Attention Delta Audiophiliacs

  • Thread starter Thread starter rats
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rats

rats

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I may have settled in with the Audiophile by M-Audio. Maybe. I got it about a week ago and I'm having problems that may soon be resolved but I'm looking for opinions from home reccers that are familiar with it.

First off, I have recently come across statements to the effect that prosumer cards (I'm assuming that this is considered a pro-sumer card?) generally are RAM hogs compared to cheaper consumer cards like SB Live. Is this true? I'm currently at 128 MB of RAM and I was running 20+ tracks of music in cool edit pro on my cheap soundcard with no problems. Will I need to upgrade my RAM in order to do this level of recording with the audiophile?

Secondly, I have had troubles getting my driver updated for Windows ME. I initially installed the driver out of the box and the install Wizard worked fine for that. However when I later downloaded the ME update, the Wizard would not show his face and left me with no clue how to perform the update. So far support has not been a big help.

Although I haven't updated the drivers, I can play and record OK basically with the audiophile. The only major problem I have is that I can't monitor what I'm recording as I record it through the card (I need to use an external mixer to monitor the input signal in the chain before it gets to the card mixed with the output from the card to hear what is being played back while multitracking). Is there some secret to monitoring the input signal through the card? Will the ME driver update cure this?

Also, last night I was tracking and several takes sounded fine when played back, then all of a sudded the new takes sounded all distorted and crackly. This was my first session using the Audiophile. Will the driver update cure this?
Should I run back to Guitar Center and return this thing now?
Does anyone know how to install the driver updates after the fact?
:confused:
 
Wow, that's a heap of questions, let me see what I can do.

First, I don't know anything about Windows ME. I'm using W2K specifically so that I can use the WDM drivers available at m-audio.com. I think the .26 driver is the latest. Re. your update, If you are having problems I seem to recall reading that there is an uninstaller at m-audio that you need to run before you reinstall.

I haven't noticed a significant impact on CPU utilization, but I know there is some impact when you use the WDM drivers to achieve low latencies. The impact for me is not from the card itself it is from DX effects and Dxi's running (I use Sonar).

If you can't hear what you are recording while you then somewhere along the line duplexing has gotten disabled. I'm pretty new to the Audiophile myself so I don't remember if there is a specific setting to enable that like there is with SBLive, sorry (I'm at work but I'll look when I get home).


>Does anyone know how to install the driver updates after the >fact?

Not sure what you are asking but you can install drivers at any time. It is possible that the crackling is due to a driver issue. Whether it fixes your issue depends of course on whether it printed your take with the crackling, or whether the playback is distorting.
 
You need to route through the "monitor mixer" to hear what's playing back along with what you're recording. Set this up in the "patchbay" section of the Delta control panel. It'll be pretty obvious.

More memory is only required if you do 96Khz recording. At 96Khz you're pushing twice the data of 48Khz, and therefore more memory will be consumed by applications & effects. Regardless, if you've got 128MB or better, you're fine. And I don't care much for 96khz recording anyhow.

Don't really know about the update....are you just talking about updated drivers? I think they come in a self extracting zip file, so extract them to a specific location on your hard drive. The go into the Device Manager and select the Properties for your Audiophile under Sound, Video & Game controllers. Once you get the properties up, look for the "Update Driver" button. Then follow the wizard....typically chose "search for an updated driver" and specify the folder you unzipped the drivers to as the location to search (there will be several options, CDROM, Floppy, Win Update, and Specify Location....you want the latter). This is a common way to install drivers on Windows...in fact most drivers are installed this way.

With your history of soundcard issues, I would look at your system before getting too worked up about any specific card. You've tried both the MIA and the Audiophile, and I am 100% confident that both cards are of high quality (in the price range). With the audiophile you'll want to experient with the DMA Buffer settings in the Delta Control panel, and the buffer settings in your software. I typically set my Delta DMA Buffer size to a pretty small value (<1024) and then I set my buffers in n-Track to about 55ms and this seems to work great, but that's using WDM drivers. If I use ASIO in n-Track, then I have to set the Delta DMA Buffers to ~55ms worth of latency and n-Track will inherit those settings automatically (e.g. software does not control hardware ASIO buffer settings).

Good luck duder!

Slackmaster 2000
 
Thanks peeps for the help.

Reqs that uninstaller tip may be the key. I will definitely try that first. I don't recall any duplexing switches but I'l take a look.

Slack I've unmuted everything to no avail but I will take a look at the patchbay. That sounds vital. Pardon my ignorance on this one, but what does a higher khz recording do to a recording? I'm not familiar with this but curious.

So this is the mysterious wizard! Did not know that! Well goddamn I'll have to give that a try!

Hey Slack I can you reccomend anywhere to read up on buffer settings and how they relate to a recording? I kinda understand the relativity but not completely. Anyway, thanks for all the help.:eek:
 
I know you don't use Cakewalk, but...

In Cakewalk, under options/audio, there's a bunch of settings to play with regarding aidio bit depth, sample rate, file bit depth, # of buffers/buffer size, I/O buffer size, etc... I messed with this for days (if not weeks) due to pops and crackles and other god awful noises in my playbacks subsequent to installing my Delta 66. Nothing worked.

I too am cursed with the wicked Windows ME, the driver for which M-Audio did not include in their OEM software package. I finally found the correct driver for the Delta/Windows ME, and problem solved (although I can't remember if there was any trick to installing it). Once I got the correct driver installed, it worked perfectly.

The only thing I have to contend with now are audio drop-outs. That's where all the tweeking on the options/audio menu come in.

Give Gidge a scream. He's an audiophile GURU (hell, I wouldn't have gotten the Delta up and running if it weren't for him... ...although I wouldn't have bought it if it weren't for him either... I wonder if he's getting a kick-back, the bastard :D ) Good luck, Tom.
 
Rats, the difference between 44, 48, and 96khz recording is heavily debated. Very simply, a dude name Nyquist discovered that the highest frequency that a system can accurately sample will be equal to one half the sample rate. So, at 44khz, we can sample frequencies up to 22khz, which is right at the max upper bound of human hearing. At 48 we of course can bump that up to 24khz which is basically above the human hearing range, and everything should be fine, right? Well, some people argue that the human ear is a little more complex than we give it credit for, and that supersonic frequencies will have an impact on sound. I'm not convinced of that, but I'm not an expert. There is also another train of thought that says that the more samples you produce, the more your DSP algorithm has to work with, but I'm not sure that's really a benefit because most DSP isn't that great sounding to begin with. I think the most rational explanation I've heard is that if a system can decently sample at 96khz, it should be pretty rock solid at 48 and 44khz.

I personally record at 44.1khz because I can't tell the difference between it and 48khz, and it's easier to stick with the standard sample rate. For instance, I can export 16/44 waves from fruity loops and import them into n-Track, and they'll sit along-side 24/44 tracks without any conversion process required. Which brings us to another point, sample rate conversion can be a more complex and critical task than it sounds, and perhaps in some cases it's better in terms of sound quality to just stick to one sample rate over a project.

The patch bay in the delta control panel is definately the source of your probems. You're one mouse click away from being straight-ahead on track.

In regards to buffer settings, there is no real good source, because too much depends on both hardware and software. The idea is this: data doesn't flow through your system in a perfect, predictable way, so we implement what are called "buffers". When you hit "play" in your application, it doesn't just grab the first sample it finds on the hard drive and send it to the soundcard, it grabs a "bunch" of samples and places them into a buffer. Samples then move from this buffer to the soundcard, where they might be rebuffered before playing back. Buffers are continuously filled as they are emptied. If there is a long enough clog between the source and the buffer (for instance the hard drive and your audio application), then the buffer will empty and the audio stream will break. Usually this only happens for a brief moment, and what you hear is a "click".

What you need to be concerned with, in terms of buffers, is latency vs. performance. Latency, in simple terms, is the amount of time it takes for you to hear the result of a change (fader move, etc) when mixing. When you move a fader in your audio application, the samples that are buffered up waiting to be played do not get changed, only the samples that occur after the fader move. So if you're buffering up say 1024 samples, and your sample rate is 44,100hz, then you're going to have a 1024/44,100 = .023 second latency (or 23ms). Using low latency ASIO or WDM drivers, it is typically acceptable to mix with latencies in the <60ms range. With regular MME or DS drivers, you'll see latency in the 250-500ms range (1/4 to 1/2 a second) which is really not a lot of fun to work with because you have to move a fader a little bit, wait, move it some more, wait, move it a tad bit more, wait, etc.

It seems like it would be best to adjust your buffers such that you'd have as little latency as theoretically possible, and that's true. However, if you set your buffers too small and they will empty faster than they're filled, and you'll get dropouts. That's the tradeoff.

Latency doesn't have an impact on the tracks you're recording because most soundcards today support "Zero Latency Monitoring." What that means, basically, is that inputs are routed directly to ouputs when you're recording. Now if you try do use something like Live Input Processing (such as using Ampfarm for guitar when recording), then you'll hear a noticable delay when you're playing.

I'm not sure how buffer settings work sometimes. The Delta control panel has gone through about 5 different buffer setting systems over the past year, and each one is more confusing than the last. It seems to me that if you use WDM, you have a set of buffers created by your software, AND a set of buffers used by the soundcard drivers. If you use ASIO, it seems like the audio application and the drivers use the same set of buffers.

Using WDM, I leave my Delta buffers at 128 (or 256) and then set n-Track up with buffers such that my latency is about 55ms (n-Track displays buffer sizes in both latency, and actual size). Using ASIO, I set my Delta buffers up to about 1024 or 2048 and n-Track inherits these settings (meaning that any changes to n-Tracks buffer settings has absolutely no affect).

Slackmaster 2000
 
Hey Slack hats off to you you have a way with psenting things. That makeas a LOT of sense about the latency buffering!

I'm still not where I need to be on the driver update issue. I followed your map to the M audio driver properties but there was no option there to update the driver! See atatched...

I did find in the Delta Control Panel the patchbay router and had the options of
wav out 1/2
monitor mixer
s/pdif in
s/pdif in (l/r rev.)
h/w in 1/2

And when I selected h/w in 1/2 I can monitor my input signal through the card but what I was specifically looking for was something I can monitor the input signal AND the playback signal at the same time. I don't hear anything when I select the monitor mixer. I double checked and the faders are all up and nothing muted that shouldn't be in the monitor mixer.:confused:
 

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Hey Rats,
I am using Windows ME and have a delta 1010 and 44. Just this past week I was looking to update my drivers.

heres how:
control panel/system/device manager/sound,video,and game controlers/M audio/driver tab/update driver.

The thing is.....I have been using an older version (one of the first ME drivers M-audio put out) and dont have any problems , .........except sometimes i get a pop at the begining of playback. The update is supposed to fix this.
When i run the update i get a message saying something like...the one im using currently is better?
dont know what to do.
hope this helps,

rodvonbon
 
a round of applause for rodbonvon! Ok so now I have updated my drivers!

I was hoping that would be the last of my problems but not so, because I'm still getting the clicks. I have my M audio buffer size latency set to 1048, but I'm still getting the clicks while recording. The weird thing is that it's only in present in medium to high input volume recordings. At low level recordings it does not appear. Listen to this...
 

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Ok dude...

Here is how you should configure your patchbay:

http://www.slackmaster2000.com/delta1.gif

...of course only the first panel of the patchbay applies to the audiophile.

Next, when I am using WDM drivers (e.g. n-Track), my buffer settings look like this:

http://www.slackmaster2000.com/delta2.gif
http://www.slackmaster2000.com/ntrackbuffers.gif

When I am using ASIO drivers (e.g. Fruity Loops), I set my delta control panel DMA buffers to 1024 or 2048 and fruity inherits these settings. All ASIO applications inherit the settings in the delta control planel. Like such:

http://www.slackmaster2000.com/delta3.gif
http://www.slackmaster2000.com/fruitybuffers.gif

Slackmaster 2000
 
Your mp3 sounds like you're clipping the input of the audiophile. Bring up the delta/audiophile control panel and watch the VU meters.

Slackmaster 2000
 
Hey Rats, I listened to the MP3s. I can hear the clicking-type noise on the track(s), but it doesn't sound like the horrendous clicking and popping I had with my M-Audio/ME problem. Did installing the new driver take care of some of the noise problem, or is this the extent of it(ie nothing changed). The noise sounds like part of your system being overdriven(like Slak stated), but also sounded to me like some sort of interference. Maybe something else in youe audio line/configuration??? ...Don't let 'em beatcha...
 
That 2nd checka boom boom sounded somewhat clipped.

Both of them sounded too noisy for me to believe that you've got your front-end set up right. The audiophile should be quieter than that.

Here's a sample from the GINA.

 
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Well- mine was. But that other rat's recorded stuff sounded like electronics hum.
 
hey rats,
have you tried to switch yer input settings under the hardware/settings tab?
maybe try -10db? or con?
i have a 1010 and a 44.
the 1010 goes in at -10 and the 44 goes in at +4.
dont know why, but it seems to make both signals pretty even.
i run my DMA buffers at:
wave-20-mill
asio/easi-2688
i use the same settings for both cards.

i dont have an audiophile, but the mp3's sound like yer input is too "hot"? maybe?....if not ignore me.


chasing a night of guiness with a tall glass of milk is not a good idea, but as "BOB" says in his seven comandments "too much is always better than not enough"
night,
rodvonbon
 
If you have a +4 vs. -10 selection somewhere, I'd go with -10 to get your signals in liine with accepted standards.
 
Hey,
its wierd.
-10 on the 44 is way too low and +4 on the 1010 is way too hot. It could be.....? i dont know.
they seem to work happy together this way:)
all i know is that when i move the faders on the board it's all relative to whats happening in vegas.
 
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