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Kaizen

Recording delay

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Hey. I record wee scraps of music here and there onto my laptop, just for the hell of having them down incase i want to use them later. The problem is, i get alot of delay, like im playing along to my programs built in metronome then when it comes up after on the screen, the sound waves are miles off the actual beat. And then obviously when i try to record vocals on top of that, they are out of sync with the out of sync guitar... Its most likely because of my "setup", but i just though i would see if anyone knows any magic tricks to make it stop... save me the hassle of re-syncing everything all the time.

I litterally just plug my guitar/mic into a preamp then take an out lead from the pre-amp directly into my sound card. My sound card is a creative sound blaster. I realise thats not exactly a proffessional kit-out, but thought it would be worth a bash asking, incase theres a big "make it work" button somewhere that ive just missed...

Ta!

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I'm pretty sure the reason is because you're taking and out from your preamp. I don't think they're fast enough basically. Not sure how you'd go about fixing it though. This happened to me. Apart from just re-syncing it i think you just have to invest in an interface.

I could be really wrong though.

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Guest Tam o' Shantie

the reason is that you're not using an asio driver, which your soundcard may or may not have anyway.

try downloading this ASIO4ALL - Universal ASIO Driver and choosing it from your recording software as the input. should eliminate those annoying delays and works even on low spec PCs.

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YUS! a "make it work" program! almost as good as a button! =P Ta very much. I'll give that a bash.

Oh and I'm using Krystal Audio Engine at the moment. Its pretty pants as far as features go... but its free, so I'm not complaining. I just download VSTs anyway.

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the reason is that you're not using an asio driver, which your soundcard may or may not have anyway.

try downloading this ASIO4ALL - Universal ASIO Driver and choosing it from your recording software as the input. should eliminate those annoying delays and works even on low spec PCs.

How do they work/what do they do?

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Guest Tam o' Shantie

"ASIO Latency

ASIO is a standard for audio device drivers created by Steinberg. As much as possible, ASIO bypasses the Windows or Mac operating system, creating a more efficient communication between the audio device and the software. Currently, all of the Steinberg programs use ASIO (of course), while other programs (including software synthesizers) have also adapted the standard. The question, "what's the latency" in this instance is only relevant if the program is ASIO-compliant and if the audio card's device drivers also contain ASIO drivers.

Then, this question will need to be answered by the sound card company. Different sound cards will have different latencies at different sampling rates -- the higher the sampling rate, the lower the latency. In that sense, the latency occurs in numbers of samples, dependent on the number of samples that need to be put into a buffer before monitoring begins. Because the latency in samples is fixed or defined by the card, then the faster the sampling rate, the quicker a fixed number of samples will pass through the buffer. Hence, faster sampling rates = lower latencies. Often, a buffer size can be set in the sound card's control panel, and a lower buffer size = fewer samples that need to be buffered. As long as your system can handle the lower buffer size, lowest is best.

The latency in this case comes into play when we are monitoring in a "tape type" fashion, which is essentially monitoring through the program. We'll discuss Windows latency later in this article, which also affects the user while monitoring, but only with the efficiency of ASIO are we able to achieve this type of monitoring. It goes something like this: While I have the program in 'input,' I hear my instrument from the inputs of the program much like a pro tape deck. If I 'roll tape,' or rather put the program into play, I no longer hear the instrument until I punch in, again like a pro tape deck. All of this, unlike a pro tape deck, occurs with a bit of latency between what you're playing and what you're hearing through the program.

Steinberg says that 11 or 12 milliseconds of latency is acceptable. You can be the judge. At higher sampling rates, 3 ms latency might be possible. If you desire this type of monitoring, which is a fairly normal and accepted way of recording, then this may be the best that hard disk recording has to offer."

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you could also try increasing your buffer size in your DAW when recording...that might counteract your latency problem a bit?

also i'm pretty sure you can get cubase/nuendo etc for free and are probably better/more recognised programs to be using.

edit: by recognised i mean have more support etc.

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My sound card is a creative sound blaster. I realise thats not exactly a proffessional kit-out, but thought it would be worth a bash asking

I don't mean to be negative but your sound blaster is really not made to record audio. You might be able to get it to sound ok-ish but there is likely to be problems with latency etc.

A Firewire or USB2 recording interface is generally a better option for this purpose.

Only things I can suggest is to decrease your sample rate and look through the options in your control panel and see if there is anything you can change. Also check the options in your recording programme to see if anything you can alter to improve performance.

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i've got a similar problem at the moment.

i'm ruinning an Alesis IO14 into a firewire port into a decent spec PC with Cubase LE. I can monitor the guitar directly through the alesis, but when i mute the direct signal and just monitor through Cubase there is about half a second of latency.

I've decreased the buffer size to the lowest it can got (64) i think (but just reading earlier posts somebody said that you're meant to increase it to reduce latency) and i've set the latency compensation to medium. It doesn't seem to have changed the setting at all really (before it was set on 1024 buffer and low latency compensation). This is all using the Alesis' ASIO driver, which judging by the rest of the soft/hard/firmware may not be very reliable... would i be better off using the ASIO4ALL driver?

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Latency is a pain in the arse, and aas has already been sugggested, ASIO drivers are the way to go, but they won't get rid of it entirely.

I have found that a good way to get rid of it is to use a wave Editor such as Audacity (free download) and record onto that. You can then zoom in on your audio track and manually cut out the slight delay you will see at the very start, then align your track back to zero.

A ittle cumbersome, but I've had good results with that.

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