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playback delay

Does anyone happen know why there is a playback delay when recording on to my laptop?

Even when an instrument is plugged directly into the laptop, there is a half a second or so delay between playing the instrument and hearing it through the headphones.

This is obviously quite a hindrance! Any ideas would be gratefully received

Comments

audiokid Sat, 02/18/2012 - 09:08

its called latency and why we invest in a quality interface. Sound cards are known to have an unacceptable amount of latency for pro audio. Read up on latency and if you want better results, look for better recording interfaces that use USB, Firewire, AES EBU.Price can vary from a few hundred to thousands.

Boswell Sat, 02/18/2012 - 09:38

This is quite usual when monitoring through software. There are input and output buffers for the sound samples in the computer so that the sound is not broken when the computer has to do something else for a fraction of a second. The delay is the time for the samples to get through the buffers.

You can shorten the delay by reducing the size of the buffers, but the danger is that you will get drop-outs. The solution is to get an external audio interface unit that has direct hardware monitoring, so that what you hear when recording has not had to go through buffers in the computer.

anonymous Sat, 02/18/2012 - 23:21

thanks

audiokid, post: 384643 wrote: its called latency and why we invest in a quality interface. Sound cards are known to have an unacceptable amount of latency for pro audio. Read up on latency and if you want better results, look for better recording interfaces that use USB, Firewire, AES EBU.Price can vary from a few hundred to thousands.

Thanks for your reply,
Indeed! ..However, I have used the same set-up (digi desk and assorted outboards) with my previous two computers without any latency at all. Does that not mean that the issue is pc based? Could the factory installed IDT audio package be the culprit? If so, how does one cure this? Cheers

anonymous Sat, 02/18/2012 - 23:38

Thank you

Boswell, post: 384649 wrote: This is quite usual when monitoring through software. There are input and output buffers for the sound samples in the computer so that the sound is not broken when the computer has to do something else for a fraction of a second. The delay is the time for the samples to get through the buffers.

You can shorten the delay by reducing the size of the buffers, but the danger is that you will get drop-outs. The solution is to get an external audio interface unit that has direct hardware monitoring, so that what you hear when recording has not had to go through buffers in the computer.

Thank you for explaining this. Is a there a limit that one can set the buffers to before drop out occurs, or is that system specific? (and solved through trial and error). I have noticed that the buffers are set quite large in my cool edit suite, but I am also worried that the factory installed IDT audio on the new pc might be hindering it as well. (?)

I have used the same set-up (same software, digi desk, outboards etc) on two other pcs without any issues at all, so in theory, the 'interface' should be adequate.(?)

much obliged to you

Boswell Mon, 02/20/2012 - 03:14

Unregistered, post: 384683 wrote: Thank you for explaining this. Is a there a limit that one can set the buffers to before drop out occurs, or is that system specific? (and solved through trial and error). I have noticed that the buffers are set quite large in my cool edit suite, but I am also worried that the factory installed IDT audio on the new pc might be hindering it as well. (?)

I have used the same set-up (same software, digi desk, outboards etc) on two other pcs without any issues at all, so in theory, the 'interface' should be adequate.(?)

much obliged to you

No, not really. Unless you dedicate your computer solely to audio and remove all screensaver, internet, email, anti-virus and similar programs, you are not going to be able to get the buffer sizes down to a level where they work with sufficiently low latency.

You may get somewhere by reducing buffer sizes, but the real solution to the problem is an external audio interface that has hardware monitoring.

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