c414b-xls what should self noise measure in db?

[kingof12]

I bought this mic second hand after reading good reviews about it's versitillity and quiet spec. However having used it now for a while I am wondering if it may have a fault as it's self noise fluctuates around the -48db mark in motu cue mix when I am using it through a dav bg1u pre amp and then into the motu 828mk3 interface. Does this figure seem right for this microphone or should it be quieter than this?

Any advice very much appreciated.

[Bossman]

What kind of room are you in?... I would imagine the ambient noise of the room would be higher than the self noise of the mic... No?

-48dB, I assume you meant dBFS. Does this number change when you adjust the mic gain on the pre-amp?

Try recording with it (if you haven't already), like spoken words, make sure you have the gain set correctly. Try close and distant mic placements and check if the noise is a problem in practical use.

And have you got any other mics to compare to?

[kingof12]

Cheers for reply. I'm in attic room in quiet area only birds and river noise. The noise increases with increasing mic gain on pre amp. Compared to rode classic tube mic which I would have thought would have an inherently louder self noise. However the c414 is a bit higher on the meters than that which is what first started me wondering if it had a problem...

[Bossman]

it could be that the 414 is more sensitive and so it is picking up more room noise with a similar amount of gain... does that sound likely?

is it a problem in normal use? can you hear the noise, when monitored at a sensible volume - try recording/playing back and use your ears (not eyes/meters) to tell you if there is too much noise.

[Tomás Mulcahy]

Yes, it's most likely room noise+ pre amp noise, but there are too many variables here, pre amp gain being the main one!! So you're not measuring self noise here, it's the noise of your room+ pre amp+converter. Post a clip of the recorded noise and we can diagnose it.

Do you own an iPhone? SPL meter by Studio Six Digital is accurate to within +/-1dB for measuring dBSPL, A or C weighted. That will give you much more meaningful info on your room noise situation. Despite what your ears are telling you, there could be LF noise in your area that the mic picks up but your ears don't (see Equal Loudness Contour ). So try switching in the HPF on the 414, if the "noise" level decreases, then there's some heavy machinery near you generating LF.

And take careful note of Bossman's correction: it's -48dBFS (FS is Full Scale, that is the type of meter your DAW has). There's no such thing as "db" and deciBels on their own are meaningless. They must be referenced to something else to have meaning.

[kingof12]

Okay thanks to you both for help with this. If I get time this week I will try and post a recording. I am maybe stressing about nothing as I am still learning the ropes when it comes to producing recordings. Also like you say there are so many variables in the signal chain I could be getting something wrong in the gain staging or elsewhere. A couple of other questions are:

At 24bit does the sampling rate make a huge difference to the recording quality ? (My motu only allows reverb for monitoring at lower sampling rates)

What signal level should I aiming for at the preamp stage ( the bg1u has 0 8 and over leds)

What level should I be looking for on the protools metres?

Thanks for the tip on dbfs although I'm non the wiser what that actually means other than being a measure of the signal level.

[Jonesd90]

The most obvious example of somebody trying to hear exactly what they sound like whilst singing is Bryn Terfel. If you watch him sing he curls the corner of his mouth so to project some of his resonance into his ear.

It's interesting as if you try hard enough it does actually work!

However it looks awkward...

Dave

[Bossman]

At 24bit does the sampling rate make a huge difference to the recording quality ? (My motu only allows reverb for monitoring at lower sampling rates)

as long as you use at least 44.1KHz then you'll be ok. don't go lower than 44.1KHz.... you don't need to go any higher than that either.. some people might argue the benefits of higher sample rates, but really there are better things to worry about. stick to 44.1KHz if I were you.

What signal level should I aiming for at the preamp stage ( the bg1u has 0 8 and over leds)

.. I've not used that unit, but you probably want the 0dB light flashing with the +8dB light coming on sometimes.

What level should I be looking for on the protools metres?

you want to be aiming for an average level of around -18dBFS. peaking up to around -10dBFS. or very occasionally as high as -6dBFS, but try not to let it get any higher than that.... theres no need. leave yourself plenty of headroom and things will be a lot easier when you come to mix.

my quick rule of thumb is to divide the level meter into three equal thirds... then aim to keep the level somewhere in the middle third.

Thanks for the tip on dbfs although I'm non the wiser what that actually means other than being a measure of the signal level.

its dBFS, not dbfs.. Decibel is a lowercase 'd', then uppercase 'B'. FS should also be uppercase

FS means Full Scale. In digital audio the level meter ranges from 0dBFS at the top (you can't go any higher than this), down to minus infinity (nothing) at the bottom. The rest of the numbers on the meter are referenced to Full Scale. So -10dBFS is 10dB below Full Scale.

[kingof12]

Thanks for all the advice. It does seem like I've been getting the gain staging right so I think your point about a more sensitive mic picking up more background noise could be the issue here. I've been recording at 96khz so had to use a plugin reverb for the monitor mix but pro tools kept crashing so hopefully if I record at 44.1khz or 48khz this will not happen as I can use the reverb from the motu. Incidently if cds are at 44.1khz why would I ever use 48khz?

[Tim Gillett]

Incidently if cds are at 44.1khz why would I ever use 48khz?

You've nailed it. There is much ignorance out there about sample rates higher than 44.1. Neil Young, bless him, isnt helping the situation at the moment. I've heard it said that that since the CD is now over 30 years old, we should have progressed to something better by now, as if since 1982 our ears have evolved to a superior model.

Tim

[kingof12]

Ha! It does seem that if 44.1 is good enough for cd then why not for the devices doing the original recording. Are the higher sampling rates just a case of the audio companies needlessly uping the spec to sell more units?

[Tomás Mulcahy]

Yes the higher rates are a numbers game created by marketing departments, not engineering. The funny thing is the (modern) A to D is oversampling in GHz in the first place! Higher sample rates can make a difference with badly designed compression and eq plugins though (the ones that don't oversample and haven't paid attention to Nyquist and Shannon).

[James Perrett]

Incidently if cds are at 44.1khz why would I ever use 48khz?

48kHz is the standard rate for video sound.

[Mattyy]

I am a little hazy on bit rate and sample frequency as well.
In reading, pros always tell you to go as high as your resources will allow.
At some point, I was led to believe that if I record everything as clean as possible and with plenty of headroom:
1. The extra bit depth aids in the compression process, ensuring that the information is present when I try to bring it up to audible levels.
2. The extra frequencies (above 20K) aid with eq and pitch shifting, again providing information that may be brought out in the mix etc...

Of course, I understand that everything in the recording chain has a dynamic range and frequency response but with technology going off the rails (ie. Adam tweeters going up to 50K) maybe higher numbers are a way of future proofing recordings?
Or is it all just distracting us from concentrating on pitches and rhythms?
Curious.

[Hugh Robjohns]

In reading, pros always tell you to go as high as your resources will allow.

Not really. That sounds like something that would be said by someone who doesn't know what they are talking about... Which isn't a 'pro' by definition

True 'pros' know that a decent converter running at 44.1k gets the job done for 90% of applications, and that 96k can be useful where the converter is poor, or where you are close miking instruments with a lot of strong HF harmonics. 96k is also useful for some forms of non-linear signal processing, but decent plugins deal with that internally anyway.

Sample rates over 96k generally degrade performance rather than enhance it: jitter gets worse and there is nothing audibly useful in the extended bandwidth.

At some point, I was led to believe that if I record everything as clean as possible and with plenty of headroom:

Sounds like a good plan..

1. The extra bit depth aids in the compression process, ensuring that the information is present when I try to bring it up to audible levels.

Not really. The extra bit depth allows you to leave a sensible headroom margin without compromising the system noise floor. In practice the ambient acoustic noise floor will be far higher than the 24 bit system noise floor, and through compression the acoustic noise floor will far outweigh the system noise floor. The same is usually true even of 16 bit systems...

2. The extra frequencies (above 20K) aid with eq and pitch shifting, again providing information that may be brought out in the mix etc...

Nope. Not unless you're planning to pitch shift down by an octave of more!

Of course, I understand that everything in the recording chain has a dynamic range and frequency response but with technology going off the rails (ie. Adam tweeters going up to 50K) maybe higher numbers are a way of future proofing recordings?
Or is it all just distracting us from concentrating on pitches and rhythms?
Curious.

Bigger numbers are a way of up selling equipment. And given that the ear hasn't really changed in several millions (or thousands) of years (depending on your belief preferences), I don't think we need to worry too much about future proofing with silly sample rates and unacheivable bit depths.

It would be better to worry about the music, the performance and the recording environment. Those are the things that really matter.

H

[tacitus]

+1. The only reason we spend time converting archive recordings to digital is that the music, performance and recording are good enough to make us want to do it.

As Hugh says, our ears haven't changed, and if we were originally blown away by quality digital recordings at 44.1KHz, there's no reason why they shouldn't still be just as good. Just bear in mind the first digital recordings were, by and large, done by top performers playing great music and being recorded by the best engineers in the best halls. You can work out for yourself what's changed since then ...

[Mattyy]

Thanks for the clarification guys
I think that there is a lot of money to be made in selling upstart musicians the idea that they can get true pro results with little more than a laptop, and a some magic $300 product.
Changing the settings in my DAW to get magically better/professional results is appealing to someone who can't afford a properly treated space, decent monitors and clean converters.
As someone who is willing to save and invest in the right information/gear/space, it is refreshing to have these myths dispelled.
That being said, I still want a better set of speakers. LOL!!
Cheers.

[Richie Royale]

http://www.soundonsound.com/sos/feb08/a ... laudio.htm

I post this on here too often . An excellent article from Mr Robjohns about the world of digital audio.