Slow Input Level Meters.

With this set up ( turntable to preamp to mixer to MU-1212 soundcard to amp to speakers) I notice that when I monitor input levels
the meters lag up to a full half second behind the audio I am hearing. I am using a 2 gig processor with Windows XP and am recording at 96khz/32 bit. Incidentally, it also does this at 44.1khz/16bit. I have all other windows closed.  Am I missing something? It's a pretty simple set up in that I don't do multitrack recording, just stereo and mono from vinyl.  Thanks.

I know it sounds crazy, but the way to fix meter lag is to alter the latency setting of your soundcard (click on the  card Control Panel in the Hardware setup) - either up or down one notch, usually. This is all to do with the way that the information 'fits' into the ASIO channel. Audition doesn't derive its own meter feeds at all - they are actually provided by the soundcard, and are based on the information provided directly by the A-D converter. The clip indicators work the same way too - which is why on a few cheaper soundcards in the past, there have been a few otherwise inexplicable errors regarding this!

Thank you.  I will try that tonight.

What's the soundcard?  Audition 2.0's asio main metering with my Echo Mia Midi card has always been horrible - but other programs are fine, so it's kind of tempting to view this as an Audition problem rather than anything else.

What's the soundcard?  Audition 2.0's asio main metering with my Echo Mia Midi card has always been horrible - but other programs are fine, so it's kind of tempting to view this as an Audition problem rather than anything else.

I'm sure that's true - it's Audition that has to sort out the stream, and I don't think it does it properly.

The soundcard is an E-MU 1212m.

I've also noticed that Cool Edit 2.1 does NOT have the same problem using the same settings ie: 96khz/32 bit. The meters are right on the money. Go figure.

I've also noticed that Cool Edit 2.1 does NOT have the same problem using the same settings ie: 96khz/32 bit. The meters are right on the money. Go figure.

Of course it doesn't have the same problem - it doesn't use ASIO drivers, and the previous scheme worked fine!

The ol'  "It works so let's monkey with it 'till it doesn't"  scheme.  Well, I'm of for a week's vacation.  If anybody has any ideas I would be greatfull.  I can use it the way it is.  It's just an annoying.  Thanks!

The ol'  "It works so let's monkey with it 'till it doesn't"  scheme.

I know what you mean, but not quite this time. There was an enormous clamouring from all over the place for Audition to use ASIO, and Adobe eventually capitulated - only I don't think that a lot of people realised that there would inevitably be some limitations to this when it happened. Some of these are quite severe, and not Adobe's fault at all. For a start, you can only run a single soundcard - no mixing and matching unless you do it within a single manufacturer's scheme designed specifically for the purpose, and there are also all sorts of other niggles relating to the way ASIO channels are shared amongst running applications, for instance. Yes, it has less latency and enables you to do almost-real-time monitoring, but that's something of a luxury, really, in everyday life. It really was perfectly acceptable the way it was.

And so, if you'd left it to me - well, I simply wouldn't have bothered with ASIO at all - I made that quite clear at the time. What I would have preferred if we had  to go down this rounte was for a universal front end to be capable of driving WDM and ASIO, but that wasn't to be either. What AA2.0 has as an engine is pure ASIO, and to get a WDM input/output to it, you have to use the Audition Windows Driver - and that's less than stellar, as well.

What you could try doing, and some have reported that this actually helps anyway, is to use your soundcard WDM drivers with the free ASIO4ALL driver, which gets rid of quite a few limitations, often helps with the metering, and generally works better than the Adobe one anyway for a lot of people.

 Thanks for all your help. I hope you will indulge me further, while we're on the subject...   My E-MU 1212m has "sessions" folders which allow me to set the sampling rate. Such as  "44/48 digital default, product default, 96 minimum, 96 stereo record, 192 stereo record etc. You get the picture.  When I record vinyl or shallac records I set AA 2.0 to 32/96khz because I like to process at that rate until I am ready to down convert the file.  What I want to know is, does anyone have experience with this soundcard and do I need to use a certain "session" setting? Or, does the soundcard default to a certain setting?   I know the bit rate is 24. Thank you.