Anybody into USB Mics?

[mayfly]

Anyone use or have opinions of USB mics for recording demos?

I'm having fun writing songs in garage band on my mac, but need a high quality (i.e. recording studio quality - not PA quality) USB mic to use with it.

Any advice?

 

[Soylent]

my roommate has done some recording with a Blue usb mic..

Sounds pretty good.  :headbang:

 

[tfarny]

I would still look for a toneport on Ebay and a real condenser mic, like an AKG perception or Rode NT1. The Toneport connects via USB, has a bunch of vocal preamps built in to software, effects to play with, and phantom power for a real mic.
Plus it's real useful for guitars and basses, what it was designed for.

 

[nathan a]

Most of the recording I've ever done (which is a large amount of demo-type stuff) has been done with the Samson CO1U, and more recently CO3U, run through Audacity.

For the money and ease of use, I think I get great sounds.

 

[Superlizard]

Mics aside, any DAW recording, you're gonna want a unit (mic pre / etc) with A/D converters that can do at least 24-bit, 44kHz.

If you don't have that minimum spec, it won't matter even if you plunk down a grand for a mic - your recordings will sound like @$$.

 

[bpmorton777]

SL...I dont think that is necessarily true. Ive been recording since 4-track cassette and have to say that 16 bit 44 sounds WAY better than any of my cassette recordings and the recordings I made on the Tascam 388 I used to own. for the kind of stuff where all the tracks are real loud..as in a guitar through a marshall and drums you wouldnt be able to here any of the extra bits you get from 24...it matters when you are recording very quiet stuff like violins on a condenser mic.

Brian

 

[AndyG]

SL...I dont think that is necessarily true. Ive been recording since 4-track cassette and have to say that 16 bit 44 sounds WAY better than any of my cassette recordings and the recordings I made on the Tascam 388 I used to own. for the kind of stuff where all the tracks are real loud..as in a guitar through a marshall and drums you wouldnt be able to here any of the extra bits you get from 24...it matters when you are recording very quiet stuff like violins on a condenser mic.

Brian

Agreed ... and home recording has a lot less to do with hardware/software and much more to do with the operator.  Take an engineer like Randy Staub or Paul Northfield, and give them a cheap DAW and they will get better sounding recordings than your average basement engineer would in an SSL-equipped studio.

At the end of the day, what you want to do with your recording will determine how "good" it needs to sound.  I remember back in the 80's, bands were selling 4-track demos, recorded then re-dubbed to another cassette, at the clubs after the show.  If you're shopping material to record companies, they'll probably want to re-record it anyway.  If you're selling your own music online, well, most people will put up with 128 kbs mp3's 'cause they want to have 1000 songs on their iPod, rather than the 200-300 you could get at 320 kbs.

I personally have very little experience with USB mics.  If it captures the sound you want, and doesn't clip or distort too quickly, by all means, use it.  Will it replace a $2500 U87?  I doubt it.  The same way the home DAW will never replace the SSL studio.  But, like anything else in our digital world, if it works for you, that's all that matters.  The ONLY caveat I will bring up is if you ever plan on recording multiple things at the same time, don't bother with the USB mic, as I believe you can only have 1 plugged in at a time.  Get a small mixer and a few microphones instead.

 

[Superlizard]

SL...I dont think that is necessarily true. Ive been recording since 4-track cassette and have to say that 16 bit 44 sounds WAY better than any of my cassette recordings and the recordings I made on the Tascam 388 I used to own. for the kind of stuff where all the tracks are real loud..as in a guitar through a marshall and drums you wouldnt be able to here any of the extra bits you get from 24...it matters when you are recording very quiet stuff like violins on a condenser mic.

Brian

Here's a well-written (if not a bit wordy) sampling tutorial:

http://www.samplecraze.com/tutorial.php?xTutorialID=8

In essence, the reason to use 24-bit (or above):

Back to bits. The most important aspect of bits is it’s resolution. Let me explain this in simpler terms. You often come across samplers that are 8 bit, Fairlight CMI or Emulator 11, or 12 bit, Akai S950 or Emu SP1200, or 16 bit, Akai S1000 or Emulator 111 etc..You also come across sound cards that have 16 bit or 24 bit etc…Each bit refers to how accurately a sound can be recorded and presented. The more bits you have (Resolution), the better the representation of the sound. I could go into the’ electrical pressure measurement at an instant’ definition but that won’t help you at this early stage of this tutorial. So, I will give a little simple info about bit resolution.

There is a measurement that you can use, albeit not clearcut but at least it works for our purposes. For every bit you get 6dBs of accurate representation. So, an 8 bit sampler will give you 48dB of dynamic range. Bearing in mind that we can, on average, hear up to 120dB, that figure of 48dB looks a bit poor. So, we invented 16 bit cd quality which gives us 96dB dynamic range. Now we have 24 or even 32 bit sound card and samplers (24 bit) which gives us an even higher dynamic range. Even though we will never use that range, as our ears would implode, it is good to have a bit. Why? Well, use the Ferrari analogy. You have 160mph car there and even though you know you are not going to stretch it to that limit (I would), you do know that to get to 60mph it takes very little time and does not stress the car. The same analogy can be applied to monitors (speakers), the more dynamic range you have the better the sound representation at lower levels.

By only going 16bit, you aren't getting the "full monty"; as the sampler isn't picking up quite everything (accurate representation).  i.e. - if we can hear up to 120dB, 16bit (96dB) isn't quite enough - there's stuff missing.

I used to record at 16bit back in the day... now I do 24bit, 44kHz minimum and sometimes 24bit, 192kHz (my A/D's limit) - I can hear the clarity difference.

EDIT:  Anybody who uses some form of A/D converter to record should know these basics if they want to have full command over what they're doing... if you care about your guitar tone - because your guitar tone is being picked up by the mic (or DI), and sent thru the A/D converter.

 

[Superlizard]

If you're selling your own music online, well, most people will put up with 128 kbs mp3's 'cause they want to have 1000 songs on their iPod, rather than the 200-300 you could get at 320 kbs.

As well, most people (esp. non-musicians) can't hear the awful quality of 128kbps... let alone tell the huge clarity difference 'tween 128kbps and 320kbps.

Still, I wouldn't use those facts to justify having myself sound like @$$.

 

[AndyG]

SL...I dont think that is necessarily true. Ive been recording since 4-track cassette and have to say that 16 bit 44 sounds WAY better than any of my cassette recordings and the recordings I made on the Tascam 388 I used to own. for the kind of stuff where all the tracks are real loud..as in a guitar through a marshall and drums you wouldnt be able to here any of the extra bits you get from 24...it matters when you are recording very quiet stuff like violins on a condenser mic.

Brian

Here's a well-written (if not a bit wordy) sampling tutorial:

http://www.samplecraze.com/tutorial.php?xTutorialID=8

In essence, the reason to use 24-bit (or above):

Back to bits. The most important aspect of bits is it’s resolution. Let me explain this in simpler terms. You often come across samplers that are 8 bit, Fairlight CMI or Emulator 11, or 12 bit, Akai S950 or Emu SP1200, or 16 bit, Akai S1000 or Emulator 111 etc..You also come across sound cards that have 16 bit or 24 bit etc…Each bit refers to how accurately a sound can be recorded and presented. The more bits you have (Resolution), the better the representation of the sound. I could go into the’ electrical pressure measurement at an instant’ definition but that won’t help you at this early stage of this tutorial. So, I will give a little simple info about bit resolution.

There is a measurement that you can use, albeit not clearcut but at least it works for our purposes. For every bit you get 6dBs of accurate representation. So, an 8 bit sampler will give you 48dB of dynamic range. Bearing in mind that we can, on average, hear up to 120dB, that figure of 48dB looks a bit poor. So, we invented 16 bit cd quality which gives us 96dB dynamic range. Now we have 24 or even 32 bit sound card and samplers (24 bit) which gives us an even higher dynamic range. Even though we will never use that range, as our ears would implode, it is good to have a bit. Why? Well, use the Ferrari analogy. You have 160mph car there and even though you know you are not going to stretch it to that limit (I would), you do know that to get to 60mph it takes very little time and does not stress the car. The same analogy can be applied to monitors (speakers), the more dynamic range you have the better the sound representation at lower levels.

By only going 16bit, you aren't getting the "full monty"; as the sampler isn't picking up quite everything (accurate representation).  i.e. - if we can hear up to 120dB, 16bit (96dB) isn't quite enough - there's stuff missing.

I used to record at 16bit back in the day... now I do 24bit, 44kHz minimum and sometimes 24bit, 192kHz (my A/D's limit) - I can hear the clarity difference.

EDIT:  Anybody who uses some form of A/D converter to record should know these basics if they want to have full command over what they're doing... if you care about your guitar tone - because your guitar tone is being picked up by the mic (or DI), and sent thru the A/D converter.

While your end result is correct, your theory is wrong.

Humans do not "hear" 120 dB.  In fact, 120 dB can be quite painful.  We hear up to about 22kHz (in reality only about 16 kHz, but they say we can "percieve" up to 22kHz).  The difference between 24 bit (120dB), and 16 bit (96dB) is the noise floor.  24 bit is quieter, not louder, hence the greater dynamic range.  The decibels in question relate to voltages, not SPL (volume).  Digital zero is an absolute limit ... going over means digital distortion ... and 16 bit digital zero is the same as 24 bit digital zero.
Sampling rates will make the difference.  at 192kHz, you can sample up to 96kHz.  Dogs can't even hear that high, but that's where we humans get the perception of "air" and "clarity".  CD's have been 16 bit, 44.1 kHz forever, and it is only recently that pro audio gear has gone the 24/192 route.  At the end of the day, the noise floor from your guitar amp, or the air conditioning in the room the drums are in, will be greater than the difference in 24 vs 16 bit.  Where it does make a difference (as bpmorton777 pointed out) is when things get quiet.  Hit the lowest note of a piano, and allow it to fade out, and you'll hear the difference between 16 and 24 bit.

 

[Soylent]

That being said...

tfarny's idea seems pretty good. :laughing7:

 

[Superlizard]

While your end result is correct, your theory is wrong.

Humans do not "hear" 120 dB.  In fact, 120 dB can be quite painful.  We hear up to about 22kHz (in reality only about 16 kHz, but they say we can "percieve" up to 22kHz).  The difference between 24 bit (120dB), and 16 bit (96dB) is the noise floor.  24 bit is quieter, not louder, hence the greater dynamic range.  The decibels in question relate to voltages, not SPL (volume).  Digital zero is an absolute limit ... going over means digital distortion ... and 16 bit digital zero is the same as 24 bit digital zero.
Sampling rates will make the difference.  at 192kHz, you can sample up to 96kHz.  Dogs can't even hear that high, but that's where we humans get the perception of "air" and "clarity".  CD's have been 16 bit, 44.1 kHz forever, and it is only recently that pro audio gear has gone the 24/192 route.  At the end of the day, the noise floor from your guitar amp, or the air conditioning in the room the drums are in, will be greater than the difference in 24 vs 16 bit.  Where it does make a difference (as bpmorton777 pointed out) is when things get quiet.  Hit the lowest note of a piano, and allow it to fade out, and you'll hear the difference between 16 and 24 bit.

Regardless, 24bit is superior to 16bit in dynamic range... so why wouldn't you want your A/D chip to pick up as much of the recorded signal, and in turn, be as accurate and faithful to the sound being recorded as possible?

And if you're a lo-fi junkie, there are plenty of VSTs/plugins you can apply to the recording (tape / tape saturation emulators, etc) after the fact.

My main point in all this being:  don't sell yourself short.

 

[tfarny]

That being said...

tfarny's idea seems pretty good. :laughing7:

Yay I win again!  :redflag:

 

[bpmorton777]

$$$ simply put. I and everyone else here has sterio 16 bit 44k sampling built into their home computer. I use macs so im not sure on how the pc stuff stacks up with built in audio. Ive gotten good results just plugging into my G5's audio inputs. (relative to my old gear).

btw, Ive owned and used disgital samplers since 86' and know all about what the various sample settings will do to the sound. Ensoniq Mirage and EPS user.

Tone port sounds like a good idea. Im not too crazy about USB mics only because Ive heard about the lagg you get going through USB. (never actualy experienced it for myself )

Brian

 

[tfarny]

A toneport or similar DAW functions as a sound card and can do 24 bit 96khz out of the box. My cakewalk sonar can record at that same rate too.

 

[JaySwear]

i'd doing the same... i ended up using a Rockband mic from a Xbox 360 game, which i highly recommend by the way.  :icon_biggrin:  kidding, i'm interested in this too. i'm heard great things about the blue mic, but can't read the rest at the moment.

 

[mayfly]

Hi Folks,

Thanks for all the replies so far.  Allow me to focus the discussion a bit:

This is to be used for demos recorded on a laptop (macbook Pro).  Portability is important, so I don't want any additional boxes around. I want to be able to go to our singer's house, set up a mic in her living room, put the mac on the couch and run through some vocal takes.

Sound quality should be good - but when I'm talking good I mean better than an SM58.  When it's time to record it for real, we'll be heading to the studio.  Remember, right now I'm using the crappy built-in mics  

Latency is something that I hadn't considered, but it's VERY important that the mic not have much.  I'm multi-tracking here and I can't abide by the evil latency.  If all USB mics exhibit this, then the USB mic option is a complete writeoff and I'm back using the average quality ADC and DACs on the laptop.  Anybody used a USB mic for multi-track recording?

oh - and the current options that I've looked into are the Sampson line and the Blue snowball  :icon_biggrin:

 

[tfarny]

Get a toneport for that situation, it's just a little box that gets its power from the USB. I use a laptop on battery power and the toneport with no probs. Lets you use any mic you want.

 

[Superlizard]

$$$ simply put. I and everyone else here has sterio 16 bit 44k sampling built into their home computer.

Myself, I've owned an EMU 1820m since '04-'05 (max 24bit, 192kHz - same converters as the professional ProTools systems) as my DAW hardware.

Not to be nit-picky, but for factual reasons, it should be noted that today's typical home computer (Apple or PC)
built-in soundcard standards for computer sounds (i.e. games, beeps etc) call for at least 24bit, 44kHz quality.  

We're not even talking about DAW specs there.

Most of these cards are capable of 24bit, 96kHz.  As well, many, if not most soundcards can do 5.1 audio - many 7.1 audio.

The days of 16bit, 44kHz being the 'puter soundcard industry standard have been gone for good for at least 5 years.

 

[Superlizard]

More info (for those who like the finer details):

Nyquist Theory:

http://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem

Quote (separate source):

The sampling rate is measured as a frequency (please see Synthesizer Tutorial part1) and is termed as kHz, k=1000 and Hz= cycles per second. These samples are measured at discrete intervals of time. The length of these intervals is governed by the Nyquist Theory. The theory states that the sampling frequency must be greater than twice the highest frequency of the input signal in order to be able to reconstruct the original perfectly from the sampled version. Another way of explaining this theory is that the maximum frequency that can be recorded with a set sample rate must be half the sample rate. A good example at this point would be the industry standard cd. 44.1 kHz means that the number of times a sample (snapshot) per second is taken equates to 44,100/second.