

Link to comment
Share on other sites

  • Members

What he said. :cool:

 

BTW, here's a good tip for tracking virtual instruments...

 

1. Render a mix in the daw. Render don't play it and record it back in... rendering will give you sample accurate placing.

 

2. Shut all the other audio tracks off. You may have to disable them, not just mute them, to keep the disk from reading them anyway. Most daws still read tracks that are muted, they just don't play them back while the mute button is on. This is so when you unmute you hear it immediately. You might want to create a script that "disables" all selected tracks.

 

3. ASIO (I hope)? Turn your buffers way down to get very low low latency. 10-15ms is good for playing live.

 

4. Play the virtual instrument part in realtime using the rendered mix as a guide. It's a lot less strain on the system with very low buffers to play a single stereo track with no effects than 24 mono tracks with plugins going. It won't (shouldn't) pop and click.

 

5. When you're done raise the buffers and get back to work with the multiple tracks. Most daws should allows changing buffer sizes without rebooting.

 

I'll raise and lower the buffers in SX mulitple times during a session depending on what I need to do. Takes about 15 seconds. This method works well for tracking vocals and monitoring through the daw if you can get your latency down to about 3-5ms.

 

Other Daw Tips?

 

I also recommend testing your daw for near sample accuracy. What I mean by that is making sure that what you record goes where you heard it when you were recording, not 10ms later or 15 ms later. This can destroy the feel of a song.

 

Play a click track and record it back into the daw. It should make a trip out and back into your converters. If you have a mixer that you use for monitoring send the signal through that. If it's more than a few ms late (when you play it back against the original click) you need to adjust your daw so that incoming signals are placed right where you heard them during recording.

 

Most people assume that's the case with all daws, it just 'aint so. That's why the daw mfgs give you options (should?) for making those kind of little adjustments.

 

Lawrence

Link to comment
Share on other sites

  • Members

We play a set of MIDI pads into our sampler to the DAW..over the years I have had to learn to play to latency ..You really just have to play slighty ahead of the beat.

Playing in a large venues, in the 60's halls, I remember having " latency" with the room sound~~ bouncing around..from monitor to FOH. . It was even worse than todays ms:eek:

Link to comment
Share on other sites

  • Members

Originally posted by The Audio Cave

What he said.
:cool:

BTW, here's a good tip for tracking virtual instruments...


1. Render a mix in the daw. Render don't play it and record it back in... rendering will give you sample accurate placing.


2. Shut all the other audio tracks off. You may have to disable them, not just mute them, to keep the disk from reading them anyway. Most daws still read tracks that are muted, they just don't play them back while the mute button is on. This is so when you unmute you hear it immediately. You might want to create a script that "disables" all selected tracks.


3. ASIO (I hope)? Turn your buffers way down to get very low low latency. 10-15ms is good for playing live.


4. Play the virtual instrument part in realtime using the rendered mix as a guide. It's a lot less strain on the system with very low buffers to play a single stereo track with no effects than 24 mono tracks with plugins going. It won't (shouldn't) pop and click.


5. When you're done raise the buffers and get back to work with the multiple tracks. Most daws should allows changing buffer sizes without rebooting.


I'll raise and lower the buffers in SX mulitple times during a session depending on what I need to do. Takes about 15 seconds. This method works well for tracking vocals and monitoring through the daw if you can get your latency down to about 3-5ms.


Other Daw Tips?


I also recommend testing your daw for near sample accuracy. What I mean by that is making sure that what you record goes where you heard it when you were recording, not 10ms later or 15 ms later. This can destroy the feel of a song.


Play a click track and record it back into the daw. It should make a trip out and back into your converters. If you have a mixer that you use for monitoring send the signal through that. If it's more than a few ms late (when you play it back against the original click) you need to adjust your daw so that incoming signals are placed right where you heard them during recording.


Most people assume that's the case with all daws, it just 'aint so. That's why the daw mfgs give you options (should?) for making those kind of little adjustments.


Lawrence

 

Hey Cave (and others) great advice man. But what is ASIO? ALso, If I do get the great river mp-2, it has two outs I guess it is dual mono since I plan on using two mics to play classical/acoustic guitar. I cannot monitor in real time from the preamp. MY DAW is all-in-box type DPS24, 24 tracks 24 bit, up to 96 (useless in my opinion), it has head phones outs near the nearfields out and monitor. I do not want to monitor there do I?

Link to comment
Share on other sites

  • Members

I don't think latency is really an issue anymore. The CPU, hard drives, RAM memory, the drivers and sound cards of today are so fast that latency is really not an issue in practise anymore. Only if you try to push the soundcard to the extreme by putting many virtual instruments in the chain you will notice that you'll need to raise the buffer size and the latency is increased.

 

Really, if you have latency issues I would recommend you to upgrade your gear. Latency really ruins everything! If you want to have a DAW with no latency issues I would recommend the RME Fireface 800 audio interface. So far I haven't had any latency issues with the Fireface.

Link to comment
Share on other sites

  • Members

Let me explain it better...

 

With native daws there is a trade-off between latency and available cpu. When running a daw with large buffers and big latency you can run a lot more plugins and vsti's. A lot more.

 

I run SX with the buffers set at 2048 (the largest setting for my MOTU 2408) which gives me an effective input latency of 47ms. WAY to large to try to monitor that way but I monitor through a console anyway. It stays that way until I want to play a VSTI from a midi keyboard and then I lower it to 128 0r 512k just to do that.

 

The big buffers give me very stable playback and lots of cpu for plugs and vsti's. It also has an additional beneficial effect. Lots more tracks.

 

No matter how fast your drive is (or isn't) you'll get more tracks if each disk read is grabbing more data for the buffers. If it has to keep going back for little chunks the disk is working harder and it's gonna be harder to keep up with an ungodly amount of tracks. If you can do 32 with a small buffer setting you can probably double or triple that with a very large buffer setting.

 

Try this. Play a song with lots of tracks at a low buffer/latency and with plugins and/or virtual instruments. Take note of the cpu meter. Look at the disk drive activity light on your computer. Now raise the buffers to maximum and look at the cpu meter. It goes down considerably. Look at the drive activity light, it lights much less frequently.

 

You just "recovered" cpu cycles that you can use for plugs when you mix. Or if you're running a 56 track project and getting audio glitches, you just put the project back within the capability of your disk drive by raising the buffers. Now it should play fine.

 

Lowering the buffers makes the cpu work harder which allows less use for other things like plugs. If I'm running a mix with lots of plugs at 85-90% cpu usage and lower the buffers to 128 or 512k? The song won't even play, it'll peak the cpu. So as I said I'll render the mix to a stereo file and disable all the other tracks, do what I need to do with my new vsti tracks, render those to audio tracks and raise the buffers to get right back where I was at 90% cpu with all of the tracks playing.

 

Even if you run low buffers for tracking and low latency you should probably increase them to the maximum for mixing. This gives you all the cpu power available to your system. Why settle for anything less when you're mixing and latency isn't an issue anyway?

 

Changing buffer settings to fit a particular application is a common practice in the pro PC native daw world. A PC daw is not really a "set it and forget it tool", it's adjustable to fit the application just like your outboard comp.

 

I use a Vaio 2.8ghz P4 with 2 internal drives. 7200 rpm audio drive.

 

Lawrence

Link to comment
Share on other sites

  • Members

 

Originally posted by psychdoctor

I have a question about latency when overdub. with a DAW. I was planning on purchasing a good mic preamp like the great river mp2 or vintech 1272 as well as an Art Pro compressor, maybe an TC M-one-xl verb unit. I was informed on another forum (Akai dps24) that I would experience latency...is that true? How does one deal with latency? It seems that latency would make it near impossible to overdub leads, vocals...just wondering.

 

 

LOL! The guys here are talking about COMPUTER-based DAWS. With your dedicated hard/software within your DPS24, you absolutely do not have to worry about latency in your specified application. Simply hook up your mic to your favorite preamp, preamp to your compressor, compressor to one of the 1-4 pre inserts, and VOILA! No latency. Hook up your future TC verb unit to the auxes on the DPS, and still no relevant latency. Track, overdub, and mix with no latency, all with full monitoring of built-in fx, or outboard fx. With dedicated hardware, latency is the last thing you should be worrying about.

 

Even if you ran recorded tracks out to a preamp then back in your particular recorder, you would be experiencing maybe only 1.5-3ms MAX of total latency.

 

Impossible to overdub on a DPS?!?! Especially when monitoring delay-verb during, with no latency?! Dude, you do NOT know the power you have. The thing is MADE for overdubbing. Period. PM me, or flood the DPS forum with questions.

Link to comment
Share on other sites

  • Members

Originally posted by Darth Balls




Impossible to overdub on a DPS?!?! Especially when monitoring delay-verb during, with no latency?! Dude, you do NOT know the power you have. The thing is MADE for overdubbing. Period. PM me, or flood the DPS forum with questions.

 

I thought I was missing something. The problem is, it was at the DPS forum (experts) that told me to keep an eye on latency with the deeps. I just did not want to go out and buy the extra equipment and deal with the latency beast. But what you said made sense to me, I just thought I was missing something..:eek:

Link to comment
Share on other sites

  • Members

I use cubase sx3 and the latest version has hardware delay compensation, which I find very useful!

It measures your input and output latencys and compensates the whole project to adjust in real time so outboard gear can be used without a hassle.

So your hardware just showes up in a bus or effects insert like a software plugin...except you can only use one instance of it ;)

 

How does Pro Tools LE deal with latency, is there delay compensation for plugins or outboard?

 

NB

Link to comment
Share on other sites

  • Members

I thought I was missing something. The problem is, it was at the DPS forum (experts) that told me to keep an eye on latency with the deeps. I just did not want to go out and buy the extra equipment and deal with the latency beas

 

They are talking about latency that is either imperceivable, irrelevant, or there for certain situations, but not yours.

 

If you were to overdub, utilizing your choice of mic, pre, and compressor, you would have literally imperceivable latency. You simply would not hear it.

 

If you overdubbed with the above mic, pre, and compressor, and patched a reverb unit into the auxes, you would have irrelevant latency. It would be more than the above example, but you simply would not hear it in the reverb, obviously and especially if your reverb predelay is set to anything higher than zero. :D

 

If you did not overdub, but patched your outboard compressor to an aux for mixing, you would indeed experience about 1.5ms of latency, and would notice this only if the effected sound would be blended in with the original track. This is what one of the DPS forumites were discussing. His solution is to align the waveforms in editing.

 

Indeed, the very phased sound you experience in the DPS when listening to the input and the track at the same time during recording is the result of a 15 sample delay a signal experiences when running through the digital mixer section of the DPS. You simply unassign the input or track from the L/R bus and record.

 

I use cubase sx3 and the latest version has hardware delay compensation, which I find very useful!It measures your input and output latencys and compensates the whole project to adjust in real time so outboard gear can be used without a hassle.

So your hardware just showes up in a bus or effects insert like a software plugin

 

Very cool. I've always wondered when and if that would ever transpire.

Link to comment
Share on other sites

  • Members

Originally posted by rickenbacker198

I use cubase sx3 and the latest version has hardware delay compensation, which I find very useful!

It measures your input and output latencys and compensates the whole project to adjust in real time so outboard gear can be used without a hassle.

So your hardware just showes up in a bus or effects insert like a software plugin...except you can only use one instance of it
;)

How does Pro Tools LE deal with latency, is there delay compensation for plugins or outboard?


NB

 

I'm finding this one of the coolest things about Cubase SX 3. I've got a Kurzweil KSP8 hooked up to my audio interface and now I'm using it on the FX sends in Cubase. Its very slick!

Link to comment
Share on other sites

Archived

This topic is now archived and is closed to further replies.

×
×
  • Create New...