Audiophile 24/96

[7string]

I just installed this card and messed with it a bit. Man, the latency is outta this world!!!! Can somebody give me some tips to tighten it up aLOT?

Also, WHERE should I adjust the levels when recording? The mixer software that comes with it doesn't seem to have much effect on the levels when recording and so far the only way I've found is to play at lower volumes to get down to -12. I'm not using an outboard mixer, just pretty much direct into the Audiophile.

Pointers anyone?

 

[dgatwood]

I just installed this card and messed with it a bit. Man, the latency is outta this world!!!! Can somebody give me some tips to tighten it up aLOT?

Reduce the buffer size. With a reasonably fast machine, you should be able to track with a 128 byte buffer size. I think the default is 1024 or 2048, both of which would be very noticeable.

Also, WHERE should I adjust the levels when recording? The mixer software that comes with it doesn't seem to have much effect on the levels when recording and so far the only way I've found is to play at lower volumes to get down to -12. I'm not using an outboard mixer, just pretty much direct into the Audiophile.

Many M-Audio devices without physical input level controls ship with a control panel that allows you to adjust the input levels in software. That said, I think the only adjustment might be pretty coarse (adjusting the -10/+4 setting)....

 

[7string]

Reduce the buffer size. With a reasonably fast machine, you should be able to track with a 128 byte buffer size. I think the default is 1024 or 2048, both of which would be very noticeable.

Ok, found that and will give it a try. Thanks!

Many M-Audio devices without physical input level controls ship with a control panel that allows you to adjust the input levels in software. That said, I think the only adjustment might be pretty coarse (adjusting the -10/+4 setting)....

I'm seeing a Consumer button and another one right below it that is -10dBV. Is this what you mean and should I set it to -10dBV?

 

[mrface2112]

yes, two things......first, reduce the buffer size as low as you can get it without introducing pops/clicks into the recording.

second, in the m-audio software mixer, make sure you're monitoring off of the "monitor mixer" set of inputs.


cheers,
wade

 

[7string]

yes, two things......first, reduce the buffer size as low as you can get it without introducing pops/clicks into the recording.

second, in the m-audio software mixer, make sure you're monitoring off of the "monitor mixer" set of inputs.


cheers,
wade

I tried that. It goes silent. It only works with the WavOut 1/2 selected.

 

[7string]

More help please!

Ok, I have the 2496 hooked up and have recorded some things. The latency is still nuts but I just don't listen to it. Now, today I went to record some vocals and --- problem. How do I hear the mix? I used to just plug the headphones into the headphone jack in the front of the computer. With the new audio card that jack doesn't work anymore. Will I need to get a headphone amp and run it from one of the 2496's outs? I have a 3-way speaker switcher that I'm using to switch between the Wharfie's and the 2.1's for checking mix translation and the third bay is open. Can I run it from there into a headphone jack? Or is there something I'm missing somewhere?

Two more days of problems and I'll start to regret getting this new schtuff...

 

[7string]

Oh... one more thing...

I record to a track and it only records on ONE side of the track (left). So I do a mix of the track to a mono or stereo track, bring it back in and suddenly I can't pan the damned thing! All this crap MUST be something easy that I'm just missing or I'm taking it back and doing the ol' lady with the tire through the window routine...

It's FRUSTRATING!!!!

 

[DonF]

Will I need to get a headphone amp and run it from one of the 2496's outs?

That would be one option. I run the output from the 2496 to the line in of my onboard soundcard, and monitor from the onboard's headphone output. So I'm using the onboard soundcard as my headphone amp.

Don't give up, man! Help is on the way. Let's just take one step at a time, and you'll be all set in short order. There are definitely some aspects of setting up the Audiophile that are "non-intuitive," but it's worth the effort.

What software, what OS, and what are you plugging in to the Audiophile (instrument, preamp, etc.)? Is your 3-way speaker switcher working at line level, or at the outputs of an amp? If it's line level, you could possibly connect the third output to the line input, if you want to use the "soundcard as headphone amp" trick.

Don

 

[gordone]

Each input on the audiophile (and any other soundcard) is MONO (One signal). So you want to setup your recording software to record to one mono track. When you playback/mixdown, your recorder software allows you to place each mono track anywhere in the stereo spectrum using the Pan controls.

 

[7string]

That would be one option. I run the output from the 2496 to the line in of my onboard soundcard, and monitor from the onboard's headphone output. So I'm using the onboard soundcard as my headphone amp.

Don't give up, man! Help is on the way. Let's just take one step at a time, and you'll be all set in short order. There are definitely some aspects of setting up the Audiophile that are "non-intuitive," but it's worth the effort.

What software, what OS, and what are you plugging in to the Audiophile (instrument, preamp, etc.)? Is your 3-way speaker switcher working at line level, or at the outputs of an amp? If it's line level, you could possibly connect the third output to the line input, if you want to use the "soundcard as headphone amp" trick.

Don

Great idea using the onboard soundcard. However, I had to disable it in XP to use the Audiophile. Maybe it will still work for those purposes though so I'll give it a shot.

I'm using Guitar Tracks 3 Pro with XP. What I run into the Audiophile depends on what I'm trying to record. Guitar can be mic'd amp or direct using an amp modeler. Bass is usually through a pre. Normal schtuff...

I run the outs from the Audiphile into the in's on the switcher and then one set of outs goes to the Wharfie's and one goes to the 2.1's for reference. The third is empty so I was thinking maybe I could feed the headphone amp from there. I'll probably go with the headphone amp just because I'll probably be needing it eventually if I have somebody do vocal tracks with me. At least that's about the only thing I'd use it for.

Thanks for the help! Much appreciated!

 

[DonF]

OK, now about those recording levels. What are your choices in GTP as far as inputs you can select to record from? The program may "see" the Audiophile as three different devices, depending on the driver in use. You have the analog hardware inputs (the RCA jacks), the S/PDIF inputs (an RCA jack on the breakout cable), and the monitor mixer. (There is a similar choice to make for your output device.)

So, to get a signal into GTP:

1. Connect the instrument, preamp, or whatever to one of the Audiophile's analog inputs.

2. In GTP, connect that Audiophile input to a track (I don't know GTP, so if this part is difficult one of us will have to do some research) and set that track to record mode. (We'll get into zero-latency monitoring later -- for now just connect the hardware input to the track without going through the mixer.)

3. Make some noise and look at the record level you're getting in GTP. If it's too low, turn up the preamp, play louder, or whatever.

Are we getting anywhere?

Don

P.S. Not to complicate things, but are you sure that the Audiophile can't coexist with onboard sound? What happened when you tried it (if you tried it)?

 

[7string]

OK, now about those recording levels. What are your choices in GTP as far as inputs you can select to record from? The program may "see" the Audiophile as three different devices, depending on the driver in use. You have the analog hardware inputs (the RCA jacks), the S/PDIF inputs (an RCA jack on the breakout cable), and the monitor mixer. (There is a similar choice to make for your output device.)

So, to get a signal into GTP:

1. Connect the instrument, preamp, or whatever to one of the Audiophile's analog inputs.

2. In GTP, connect that Audiophile input to a track (I don't know GTP, so if this part is difficult one of us will have to do some research) and set that track to record mode. (We'll get into zero-latency monitoring later -- for now just connect the hardware input to the track without going through the mixer.)

3. Make some noise and look at the record level you're getting in GTP. If it's too low, turn up the preamp, play louder, or whatever.

Are we getting anywhere?

Don

P.S. Not to complicate things, but are you sure that the Audiophile can't coexist with onboard sound? What happened when you tried it (if you tried it)?

Actually there's about a half-dozen or more choices when I choose the source and I found the one that is 'mono' so that problem is fixed. And I've decided to get a small headphone monitor amp because you never know when I'll have somebody else singing harmony with me and I'm going to go out of the 3-way switcher that I already have so THAT problem is solved too.

Now... ON TO LATENCY!!!! Man, I set the Latency samples in the M Audio Delta Control Panel to 128 samples. I get a few clicks and pops here and there, nothing very loud at all and not very often at all. Certainly not often enough to irritate me. But the latency is STILL way too bad to try to record with. Right now I just click the monitor off button, uncover one ear and monitor through my amp modeler, which seems to do the work. However, when it comes time to do vocals it's not going to work because the mic will pic up the source. So this issue does have to be fixed or any attempts at singing along to a track will end up on America's Funniest Home Recordings.

The help is much appreciated, DonF!!!!

EDIT: Oh, and during installation of the Audiophile it basically made me disable the integrated sound card because of a resource issue, which is weird since this machine is a Pentium 4 3.20Ghz, 3.19Ghz 512MB ram.

 

[DonF]

You might have to move the Audiophile to a different PCI slot, if there's truly a resource conflict. But since you really don't need the onboard sound, this is a non-issue for you. On to latency.

The setting in the control panel is for the size of the DMA buffer. That's only part of the latency equation. 128 samples at 44.1 kHz is only 2.9 milliseconds. This is roughly equivalent to the time it takes sound to travel 3 feet. The actual latency you're experiencing is probably much worse than that, if it's affecting your ability to record.

The rest of the latency comes from your software's internal buffers. The software retrieves the samples from the DMA buffer, and places them into another buffer until it's ready to do whatever it's going to do with them (write to disk, add effects, etc.), in between screen redraws and other stuff that the software is doing. If the software doesn't empty the DMA buffer in time, you get a pop or click due to the discontinuity of the waveform. The time it takes for the software to process the samples and spit them out is added to the hardware latency to get your total system latency.

So, what to do? It may be possible to tweak some settings in the software to achieve lower latency, but I haven't been able to locate any documentation specific to your software, so you're kind of on your own there. But the best solution from the standpoint of minimizing latency for overdubs is to use the Audiophile's "zero-latency monitoring" feature.

First, you have to turn off any software monitoring that GTP is doing. Otherwise, you'll be getting a direct signal from the Audiophile, as well as the latent (delayed) signal from the software.

Then, go to the Delta control panel's "Patchbay/Router" page. Set the "HW Out 1/2" output (the RCA jacks) to get its signal from the Monitor Mixer. Then, on the "Monitor Mixer" page, your software's output will be on the "WavOut 1/2" faders. Make sure they're not muted, and bring them all the way up. Also make sure the Master fader is not muted, and is all the way up. Now, un-mute the "HW In 1/2" channels, and bring up their faders. You should be hearing the signal from your input, with no latency. The same signal should still be going to the software's inputs, as long as you have the right device selected. This could take some trial and error. But zero-latency monitoring is a very cool feature, so keep trying.

Are we there yet?

Don

 

[7string]

You might have to move the Audiophile to a different PCI slot, if there's truly a resource conflict. But since you really don't need the onboard sound, this is a non-issue for you. On to latency.

The setting in the control panel is for the size of the DMA buffer. That's only part of the latency equation. 128 samples at 44.1 kHz is only 2.9 milliseconds. This is roughly equivalent to the time it takes sound to travel 3 feet. The actual latency you're experiencing is probably much worse than that, if it's affecting your ability to record.

The rest of the latency comes from your software's internal buffers. The software retrieves the samples from the DMA buffer, and places them into another buffer until it's ready to do whatever it's going to do with them (write to disk, add effects, etc.), in between screen redraws and other stuff that the software is doing. If the software doesn't empty the DMA buffer in time, you get a pop or click due to the discontinuity of the waveform. The time it takes for the software to process the samples and spit them out is added to the hardware latency to get your total system latency.

So, what to do? It may be possible to tweak some settings in the software to achieve lower latency, but I haven't been able to locate any documentation specific to your software, so you're kind of on your own there. But the best solution from the standpoint of minimizing latency for overdubs is to use the Audiophile's "zero-latency monitoring" feature.

First, you have to turn off any software monitoring that GTP is doing. Otherwise, you'll be getting a direct signal from the Audiophile, as well as the latent (delayed) signal from the software.

Then, go to the Delta control panel's "Patchbay/Router" page. Set the "HW Out 1/2" output (the RCA jacks) to get its signal from the Monitor Mixer. Then, on the "Monitor Mixer" page, your software's output will be on the "WavOut 1/2" faders. Make sure they're not muted, and bring them all the way up. Also make sure the Master fader is not muted, and is all the way up. Now, un-mute the "HW In 1/2" channels, and bring up their faders. You should be hearing the signal from your input, with no latency. The same signal should still be going to the software's inputs, as long as you have the right device selected. This could take some trial and error. But zero-latency monitoring is a very cool feature, so keep trying.

Are we there yet?

Don

Ok, I looked around in my software and see no place to turn off software monitor. I did, however, find this:

Enable Read Caching (which is NOT checked)

Enable Write Caching (which is NOT checked)

I/O Buffer Size (KB): 64

Then there's Driver Mode, which is set to Use Default. The other options are:

WDM/KS
ASIO
MME (32-Bit)

According to the software the Effective Latency is now set at 2.9msec

We're getting there!

 

[DonF]

I think they call it "input monitoring." Look for that in your audio options.

You should definitely try the ASIO driver if you can. I think the default is WDM. MME would be a last resort.

Let me know what happens.

Don

 

[7string]

I think they call it "input monitoring." Look for that in your audio options.

You should definitely try the ASIO driver if you can. I think the default is WDM. MME would be a last resort.

Let me know what happens.

Don

Well the ASIO definitely introduced constants pops and clicks so that's out. But I made some other changes and the latency problem seems to be cleared up too! I'll do some more testing later and get back to you.

Thanks for all your help! It really is appreciated!!!!!

 

[DonF]

All in a day's work.

I'm interested to know what you did.

Don

 

[7string]

All in a day's work.

I'm interested to know what you did.

Don

Well, basically, I went into GT Pro 3 and under the Audio Options I moved the Buffers In Playback Queue from Safe to Fast which is 2.9msec. That did the trick! I'll know for sure Monday when the headphone amp gets here and I can record vocals, but recording instruments seems to be right on. No noticeable latency at all! God, I love this! hahahaha

 

[7string]

Ok, I went ahead and switched to ASIO and moved the DMA Buffer Size Latency to 256 Samples and it is working with no pops at all! I figgered if I messed around long enough I'd either get it right or screw it beyond help. hahaha

Anyway, Donf, thanks much for pointing me in the right direction! Now, which is better to run, RCA In or SPDIF In? Or does it depend on the project and how I'm recording? What's the difference in quality? In other words, which is better, why and what's the differences between the two?

Damn, my ex always SAID that I made things too complicated...

 

[DonF]

...which is better to run, RCA In or SPDIF In? Or does it depend on the project and how I'm recording? What's the difference in quality? In other words, which is better, why and what's the differences between the two?

The difference between the two is that the S/PDIF in/out is digital, and the analog RCA ins/outs are, well, analog.

If you have a digital source that provides S/PDIF connectivity, that's what you should use. If you have an analog source, use the RCA analog inputs. Using the S/PDIF input would allow you, for example, to use one of these (click), and still have your two analog inputs active for microphones, or whatever. Or you could add two very high-quality channels of A/D conversion with one of these (click). So, using S/PDIF and analog together, you can record or play back a total of four channels at once with the Audiophile.

Myself, I've only used the S/PDIF input a couple of times. Once to record some music off of a DVD, and another time to copy something from an old DAT tape. Worked like a champ in both cases.

Don