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Microphone Delay
mzso
post Apr 18 2012, 12:43
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Hi!
I bought a microphone, plugged it in and I have an irritating few tenth of a second delay. Really annying becuse Its basically impossible to talk when you're hearing your ownvoice, withthis delay. I guess it would be similarly irritating if I would chat with someone else. Where should I look for the problem? (I have a xonar D1 with UNi Xonar drivers, if its any relevance)
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Brand
post Apr 18 2012, 15:58
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Some programs let you choose ASIO or WASAPI, which would give you lower latency.

And do you need to hear your voice when speaking?
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mzso
post Apr 19 2012, 10:01
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QUOTE (Brand @ Apr 18 2012, 15:58) *
Some programs let you choose ASIO or WASAPI, which would give you lower latency.

And do you need to hear your voice when speaking?


Latency is bad even if only the other hears it. Plus its good to know what the other person actually hears if I chat.

QUOTE (Dynamic @ Apr 18 2012, 15:03) *
It's not a problem I get, but I don't have that soundcard. Firstly, because I don't need to hear back my recorded line, so I don't get any audio feeedback.

My soundcard has a direct monitoring output in analogue, and I don't use any DSPs on soundcard or Windows mixer, except occasionally in the application level (such as fb2k).

Virtual speakers (presumably for headphone use) would have a modest delay, but I'd be surprised it needs to be so big as that. Try removing it to troubleshoot [edit: I know you have removed it and still have a problem, but if there are two similar sources of delay, you might miss the fix by leaving one turned on while hunting]. Also beware that you might inadvertently be stacking both an internal hardware DSP delay in the Xonar and one in the Windows mixer system (which might be doing sample-rate conversion and mixing). If you use (intentionally or otherwise) very high quality SRC in either, that's likely to introduce a delay.

Some filter designs in DSP want to look beyond the 'current' sample so look ahead for so many samples, more samples ahead if you set them to higher quality mode. Some are 'causal' (just like electronic passive filters) and can't look 'ahead' into the 'future' stream of samples, so can only rely on present and past sample values to determine the output. Many sound DSPs assume you normally feed only recorded, pre-buffered audio to them so will introduce a delay to have enough 'future' samples to process because a non-causal design produces better performance in other ways.

I'm guessing you might be using Windows 7? The OS could have an influence.

I actually tried with the built in soundcard. Same thing. And yes I'm using windows 7. I tried matching the input sample rate to the output one, but nothing changed. Out of ideas. It was working without noticable delay on my brothers computer without setting anything...
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