I’ve spent the last little while quality checking my work. I won’t go into details in this post, but I have a workspace which is set up on a great computer, using a Behringer Xenyx 1208 FX Mixer Board and a Behringer C-1 Cardioid Microphone. These two pieces of equipment have served me well, but I am just now having issues getting the quality of the room tone to behave.
On average, my recordings, after creating a curtained space from which to work, are on average (-56) to (-60.5). As anyone more experienced than me would probably note, this is not desirable. My computer is outside of my curtained recording space, and the fan noise is barely perceptible through the curtains. You have to be REALLY listening. Perhaps that won’t do.
I’ve tried a USB soundcard and direct motherboard input for recording, not much difference. I’m speaking directly into the microphone for the most part. Any ideas?
I’m including a picture of my workspace and sound samples as per the recommendations-- two 10 Second “Catskill Farms” samples, one pass, one fail, the only difference being, I think, relative vocal volume. The fail would be more normal for me.
Well, you lucky dog. Your sound files will open in the current Audacity 2.4.2, but not in the exciting, new, Audacity 3.0 (under test). That’s concerning.
Do you regularly have Audacity crash on you?
I’ll rip the studio apart a bit later, but your submitted samples are insanely noisy, and it’s not (that I can tell) air conditioning or computer fan noises. It’s regular, rain in the trees, electronic noise (fffffff).
Behringer Xenyx 1208 FX Mixer Board
There doesn’t appear to be a 1208 mixer board.
direct motherboard input
How? You can’t plug a C-1 directly into a computer. If for no other good reason, the C-1 needs 48 volt phantom power and the computer only supplies 5 volts.
Don’t fall in love with the ACX -60dB noise specification. If you can’t reach -65dB noise, keep trying.
Does your microphone look a lot like that?
Are you speaking into the side grill just up from the company name (assuming it’s right-side up)? Are you about a Hawaiian Shaka away?
I always do that with the 1202 FX. My brain goes straight to “1208,” I cannot explain how many times I’ve done that.
Yes, my mic is the exact same one as in the picture with the shaka. I’ve been using the 1202 FX with it, plugged into a Behringer UCA 222 USB Soundcard for 6 years for various projects. It has always served me well, with a few exceptions over time.
I’ve tried seeing how much of a different it makes to have another cardioid mic plugged in, particularly the Samson C01. @Trebor: Having either mic plugged in seems to result in similar floor noise, and unplugging the mic while leaving all mixer-board levels the same results in about a -10 drop in noise floor from the baseline with no tweaks.
I’m actually shooting for -65 here, absolutely.
What I mean with “direct input” is that I’m plugging right into the motherboard from the 1202 FX, rather than into one of my two UCA 222.
So far as Audacity, that’s so odd, I quite literally only just updated to the new audacity. If you recommend the Beta of the 3.0, I’ll try it. No crashing though. Generally audacity is well behaved for me.
With all that being said, what do either of you think?
I’ve just conducted a bunch of tests, including the monitor test you described, as well as using separate power supplies for my mixer board. I have two sets in the house. Swapping the power supplies did nothing to mitigate the sound problems, and as far as I can tell from the monitor test, no difference was made (I also did the same test with the monitors unplugged, to varying but equally unsatisfying results.)
There are four light arrays in the room, all four containing 3 led lightbulbs. On or off, there does not seem to be an appreciable difference.
I’m willing to go through some seriously silly non-sense to get this working, so let me know what you might be thinking. Gain adjustments don’t seem to help; I have my best values at about the second notch or a quarter turn from the lowest setting. I have my system microphone gain set to 77, a value which seems to be best suited to the task at hand.
That is the likely case, I’m afraid. I broke out my Tascam US-16x08 and finicked enough with it to get it to record to Audacity. It’s definitely a better recording, but it didn’t necessarily improve enough. When I bought the C-1, it was $60.00, if I recall. It’s old now.
Would you guys recommend anything before I start recording and using tons of filters to mask the problem? Already here.
My Audacity layout is a little wacky, but it’s handy if anything goes wrong.
I pull the meters across the whole Audacity window. The left and right extremes of the meter are drag and positioning clickies. As you drag the meters around, the other windows should move out of the way. I change the lower (quiet) limit from the default -60dB to the maximum for 16-bit recording, -96dB.
Yes, the red one. I like the passthrough device they have for listening back to your audio input.
I’ve connected the 222 via Tape-Out and sent the master Fader all the way down. The result was a constant floor of about (-69)-(-66) at .77 system gain.
Input controls such as system gain continued to create a response. I think I may have misunderstood you here. The control is that at 0.00 system gain, I receive an invariable floor of -90.
You’re teaching me quite a bit here, I appreciate it.
I was able to apply the same line of thought to my Tascam US-16x08 device.
It has a software EQ panel, and with those settings turned on for one channel, you can see the results in contrast. With the EQ panel turned on for the left channel and left off for the right, I get results of about -69 on the left (The channel with the EQ on,) with little variation. The opposing channel (right channel, EQ off) is much more variable, but generally lower with a floor of about (-78)-(-74).
Recordings from either channel, or setting rather(It’s the EQ panel that creates the higher noise floor) produce about the same results-- about (-59)-(-61) after Filter Curve and Noise Normalization; Frankly I seem to get only just slightly worse results from the Mixerboard at 0 db on the main fader, all Audacity settings remaining the same-- about -59 or so, sometimes less sometimes more.
I think I’ll look into a solution there-- in the mean time, I think I’m getting somewhere by switching out to the Tascam, since it has a USB interface. I saw some recommendations for the Audio Technica 2050. I’ll do more research before throwing money out the door, maybe rent a microphone as you suggested.
After a fair amount of research, I’ve decided to replace my microphone. I tested the Samson C01 again, and found that I was getting (-3)-(-5)db more on average in recordings just from using something a little more expensive, and I was passing ACX organically, after curve and normalization within these levels. I’ve got my eyes on two different microphones to bring myself just over the top:
I feel like I’m on the home stretch here guys, what are your honest opinions; am I overlooking something somewhere in the data? With just another 9db on the s/n ratio, I should be able to clear -65db every time in optimal conditions, yeah?
Microphones don’t have a S/N spec because the signal is variable. They have a “self noise equivalent”. I checked the Audio Technica spec and it says 12dB SPL which is quieter than a “soundproof” recording studio.
Dynamic mics don’t have a noise spec because there are no internal active electronics. They do generate something called “thermal noise” but it’s insignificant compared to acoustic noise and preamp noise.
A “hotter” mic (with the same self noise) will give you a better electronic S/N because the signal is higher relative to the preamp noise.
Microphone sensitivity doesn’t make any difference with acoustic S/N… A more sensitive mic will pick-up more room noise but it also picks-up more signal. A directional mic helps because the signal only comes from one direction whereas the noise comes from all directions…
You guys have been great. With your help, and some memory jogging, I’ve got it down to an amateur science and I’m seeing consistent sub-65db recordings in post with my new AT-2035; this was a game changer for me.
Tomorrow, I’ll be getting in a fan and thermal heatsink which is about 20db quieter for my CPU, and I think everything will be grand. Meanwhile, I’ll do some more research and see if I should be using the fancy low cut and -10db pad switches on the AT-2035.
see if I should be using the fancy low cut and -10db pad switches on the AT-2035.
The -10dB pad is to keep the microphone from damaging very loud sounds such as playing your bugle into it.
The low cut filter does the same thing as the first step in Mastering. Effect > Filter Curve > Low Rolloff for Speech. It’s a rumble, thump, and wind filter. That software step is required to make the rest of the mastering suite work, but the one in the microphone is optional. It can help with P-Popping, but it can take some of the fullness and sultry punch out of your voice. I’d leave them both alone.
Stick to the spacing recommendations and note a pop and blast filter is a good idea if you’re prone to popping.
This is a sound test from someone getting used to using his microphone and he has award-winning P-Popping.
Listen on a big sound system. Some of those Ps will knock your wine glass over.
A word on noise. Once you can reliably get your uncorrected noise down in the -65dB range, then you can apply Noise Reduction of the Beast. Drag-select clean room tone. Effect > Noise Reduction > Profile. Select the whole piece. Effect > Noise Reduction 6, 6, 6 > OK. The result should sound exactly like you but with a significant drop in background noise.