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Question: what kind of bass amp are you using?
I've been doing a lot of recording onto my PC... why should I be recording at 96/24 in the first place?
Ben, what is a "patch bay" and how does it work?
I understand that phasing problems can occur when signals from two microphones placed some distance apart capture sounds from the same source.
Ben, where do you recommend for the best falafels in the Bay Area?
What (recording) set-up would you recommend to get the best results for the least amount of money.


Tue, 17 Jul 2001 09:52:52
from: wearethedavies@yahoo.com
Question: what kind of bass amp are you using? I really like the sound!

Prof. Ben's Response
Currently I'm using a Gallien Krueger 800RB (300w + 100w) going into a no name 2-10 with JBL speakers, and a no name 1-15 with a complete piece of shit speaker. I hope I blow it in the next few months. The knobs are run more or less as follows: input volume - 2 o'clock, mid shift button in, high - 1 o'clock, high mid - 1 o'clock, low mid - 11 0'clock, low - 1 to 2 o'clock, boost - 9 o'clock-ish, crossover - 11 o'clock, crossover engaged, and master volumes to taste depending on room size. If something doesn't sound right at a show I have no problems tweaking it.

Before that, I had a Yamaha 100B (100w) driving the 2-10, and the line out from the 100B going to a 200W power amp powering the 1-15. I used a Sans-Amp to shape the tone. That pedal is great, but it makes me feel like I'm cheating. I still have the Yamaha Head and it's for sale. It's a little flakey, so I'll let it go for $75.

We also must not forget the extremely cheap Dan-Electro bass with it's series-wired pickups, and the TS-7 Tube Screamer (modded to TS-808 specs plus with added low end).

If you speak of the CD, Dan Kennedy played on the recording. He also used an 800 RB, but he ran his into a 4-10 cabinet and had an Ibanez bass (spanish name, japanese made, there's a funny story behind that).

Next Question!

*****

Fri, 20 Jul 2001 01:10:31
from: dank@lunar.com
First off, Ben, congrats on the Professorship! Are you on the tenure track?
Second - I've been doing a lot of recording onto my PC's hard disk via an Echo Darla 96/24 soundcard. I've been eating up tons of disk space by recording exclusively at 96KHz sample rate and at 24 bit resolution. But I plan to render everything down to 44.1/16 anyway so that I can burn my work onto CD's. So the question is - why should I be recording at 96/24 in the first place? I've asked the software people (Sonic Foundry), but they never answer my e-mails.


Hello Dan, (this is the former bassist from Replicator who also uses a GK 800 RB). To the first question, I'm an hourly professor.

Second, yes, that will eat up a lot of space as it will use more than twice the data as normal CD quality audio. Before I answer you question, le me explain sampling rate and bit rate.

When recording sound digitally, an analog-to-digital converter takes the analog sound and cuts it up into tiny fixed segments. The sampling rate is how many segments per second the sound is sliced, and the bit rate is how many bits of information each slice contains. The larger the bit size, the more resolution between maximum level and silence. CD audio is 44.1 thousand samples per second, and 16 bits per sample. Since computers work in binary, this is 2 to the 16th power, or 65,536, which means there are 65,536 volume levels of audio between overload and the noise floor, or silence, depending on how you look at it.

Sound is naturally a wave, like as ocean wave, or a sine wave, or an arm waving goodbye like a pendulum. It has smooth crests and valleys. When sound is converted to digital, the wave is converted into something comparable to a staircase. A CD has 44.1 thousand steps every second, and the height of those steps MUST be precisely on one of the 65,536 volume levels evenly distributed between maximum level and silence. Obviously, this isn't an identical representation, but it works well enough for most consumer uses.

Now to Dan's set up. Recording at 96 thousand samples per second makes the distance between the "steps" much closer, and instead of 65,536 volume levels at 16 bit, there are 16,777,216 volume levels between maximum and minimum, which allows much more accurate level representation.

What I've heard is that when you mix digitally, the higher resolution signals combine in a more audibly pleasing manner than the lower resolution signals. I've heard that recording at 96/24, mixing to 96/24, then dithering down to 44.1/16 to burn to a CD will produce a nicer sound that dithering each track down to 44.1/16 and THEN mixing digitally. I've never done this personally, though.

One thing I'm unsure of, though, is how the bitrate translates to dB (decibels) of audio range. 1,111,111,111,111,111 is the loudest volume in 16 bit binary, and 111,111,111,111,111,111,111,111 is the loudest volume in 24 bit binary. Turn those ones into zeros and you have silence. I've heard that when you sample down, you digitally throw away the numbers at the right, which in my head means that where there was one possible level difference in 16 bit, you now have 256 possible level differences. I've ALSO heard that 24 bit gives you more dB range, but how can zero be any quieter than zero. My roommate Marvin, a professional mastering engineer, has completed a first draft of a book on digital audio, and I'm one of the readers for revisions. I'm sure this is explained in the book, so when I find out, I'll post it here.

Correction: I talked to Marvin, my mastering engineer room-mate. I made a mistake, 0,000,000,000 is not digital black, or silence, it the sound wave maxed at the opposite phase. 1,111,111,111 is max level for peaks, and 0,000,000,000 is max level for troughs. Halfway in between is silence. Marvin says that the gain from going from 16 to 24 bit is much better that going to from 44.1k to 96k. He said that if he was recording an album he'd do it at 44.1k/24 bits, and do all the effects processing at 24 bits when using plug-ins. Then mix down at 24 bits. If you're getting it professionally mastered, get the mastering house these 24 bit files, so that mastering can be done at 24 bit. The dither to 16 bit should be the last step in the process, ideally.

*****

Mon, 30 Jul 2001 19:31:18 -0700 (PDT)
from: dank@lunar.com

Ben, what is a "patch bay" and how does it work? Is it just a bunch of TRS jacks wired together for the sake of convenience? Why is everybody telling me that I should get one for my home studio?

A patchbay is a bay where you patch audio connections. If you have a fair amount of gear, these can make sense.

For the consumer, a patch bay is a rack unit with a bunch of TS or TRS jacks on the front, each with a corresponding jack on the back. In my studio, it's the top piece of gear in my effects rack. All the inputs and outputs of my two stereo compressors, my digital processor, my guitar processor, and my stereo EQ hook up to the back of the patchbay. Now, when I need to hook up to any of these units, I can plug right into the front of the rack to the corresponding jack for the input and output routing of my gear. Remember, most rack gear has the inputs and outputs on the back side anyway, so this not only organizes the inputs and outputs to one face, it eliminates having to lean around behind gear.

In pro studios, the patchbay is where everything comes together. The connections are actually soldered on the back. Every input, output, and insert from the console goes to this patchbay, as well as the inputs and outputs from the tape machines, and all the outboard gear. The patchbays usually use these mini-jacks with smaller cables, so you can fit more patches into a smaller area. It makes things very easy, but it can also become a rat's nest really quick.

Personally, I found I didn't need a patchbay until I got more gear than I could leave hooked up at once. When I had one stereo compressor and one reverb, I just left them hooked up all the time, with the reverb in an aux send and return, and the compressor's insert cables resting on the back of the board. I don't know how much outboard gear you have, but if it's not much, or if you don't keep it in a rack, then I don't think a patchbay is necessary yet. if you have a fair amount, and have a rack,B>I understand that phasing problems can occur when signals from two microphones placed some distance apart capture sounds from the same source. Does this necessarily mean that the signals from a room mic and a close mic (for one particular event) should be always be panned to opposite speakers in a final mix? If the signals both go to the same speaker, won't some signal cancelling occur?
patchbays can be very handy.

*****

Tue, 31 Jul 2001 18:29:35 -0700 (PDT)
from: dank@lunar.com


Phasing can be a serious problem. What happens is that two microphones, placed at different locations, receive the sound at slightly different times. In certain frequencies the peak on one sound wave can arrive at one mic when the trough of the same arrives at another mic. The signal will then electrically cancel itself out in the mixer. It can also be very easy to correct and sometimes even useful or interesting sounding.

There are a few conditions where this can be extreme.

1. Matching drum overheads. If your's using stereo drum overheads, and the mics aren't in the same place, it's a good idea to make sure the mics are equidistant from the snare. If the mics are a few inches different, you can cancel out a lot of the crack of the snare. What good are overheads without a solid snare crack?

2. Matching mics in general. All mics have different frequency responses. That means they pick up certain frequencies more efficiently than others. Some mics excel at picking up high frequencies, other pick up low frequencies well. When you have identical frequency responses in two mics, they will either reinforce that response, or work against that response. That's why combining two different mics is sonically interesting, you can put a mic with solid low end in front of a guitar amp to get a solid direct chunk, and a mic with good high end in the room to get bright room reflections, and pick up a more diffused, less piercing high end. Any phase problems I've had with two different mics has actually sounded interesting, and was a viable sound for recording, IMO. This, however doesn't mean you shouldn't use matched pair mics BUT...

3. If you use omni-directional mics for room mics, don't put them too close together, or put something physically between them. I forget the speed of sound, but it's roughly 500 ft/s (feet per second). Sound is measured in Hz (hertz), which is also known as C/s (cycles per second). If you invert the frequency value to end up with s/C (seconds per cycle), and then multiply by the speed of sound (which is a value of ft/second), you'll get feet/cycle, which is wavelength. So a 500 Hz sound wave is 500 c/s, or 1/500 s/c. Multiply that by 500 ft/s and you get 1 ft/c, or a wavelength of 1 foot. Half of that wave length is 6 inches. Since one cycle is 360 degrees of phase, 1/2 the cycle would make it 180 degrees. So two omnidirectional mics placed 6 inches apart would do a good job cancelling the 500hz frequencies.

Actually, this can be used to your advantage. Record a room with one omni mic. You might hear a boominess in the low end from standing waves. With an EQ, find the frequency that the boominess resides at, figure out the wavelength from the formula above, divide that by 2, and put the mics that distant apart to use the phase cancellation to help reduce the artificial boost from room acoustics (all calculations are approx. as i'm not using an accurate value for speed of sound).

When it comes to phase, always test in mono. Use your ears. Some people like slightly phased sounds, and you might too. In most every case, any phase problems can be corrected by moving a mic a couple inches or a couple feet.

*****

Mon, 6 Aug 2001 02:23:51 -0700 (PDT)
from: phrancis@dasdigital.com
Ben, where do you recommend for the best falafels in the Bay Area, because the sound of your bass rig makes me hungry for chickpeas.

I've not had much falafel, but I highly recommend La Mediterrain for Mediterranean food.

*****

Mon, 6 Aug 2001 21:27:49 -0700 (PDT)
from: dancerinthedark@bjork.com
This is not too techincal a question. My band (four piece, shoegazer/indie pop) has wanted to record for sometime, but have been unable to invest the money. Recently, the bass player had a small windfall of cash, and wants to dedicate it towards recording, manufacturing, and promoting a record. We have eighteen songs, and have been together for over four years. We have about $5,000 to spend for EVERYTHING. I am under the assumption that we could afford to buy some gear and record whenever we like. It may not be as "professional" sounding as some of the recordings out there, but we would be able to learn for ourselves how to use the stuff, and we could use it to record other people as well. Is this possible? If so, what set-up would you recommend to get the best results for the least amount of money. I know very little about recording, but my bass player has been in bands that have used studios before. I really want us to get the most for our cash since we have worked so long! and hard for this.

Home recording is getting easier and more acceptable as time passes. There are many great albums that were recorded at someones home studio. Songs for a Dead Pilot by Low, The Long Arm of Coincidence by Jessamine, and countless others have been recorded at homes or practice spaces, all with modest gear.

In today's age, you have three options.
1. Computer gear.
2. All in one units.
3. Modular traditional setups.
Let's go over these.

Computer gear: perhaps the easiest way to get into recording nowadays is with a modern computer with a good soundcard. A computer can take the place of a recorder, mixer, and effects. Number of tracks is only limited by processor speed. What you still need to buy is microphones and something with mic preamps, either outboard units, or a mixer with many built in.

Advantages: not too difficult to use, clean sounds, easy editing, can record, mix, master, burn all on the same unit.
Disadvantages: quite an expense to record a lot of tracks at once, annoying to make backups of individual tracks, mixing on a computer feels funny.

All in on units: in the 80's portastudios were invented. in one box you had a mixer and a 4 track recorder that used cassette tapes. And 8 track version was mad for a while. As digital technology grew, this format of mixer/recorder in one small desktop unit was developed into a much more powerful tool. Nowadays, for 1000-2000 dollars, you can buy a 16 or 24 track digital recorder that sounds clean, has some built in effects, and is very easy to use. I does everything a computer does, but doesn't have to be built around a mac or windows OS, so they're much faster and more reliable. These usually have 2 or 4 mic preamps, but if you want to record a full band live, you'll probably need to get more.

Advantages: Clean sound, easy to transport, easy to use.
Disadvantages: Limited number of tracks recordable at once, not very expandable, hard to back up multitrack information.

Modular Traditional Setups: these are what you think of when you tend to think of a recording studio. You have a multi-track recorder, a mixing board or console, mics, cables, outboard gear like compressors, preamps, reverb, effects. You can get gear that works for your setup, and you hook all this gear together however you like to make it work best for you. You can start small and the studio can grow one piece at a time. price usually reflects qualtiy.

Advantages: Extremely flexable and scaleable. Easiest way to work with analog. Can ultimately provide (IMO) the best sound.
Disadvantages: Can get pricey, hard to move around.

My biggest pieces of advice are A. don't skimp on the mics, and B. buy used! I started out with and analog 8 track and an 8 channel board. It taught me a lot. I still have that 8 track, as well as 16 tracks of ADAT, a 16 channel board, a small number of nice mics, and a few pieces of effective outboard gear. I much prefer setup 3, modular and traditional.