Microphone issue: surrounding noise

[bouldersoundguy]

Now that you've solved the basic trouble you had here's something you should try:

Set the first knob (Gain), while singing strongly into the mic, so that the meter on the preamp reads around 0dB, sometimes flashing the +6dB light.

Then adjust the second knob (Output) so that the level meter in your recording software reads around -18dBFS on average, with peaks around -12dBFS and not higher than -6dBFS or so.

Once you get the Output knob set you should be able to simply set the Gain knob for whatever different sources or mics you might want to record.

 

[ecc83]

Now that you've solved the basic trouble you had here's something you should try:

Set the first knob (Gain), while singing strongly into the mic, so that the meter on the preamp reads around 0dB, sometimes flashing the +6dB light.

Then adjust the second knob (Output) so that the level meter in your recording software reads around -18dBFS on average, with peaks around -12dBFS and not higher than -6dBFS or so.

Once you get the Output knob set you should be able to simply set the Gain knob for whatever different sources or mics you might want to record.

That is Recording Levels 101 BSG but our OP is using OBS and so might be stuck with 16bits?
Now of course, whilst a 16 bit system CAN be fine it does not have the "headroom" ease of use of 24bits )44.1kHz).

He seems to have come round a bit to AIs? Had he not and been UK based I was about to offer him a loan of my KA6!

Dave.

 

[rob aylestone]

The problem you have is simply that the audio ins and outs built into the most expensive motherboards are designed for gaming, audio output and low cost. Computers usually need very little quality consideration to have a headset plugged into them, quality audio is not a a requirement. The other issue is a technical one. With all the high speed data transfer on the busses and various controllers, inside the PC is a mess from the perspective of RF and EM interference. Low level, unbalanced audio is contaminated, hence why external devices perform better, simply by getting sensitive analogue component areas away from the noise. It's always been this way, and I've tried and still have some vey expensive internal cards. In fact, my whizz expensive audio only PC has an internal card, and it's just not good enough on the noise front, swapping to an external Tascam, for multichannel a also brought down the noise considerably.

Could you record 30 seconds of your mic and you speaking, with peaks approaching 0dB so we can assess how bad your issue is? I've used the mic you have and frankly I didn't dislike it at all, and the noise performance was NOT an issue. I guess we are wanting to hear the noise for ourselves. Listening to stuff I record in the control room not the studio, for convenience does reveal a small amount of fan noise, but it's so low, the wanted signal makes it a non-issue for me. I'll record voice,sax and acoustic guitar in the 'wrong' room with no worries.

You mentioned you have nailed the instruments. Are these real, synthesised, sampled or what. Were any of these recorded with the same mic setup. Finally, what internal card are you using? If it's a sound device built into the actual mother board, then I have never found a decent one for recording. Many are excellent for output, however. Balanced audio facilities, usually equate to good, and 3.5mm sockets don't!



EDIT bummer! My ipad didn't show me page two, so that was a waste of finger work. Never mind, good to see you sorted it.

 

[ecc83]

Hi Rob.
If that expensive soundcard is beaten on the noise front by a Tascam AI I would say you have an unfortunate combination of PCI card and MOBO? Maybe it has a dirty supply rail?

I have just checked the rec/play noise of a 2496 in this Asus 6 core new build and it returns -98/97dBFS unweighted with no input.

The M-Audio spec quotes -104dB "A" so I would think my figures are pretty close? In fact I have used 2496 cards now in some 4 computers and they have all run better than -95dB u/w.

Yes, my KA6 manages 101dBFS U/W but that is a MUCH later AI with converters a couple of generations on from the M-A!

But all theses figures are pretty meaningless since as soon as you connect even the best pre/mixer+mic they will degrade by a good 10dB. And we should get our rooms so quiet!

And of course. Remember, the AI pre amp is in a wee tin box with a computer!

Dave.

 

[bouldersoundguy]

That is Recording Levels 101 BSG but our OP is using OBS and so might be stuck with 16bits?
Now of course, whilst a 16 bit system CAN be fine it does not have the "headroom" ease of use of 24bits )44.1kHz).

He seems to have come round a bit to AIs? Had he not and been UK based I was about to offer him a loan of my KA6!

Dave.

Oh well, then he uses -12dBFS average and peaks not over -6dBFS or so until he gets 24 bit conversion.

 

[YanKleber]

Now that you've solved the basic trouble you had here's something you should try:

Set the first knob (Gain), while singing strongly into the mic, so that the meter on the preamp reads around 0dB, sometimes flashing the +6dB light.

Then adjust the second knob (Output) so that the level meter in your recording software reads around -18dBFS on average, with peaks around -12dBFS and not higher than -6dBFS or so.

Once you get the Output knob set you should be able to simply set the Gain knob for whatever different sources or mics you might want to record.

Our mate c7sus had already given me such instructions:

Try setting the gain so your vocal is hitting just about 0dBv on the preamp. With the output adjusted properly that should peak at about -18dBFS in your DAW software.

And I thanks to this as one of the reasons why I achieved the success of my tests right in the first attempt!

 

[YanKleber]

That is Recording Levels 101 BSG but our OP is using OBS and so might be stuck with 16bits?
Now of course, whilst a 16 bit system CAN be fine it does not have the "headroom" ease of use of 24bits )44.1kHz).

He seems to have come round a bit to AIs? Had he not and been UK based I was about to offer him a loan of my KA6!

Dave.

I just checked my DAW options and the 'Sample Format' was set by default to 16bits but I have the options 24bits and 32bits available. Should I change these settings will improve something? Would it cause a perceptible difference? Also the 'Sample Rate' is set to 44100 Hz.

And by the way, Dave, thank you for thinking about to loan me your AI but I am very far from you -- Brasil!!! Yeah, that huge country that was humiliated by Germany in the WC finals. LoL.

Anyway, I am going to pass an AI for now because I think that the only thing that it could benefit me at the moment would be a latency compensation, that's correct? I don' think that it will be a problem to me because I am not getting any noticeable latency, and even if I do I can always fix it in the DAW by dragging the vocals samples a bit forward in the track to match all the rest. Usually I have 'monitored' my own voice through the air while I hear the backtrack in the headphone so not a latency problem when monitoring too.

All in all I think that in my particular case wouldn't worth to spend something aroung $150-250 on an AI. Remember that I am not a pro and am doing it just for hobby in an improvised home space. And believe me, I really got a stuning result (for my standards) after the latest corrections on my setup.

 

[bouldersoundguy]

I just checked my DAW options and the 'Sample Format' was set by default to 16bits but I have the options 24bits and 32bits available. Should I change these settings will improve something? Would it cause a perceptible difference? Also the 'Sample Rate' is set to 44100 Hz.

Setting word length ("bit depth") to 24 bit in your DAW does no good if the converters aren't capable of 24 bit. It will simply fill in the extra bits with zeros. Working in 24 bit gives you a lot of extra space to accommodate unexpected variations in input level.

Anyway, I am going to pass an AI for now because I think that the only thing that it could benefit me at the moment would be a latency compensation, that's correct? I don' think that it will be a problem to me because I am not getting any noticeable latency, and even if I do I can always fix it in the DAW by dragging the vocals samples a bit forward in the track to match all the rest. Usually I have 'monitored' my own voice through the air while I hear the backtrack in the headphone so not a latency problem when monitoring too.

Record latency (which can be compensated for in the DAW) and input monitoring latency are two different things. The DAW should automatically compensate for record latency and/or have a way to set it manually so you don't have to move anything. Input monitoring latency is when you hear your live voice/instrument delayed in your headphones while recording. The only way to reduce or eliminate that is to use hardware with the ability to let input signal monitoring bypass the trip through the computer. AIs do that. A mixer could be used to do that. No stock sound cards do that as far as I know.

Remember that I am not a pro and am doing it just for hobby in an improvised home space.

Almost everybody on this site matches that description, and almost all of them find an AI to be the best way to go.

 

[ecc83]

Yes,
ALMOST! Everyone here has an AI but I dun different!

Because I have an electronics background and some audio kit to start with in 2005, I went for a PCI sound card. Wasted money and time on an S(of a) B and a Terrtec (awful people!) . A Trust Optical Expert was in fact pretty good for £20 but I eventually fetched up with the M-A2496. All my noise and driver problems were banished and son could get on with guitar and MIDI recording unhampered by "technology"!

One of these: M-Audio Audiophile 2496 Sound Card 612391350109 | eBay
Would be a good move IMHO....Till you get an AI!

Dave.

 

[bouldersoundguy]

If it has hardware input monitoring that bypasses the trip through the computer I'd still call the Audiophile 2496 an audio interface.

 

[ecc83]

If it has hardware input monitoring that bypasses the trip through the computer I'd still call the Audiophile 2496 an audio interface.

Ooooo! Pikky! Well, even the ***t device built into the MOBO is TECHNICALLY an AI!

One thing the 2496 can do that few external interfaces* can is allow me to record computer sound. Thus I can capture radios 3 or 4 without any pluggeratin' whatsoever.

The MixControl software that goes with my 8i6 might be able to do this, route PC sound to an input but I have not so far had the time to investigate it deeply.

Ah! Just saw what you said about bypassing the PC. Hmm? Not that cute about computers to say. Ask me about biasing valves!

Dave.

 

[YanKleber]

Modulator, I see your point. Better to leave it at 16Bits then since it is sounding OK for my ears. Thank you for the tip!
About the 'monitoring latency' what I mean is that I don't monitor my voice on headphones. I just hear my voice in the ambient as I sing. In the headphones I just hear the backtrack.

 

[Rod Norman]

Without actually hearing the sound, I'm guessing it might be a ground loop. Try plugging your mic through different inserts (DI, Tube pre, etc.) and see if there is any difference. Try to isolate the cause. I do hope there is a ground plug in there somewhere at the wall; yes?
Rod Norman
Engineer

Hi all!

After almost 30 years of delay I am finally trying to make this dream about to record an album to come true.



Thanks to all the awesome DAW software I am doing good with the instruments part. However, when the thing comes to the voice recording I am really being knocked out. The main issue is that whenever I record my voice, it is getting an unwished environment noise, like a buzz mixed with an hiss.



My gear is a C-3 mic and a Mic100 tube preamp, both from Behringer. They go plugged directly into my PC (onboard card) and the place where I do stuff is a regular room without any acoustic treatment. I believe that 99% of the problem is on my own inability of set up the equipment. The microphone has a DIY pop-killer. The preamp has two controls (gain and output) and the only thing I know is that cranking up one or another control will make the recording level to get up. Dohhh! I know that a wrong balance between those two controls will only probably add more noise to the equation though...

I have found a few tutorials teaching how to build a portable vocal booth (and that supposely would help to reduce the environment surrounding noise) but I am not feeling too much excited on add an extra stuff before to be sure that I am doing OK with what already I have.

Two other information that may be important:

1) I am not a singer so I have a bit of fear of the microphone and usually stay farest away from it than I probably should (around 10 inches between my mouth to the mic)
2) My place is very quiet (I live in country side) and the only audible sound in the environment is the usual hum from computer motors

Can someone give me tips to put me in the right track?

Thanks thanks THANKS a lot!!!

 

[YanKleber]

Without actually hearing the sound, I'm guessing it might be a ground loop. Try plugging your mic through different inserts (DI, Tube pre, etc.) and see if there is any difference. Try to isolate the cause. I do hope there is a ground plug in there somewhere at the wall; yes?
Rod Norman
Engineer

Thank you for your attention, Rod, but the issues are all gone now.