Help with interface!

  • warzone (nov 5-9) signup begins in...

damird

ILLIEN
ill o.g.
I just bought M-Audio Mobile Pre USB recording interface.

It has Phantom Power and the mic that I'm using on it is M-Audio Nova Condenser microphone.

However, when I try to record vocals, I only get recorded on the left side/speaker. I use Sony Acid 4.0 to record.

Can you guys tell me what I'm doing wrong? How can I make it record on both sides?

The interface itself has 2 XLR Mic channels. Am I somehow supposed to plug in the mic into both of them to get sound on both sides?
 

damird

ILLIEN
ill o.g.
I tried lookin' for mono setting, I can't find it. Can anyone guide me through finding it, please? Or is there any other recording program i could use?
 

damird

ILLIEN
ill o.g.
BROUSSARD BEATS said:
sometmes you need stereo jacks at the ends of you mic plugs 1line at the end means mono 2 lines stereo

Sorry, I don't know what you mean. I use an XLR cable to connect my condenser mic to the interface. I dont use 1/4" or 1/8" jacks.
 

Big Tone

You done fucked up
ill o.g.
you can change the input settings to record only the left side. this way you get a mono waveform instead of stereo with an empty right track. ive moved recently and my computer is still packed away so i cant paste a pic for you.
 

damird

ILLIEN
ill o.g.
Big Tone said:
you can change the input settings to record only the left side. this way you get a mono waveform instead of stereo with an empty right track. ive moved recently and my computer is still packed away so i cant paste a pic for you.

I did this and it worked. However, there is so much static when I record and i'm only able to record in 48,000 & 16 bit. Also, its obviously mono, but does this mean that my recordings are gonna be shitty no matter what? I mean whatt's he point of having a condenser mic and an interface if i'm only gonna get shitty quality from it? With my dynamic mic, that I hook straight to the computer, I get 192,000 and 24 bit stereo.

Can someone clarify for me a little bit? I'm sorry i'm so clueless, but i'm trying to learn. Is there a way I can get the best possible performance out of my condenser and the interface? I mean, I was told that this was the way to go, so I decided to invest in it.
 

Hypnotist

Ear Manipulator
ill o.g.
1) Stereo tracks should only be used when there is separate left/right information. Two microphones in stereo should be used in two separate mono tracks, panned left and right. (Room mics on a drum set, overheads, etc).

2) When you record at 48kHz are you dithering? Dithering means taking it back to 16-bit without adding any static noise.

3) Check your XLR cable. Sometimes these are faulty and create static because the ground (pin #1) is faulty.

I've never used ACID to record vocals, or the interface, so I can't help you there.
 

damird

ILLIEN
ill o.g.
Hypnotist said:
1) Stereo tracks should only be used when there is separate left/right information. Two microphones in stereo should be used in two separate mono tracks, panned left and right. (Room mics on a drum set, overheads, etc).

So does this mean when I'm recording my vocals, I should just record in mono? My interface does have 2 channels. However, my condenser mic is connected to only channel 1 through XLR cable.

Hypnotist said:
2) When you record at 48kHz are you dithering? Dithering means taking it back to 16-bit without adding any static noise.
I don't quite understand this concept of "dithering." Before I had an interface and before I used the condenser, I used a dynamic Audio Technica Mic. I would connect is straight to the back of the computer where the mic output was. When I would use that mic and record in Acid, it would allow me to set settings to 192,000 and @ 24 bit. Now that i'm using the interface and the condenser mic, it won't allow me to use those settings. An error message pops up and says that the setting cannot be used due to a conflict with the interface i guess. Again, I have M-Audo Mobile Pre USB and M-Audo Nova Condender Mic. I'm using it on Sony Acid 4.0


Hypnotist said:
3) Check your XLR cable. Sometimes these are faulty and create static because the ground (pin #1) is faulty.
The XLR cable I have is brand new. I believe I paid about 20 bucks for it. Do the different XLR cables play important roles in recording quality?

Hypnotist said:
I've never used ACID to record vocals, or the interface, so I can't help you there.

Well, can you recommend me a program that's simple, affordable, and able to run on P4 1.9 GHZ with 256 SDRAM? lol.
 

Hypnotist

Ear Manipulator
ill o.g.
Yes. One microphone = one signal = mono. Don't worry about when you actually bounce the signal down to a stereo mix. Your stereo mix will sum up your mono channel to equal left/right information, and give you a pseudo-stereo mix of your mono channel. All lead vocal tracks are in mono, panned right up the middle. Now if you take a voiceover and import it from a CD, it's stereo because it came from a CD, but only one of those channels is important, as both left/right information is exactly the same, which makes it mono.

In terms of recording at 192kHz, you must have a really good soundcard with this capability of 192 inside your computer (where you plug in to the back of your computer to mic input and soundcard line out). However, when I looked up your M-Audo Mobile Pre USB, the specs said that the highest sample rate this can perform is at 96kHz. USB is a slow connection, even 2.0 is slow compared to high-powered digital audio cables that one would use in the studio for a real 192 interface. Don't worry about dithering if you don't understand it. But try to read up about it.

Yes. Some audio cables, like Monster Cable, can cost up to $100 for a 20-foot XLR cable. There are other brands that I once thought were fine, but after I bought them they almost damaged my microphone because it wasn't grounded right. Inside the cable, you have a few leads and shielding, which could be varying in quality themselves. I don't know if that's the problem with the static, but it could be a possibility. Also, what could be a problem is that you're clipping your audio level, and it's too high. This would cause some funky unwanted sounds to come out. You could be clipping at any stage in the signal flow: At the microphone itself (if you're screaming at it or you put a kick drum on a condenser mic), at the pre amp input, at the pre amp output, in your audio software, post-fader if your level is too high in your software, and your soundcard output, your receiver, and then your speakers.


There is much for you to learn. I don't know who told you to get a 192kHz soundcard or told you to record at 48 or 96, but at your level (and trust me, I'm not talking down to you) you should only be messing with 16-bit, 44.1kHz recordings. If you go above that sample rate or bit depth, you could be causing yourself more problems if you don't know about it. But if you WANT to record at 48 or 96 (I would suggest 88.2 instead of 96 because you're converting back to 44.1 eventually) then read as MUCH AS YOU CAN about the process of recording. Most people don't have a clue what any of these numbers mean, aside from the fact that it's "better resolution of audio". But if you don't have the essentials down, like how frequency and amplitude (the only 2 factors in audio) are handled in the digital domain by applying sample rate and bit depth, then you won't get much out of a 96 or 192kHz recording.

I would suggest starting with Adobe Audition (Cool Edit Pro after Adobe bought it from Syntrillium) because it's pretty user-friendly, and everything you need is right in front of you. Check to see if Adobe Audition works with your USB Mic Pre (I'm sure it does), and read up!!!

Hope this helped.

Pce.
Hypno
 

damird

ILLIEN
ill o.g.
Hypnotist said:
Yes. One microphone = one signal = mono. Don't worry about when you actually bounce the signal down to a stereo mix. Your stereo mix will sum up your mono channel to equal left/right information, and give you a pseudo-stereo mix of your mono channel. All lead vocal tracks are in mono, panned right up the middle. Now if you take a voiceover and import it from a CD, it's stereo because it came from a CD, but only one of those channels is important, as both left/right information is exactly the same, which makes it mono.

In terms of recording at 192kHz, you must have a really good soundcard with this capability of 192 inside your computer (where you plug in to the back of your computer to mic input and soundcard line out). However, when I looked up your M-Audo Mobile Pre USB, the specs said that the highest sample rate this can perform is at 96kHz. USB is a slow connection, even 2.0 is slow compared to high-powered digital audio cables that one would use in the studio for a real 192 interface. Don't worry about dithering if you don't understand it. But try to read up about it.

Yes. Some audio cables, like Monster Cable, can cost up to $100 for a 20-foot XLR cable. There are other brands that I once thought were fine, but after I bought them they almost damaged my microphone because it wasn't grounded right. Inside the cable, you have a few leads and shielding, which could be varying in quality themselves. I don't know if that's the problem with the static, but it could be a possibility. Also, what could be a problem is that you're clipping your audio level, and it's too high. This would cause some funky unwanted sounds to come out. You could be clipping at any stage in the signal flow: At the microphone itself (if you're screaming at it or you put a kick drum on a condenser mic), at the pre amp input, at the pre amp output, in your audio software, post-fader if your level is too high in your software, and your soundcard output, your receiver, and then your speakers.


There is much for you to learn. I don't know who told you to get a 192kHz soundcard or told you to record at 48 or 96, but at your level (and trust me, I'm not talking down to you) you should only be messing with 16-bit, 44.1kHz recordings. If you go above that sample rate or bit depth, you could be causing yourself more problems if you don't know about it. But if you WANT to record at 48 or 96 (I would suggest 88.2 instead of 96 because you're converting back to 44.1 eventually) then read as MUCH AS YOU CAN about the process of recording. Most people don't have a clue what any of these numbers mean, aside from the fact that it's "better resolution of audio". But if you don't have the essentials down, like how frequency and amplitude (the only 2 factors in audio) are handled in the digital domain by applying sample rate and bit depth, then you won't get much out of a 96 or 192kHz recording.

I would suggest starting with Adobe Audition (Cool Edit Pro after Adobe bought it from Syntrillium) because it's pretty user-friendly, and everything you need is right in front of you. Check to see if Adobe Audition works with your USB Mic Pre (I'm sure it does), and read up!!!

Hope this helped.

Pce.
Hypno

Dude. Thank you very much for a detailed response. I will read as much as I can.
 

Sanova

Guess Who's Back
ill o.g.
Battle Points: 9
Im having trouble with the same interface. My condenser is connected viz XLR as well and i have it set to mono, but when i record no input goes to the computer. the interface is hooked up properly via usb and i can hear myself talking via direct monitor, but No Input Records. help?
 

Sanova

Guess Who's Back
ill o.g.
Battle Points: 9
Nvm I got it down. Good lookin out tho. It was a technical issue within the program <cooledit>
 

damird

ILLIEN
ill o.g.
I got another question. I record about 4-5 takes in Acid.. and then the program freezes. I usually have to manually shut down the computer or wait 5 minutes to end task on Acid. When i would record without the interface and through a dynamic mic, this never happened. Now that I'm using the interface and the condenser, it happens all the time. What could be the problem?
 

Sanova

Guess Who's Back
ill o.g.
Battle Points: 9
Try adjusting the audio settings on acid. if the input is set to 'try as WPDM' then itll fuck up. Also, it depends on if ur running a cracked version of the software.
And lastly go to m-audio.com and download the updated driver for the mobilepre. i highly reccomend it if you havent done so already. 1
 
E

Equality 7-2521

Guest
Just to clarify something: dithering is for converting bit depth not sample rate.

I recommend recording at 24 bit / 44.1 kHz & then dithering back down to 16 bits for burning to CD.

For whoever didnt understand dither, it is low level random noise added to the audio which will effectively raise the level of the quietest sounds in your mix e.g. reverb tails etc so that they are not cut off abruptly when the level goes below the lowest bit. To hear this, load up the Waves L1 and lower the input levels (you will need to turn up your master volume going out to your monitors because the L1 input levels must be as low as possible). Loop a reverb tail and have dither set none. You will hear the sound cut out randomly. now set dither to 16 bit. You will hear a bit of random noise but your reverb tail wont cut out.

Also, if your having problems with faulty XLR cables, it is not related to the cable being an XLR. It's not like XLR's have a higher chance of being faulty. XLR is the professional standard. Try getting some Canare cable & Neutrik connectors next time.
 

Sanova

Guess Who's Back
ill o.g.
Battle Points: 9
^ good info. Waves diamond bundle is a good plugin set. I recommend investing in it.. if you havent found it freely available on the net *hint*
 

Hypnotist

Ear Manipulator
ill o.g.
Yea, good job clarifying.

Bit depth is basically how many amplitude levels there are... Imagine a volume knob with only 4 volume notches, and you go to turn it up and it's too loud; turn it down and it's too quiet... A wider bit-depth will better enhance the quality of audio, especially if your recording has a wide dynamic range. (It goes from quiet to loud and uses up a lot of the amplitude spectrum) Hip hop and rock don't do this. They're usually maxed out, or "slammed" with compression, so you have almost no dynamic range.

However... when you record at 24-bit, your "word-length" is 8 bits longer than 16-bit, and there is more resolution of amplitude values. All CDs (redbook) are 16-bit, 44.1kHz, stereo files. So when you eventually convert back into 16 bit, if you don't dither, then you just cut off the other 8-bits of your word length. This creates what's called "quantization noise", when the amplitude levels go from low to high, without smoothing themselves out. It's almost like clipping, only it's not maxing out your amplitude, just rough changes within your levels.

ANY ways... What dithering does is takes the last string of your word length (the last 8 zeros and ones) and carefully places them back into your 16 zeroes and ones to make it almost seamless with your amplitude levels, and you don't get "crunchies" as my professor used to call em.

I may have done a horrible job explaining this, but if you draw it out, or look online at figures that illustrate it, maybe that will explain better in case anyone wants to know.

Pce,
Hypno
 

damird

ILLIEN
ill o.g.
No man, I greatly appriciate you guys trying to explain these things for me. Sure, I still don't quite get all of it, but that's fine.

I have another question. What can you guys recommend for me to do in other to learn to EQ beats and vocals so they sound consistent and not like trash. I use FL studio 5.0 to produce, but I mostly mess with outside samples that I cut up, sound founds, and non-freeware plugins. I've been producing with FL for almost 3 years, but I still don't think I EQ my tracks that well.

When it comes to vocals, I already stated I used Acid 4.0. Now that I'm following your guys' recommendations, I'm recording in 44-48 @ 16 bit. However, even though my vocals sound way better than they did on my dynamic mic, I still find them too plain. How do I go about EQ-ing vocals? Earlier in the thread you guys talked about compressing the vocals and few other things. I really don't understand which route should I take about doing this. How should the compressor be set up? Can someone clarify? I know that I'm asking too much, but I really need to learn and you guys here have already been helpful.
 
Top