AfaraWayland

For the time being

2011-11-07T17:15:00.000-08:00

AfaraWayland will be shifting back towards Music Production in the near future.

Ramsey Vintage Radio Battery Eliminator Kit

2010-01-03T21:39:00.000-08:00

Recently a man asked for advice on how to resurrect a vintage piece of audio equipment. The device in question was a Sound Scriber, an item I am partially familiar with. One of his concerns was how to supply power to the unit. Unfortunately, there doesn't seem to be much information about the power requirements of the Sound Scriber, or (more importantly) circuit schematics on the internet.

A few weeks after our conversations I received Ramsey Electronics 2010 catalog. Among the vast selection of electronics kits I came across Ramsey's new Vintage Radio Battery Eliminator Power Supply Kit. Though the battery eliminator kit may not help the gentleman I spoke with, it is certainly interesting.

This power supply is similar to other battery replacement options available from other manufacturers. The distinguishing characteristic is that Ramsey has made this type of device into a DIY kit.

Ramsey describes the kit best:

Many classic radios operated on batteries only, and in many cases a series of three batteries for each radio were required! The A battery provided power to the filament of a vacuum tube, the B battery provided the plate voltage for the tube, and the C battery provided bias to the control grid of the tube. The new ABCE1 Battery Eliminator gives you an easy way to replace all these batteries with a simple household AC power connection (choose between 120VAC or 240VAC with a simple jumper selector) and resurrect your vintage antique radios! The ABCE1 provides all the required operating voltages, and all supplies are isolated from each other and input line voltage. This allows easy interconnection for virtually any required configuration.

Searching a little farther I found some technical information:

"A" Filament Supply: Adjustable from 1.25VDC to 6.5VDC @ 2.5A max
* "B" Plate Supply: 22VDC, 45VDC, 67VDC, 90VDC, 135VDC fixed outputs @ maximum of 60ma total for all loads.
"C" Bias Supply: -3VDC, -4.5VDC, -9VDC, -22VDC fixed outputs @ maximum of 20ma total for all loads.
* The "A" filament supply + voltage is bonded to the AC input line equipment ground.

--------------------------------------------------------------

This is the ARBE-III Universal Battery Eliminator made by Antique Radios Inc. It is yet another radio battery-replacement option, but unlike the Ramsey kit it arrives pre-assembled. The ARBE-III is even 40 dollars cheaper than the Ramsey kit.

The ARBE site states:

ARBE-III is a solid state, fully regulated, universal power supply designed specifically for use with pre 1930's battery operated radios. The latest version is now available with added improvements. We have added a new black anodized heat sink which improves the power handling capability. Notice also that a red neon lamp has been added to show when the unit is turned on. Three electrically isolated power sources are provided in one self contained unit. A custom wound power transformer provides maximum flexibility for the power requirements for many different types of older radios. The B & C voltages are regulated and short circuit proof. The unit comes fully assembled, tested and is warranted for five (5) full years from date of purchase. Bring that Atwater Kent, Crosley, Federal, Kennedy, RCA or other early battery radio back to life again. Comes complete with owner's manual, warranty, and wiring diagrams showing how to make the connections for many of the early battery operated radios.

Voltage specs:
"A" adjustable from 1.25 to 6.5 VDC and is capable of providing up to 3.0 Amperes continuously (3.25 Amperes intermittently).
"B" 5 fixed voltages; 135, 90, 67.5, 45 & 22.5 VDC @ 60 ma
"C" 3 fixed voltages; 4.5, 9 & 22.5 VDC @ 20 ma

For those who need similar power supply options for vintage equipment, either of these two options would be very useful.

-Andrew

FM Receiver Basics

2008-11-09T21:47:00.000-08:00

I recently put together a short report on the typical parts of FM receivers as a part of a radio communications course I am taking.

I figure anyone looking for a basic understanding of FM radio receivers can benefit from this summary.

The summary turned out to be too long for conventional posting, so I decided to make it available via PDF.

Here are links to related the PDF files.

Downloadable Diagram

Downloadable Report

Speaker Repair: JBL JRX-Series Stage Monitor

2008-10-13T19:39:00.000-07:00

A while back, I started noticing that the floor monitors we use for live shows were not reproducing high frequencies. This post demonstrates some tips that I used to correct the problem.

Here is a short video I produced to help in diagnosing certain speaker problems.

Although it is not an all-encompassing guide to determining all speaker troubles, it does show how one can use a quick continuity test to determine whether or not a speakers wires have "blown."

User beware: applying high DC power to a speaker can cause mechanical fault, and quite possibly, electrical fault. When using the continuity method, only apply the meter for a few seconds. High current levels can sizzle finely wound voice coils.

Since speakers come in all types and tolerances, make sure the speaker under test can withstand a momentary forward or rear excursion before testing in this fashion.

Here is a view of the JRX monitor input section. Our monitors are the JRX112, 2-way 12" woofer and 1" high frequency compression driver model.

The cabinet-enclosed speakers are directly underneath the input section, so removing the input section is necessary.

Unscrew the HF compression driver assembly to remove it.

Once removed, the assembly can be separated into three distinct parts: Speaker Magnet, Diaphragm, and Enclosure/Driver Mounting Device. Of greatest practical concern is the pass/fail operation of the driver diaphragm.

The driver diaphragm is a simple device. It is merely a coil of wire surrounding a form connected to a rigid diaphragm, which, when inserted into the magnet, becomes a powerful electromagnet. The combined unit, the electromagnet, converts the incoming electrical energy into outgoing sound pressure energy

After the test shown in the video above, I determined that the diaphragm had to be replaced. I used resistance/continuity tests to determine the speaker "open." The replacement diaphragm ran about 20 bucks online, and is shown on the left. When ordering parts like this, be sure you are getting what you intend, and paying a fair price. I found several instances of deceptive pricing and advertising in my searches.

Here's what the new part looked like when inserted into the magnet. Exactly like the old.

As always in troubleshooting, we must try and determine what was the exact cause of the original fault. So, I looked back upstream to the input/crossover section.

Here's another view. The two bulb-like components look to be fuses. Both appeared normal.

One of the power resistors of the crossover appeared to have some type of residue at its base. Corrosion can impede or facilitate current flow. I checked the in-circuit resistance and read a low ohm value. A shorted resistor could possibly cause over-currents in the driver.

To make sure that the resistor was in fact shorted, I lifted a leg of the resistor and re-measured. I found that the resistor was not shorted, so the presence of low ohm-age on both sides of the resistor must be explained by the circuit. Guessing even further, the fuse would probably have blown in the case of a short fault.

All other components seem normal.

In the end, the non-working driver is most likely explained by thermal fault-open, caused by feedback loops and/or distortion in the live-sound environment.

I have been told that solid state amps, like the ones we use on stage, created odd order harmonics at ever-increasing frequencies when clipped. This clipping distortion causes power spikes to go to both woofer and tweeter. Though a woofer is robust and can withstand these higher levels of power to some extent, the tweeter is fragile by comparison. Higher current and voltage values, though not enough to blow a fuse, obviously cook the thin coiled HF driver wire, and cause the wire to break under thermal stress.

Feedback essentially does the same thing, except that instead of merely over-driving the amp, it overdrives the amp at particular frequencies which in turn are added and recombined continually until clipping occurs. The end result of either situation is that too much power is given to the HF driver, causing thermal fault, and subsequently, an open in the HF driver coil.

Thanks for viewing-

Andrew

C.L.A.S.P. by Endless Analog

2008-08-15T10:45:00.000-07:00

Designed and perfected by Endless Analog

Closed Loop Analog Signal Processing: A revolutionary system that merges the real sound of analog tape recording to the effortless flow of DAW recording and editing.

This system uses custom hardware, software, and control integration with, within, and between existing tape recording devices and digital audio workstations to capture REAL analog tape sound to computer hard drive. It does this while simultaneously providing accurate monitoring, routing, and machine/DAW control.

Simply put, the best parts of Analog are combined with the bests parts of Digital audio recording.

On 8/14/08 I drove up to Nashville's Ocean Way Studio to see a demonstration.

A good way to become familiar with CLASP is to examine real world situations.

An existing DAW system is set up to record a session. [This DAW system should include audio interface and ADDA converters.] C.L.A.S.P. VST plug-ins are opened on each track to be recorded to, along with an additional bridge plugin placed on any mono channel. The DAW then controls the Tape Machine through its own GUI and transport. When a track is armed on the DAW, that corresponding track is armed on the TM.

[The C.L.A.S.P. system works by using MIDI machine control, tailoring it to each specific type of controllable tape machine. Tape machine/DAW sync is maintained through the hardware unit, and is controllable by either the hardware front panel or the software GUI. Technical to layman translation: the CLASP Hardware receives MIDI machine code from the DAW, and coverts that system exclusive control to a form that the individual's TM can understand.]

The tape machine is set to IPS speed and switched to reproduce mode. In normal situations, that would mean the AE would monitor the playback. But with CLASP, the AE monitors the tape machine inputs, while the tape outs are simultaneously being recorded to hard drive. How does this work? This input monitoring is accomplished through transparent routing in the CLASP hardware unit. The AE is essentially monitoring what is going to tape while recording, which is how tape recording has always been done.

The inherent latency from record to play head is compensated for and resolved to the sample accurate level by the hardware. CLASP does not add playback latency. Playback latency can only be as low as your DAW's inherent latency.

[Playback is heard from the DAW, which is playing back the recorded analog tape recording-rerecorded to hard drive.]

The TM only acts when a track is record-armed on the DAW. The tape machine does not shuttle back and forth to get to specific times on tape, because that time-specific information was captured (post TM) to the computer. The TM is used as a simple analog processor.

This allows one to speed through editing, comping, overdubbing, and anything else usually done with DAW, all while retaining the analog tape flavor.

The greatest benefit of CLASP besides time-saving is that with it one can use less tape. One reel of tape could last an entire project if you so desired. Since the TM becomes part of the closed loop going to the A/D converters, and subsequently the hard drive, a single reel can record well over the amount tape normally required. For example: 60 minutes of rerecorded analog sound on hard drive on just 15 minutes tape, then replacing or reusing that same reel for the next 60 minutes...

People who have been recording to tape, then dumping to DAW, or vice versa can now forget about the time and tape wasted in the dump, because tape transfer occurs simultaneously while recording.

Forget about linear recording, keep the tape sound.

I think this product deserves some attention!

This product is in its final beta testing stage, and first orders are due out Sep 15th, this year. Inquiries can be made here http://www.endlessanalog.com/buy.html

[I was thoroughly impressed with this system and the presentation made by Chris Estes, the founder of Endless Analog. I am in no way affiliated with Endless Analog, or Chris. All photo © 2008, Endless Analog]

Fender Bassman Head-Switch

2008-07-06T19:12:00.000-07:00

A short time ago, one of my employers asked me to swap the amplifier head of a well-worn Fender Bassman with the amplifier head of a reissue Bassman. The speaker cabinet of the original had seen so many towns and stages that the wood was finally starting to give way. The owner and I could hear vibrations of parts and wood whenever the amp reproduced certain notes of the instrument. So the theory of the switch was to transfer the original's amplifier into the reissue's cabinet.

With the rear guard removed, you can see the classic point-to-point wiring and component selection of this 50 years young bass/guitar amplifier. Also shown are the 4 original Jensen speakers. I believe that only two of the capacitors are blatantly new.

Here you see the highly updated, PCB-laden reissue version of the Bassman. I am not sure of the speaker make shown below.

Lets face it...the reissue usually doesn't recreate the original. In order to save the sound of the original, the old and worn-out cabinet had to go. For the owner and I, saving the sound of the original outweighed the need for the reissue's precious completeness.

I first devised a plan of action. Since the four speaker outputs of either device where unlabeled, I marked the way that they were already connected.

Interestingly enough, I discovered the speaker outputs of the reissue in a different order than the order of the classic.

The next step was to remove the head-unit from the original. The head-unit is fastened with two nut/bolt assemblies. One must be careful in removing these assemblies, because the weight of the head unit is enough to make removal a two-person job. I had to grip the locking nut with piers and hold the amp up while the owner unfastened the bolts.

Here is another view of the original head (above,) compared to the view of the reissue head (below.) On both, the massive transformers (not shown) are affixed to the front of the chassis. The iron-laden transformers account for most of the head-unit's weight.

After another two-man action, we fastened the original head to the reissue's speaker cabinet. I reconnected the speakers to the appropriate output, retaining the original output scheme.

Although the reissues' speakers are not noteworthy, our opinion is that the amp itself contributes the most to the total amp/cab sound. In other words, it is the amplifier itself and not the speakers that affect the tone and sound quality of the combined amp and cabinet.

The last part of the switch was to replace the rear guard and test the amp.

I must tell you that literature suggests connecting the amp to the speakers before you power the unit back up. Especially in tube amps, if the amp outputs are not connected to some sort of load (usually presented by the speakers,) the amp may be loaded down to the point of internal run-away and ultimate meltdown. With no or little impedance seen at the output, the amplifier will start to work harder and harder to produce the current level allowed by the ultra-low impedance load. Unfortunately for the amp, the internal components can only allow so much current and heat before self-destruction.

In conclusion, the head-switch was a success. The reissue's cabinet lacks the certain resonances exhibited in the original's cabinet. The greatest part of all is that the unmatchable tone and quality of the original is retained for future use.

Thanks for tuning in,

-Andrew

The Laser Drive

2008-04-27T20:57:00.000-07:00

The Laser Drive

To all my viewers, let me take this opportunity to make certain that I do not in any way condone the cruel manner to which I seemingly approach the electric shock of my own person.

Electricity is a danger beyond comprehension, that often causes death. In any form, electricity is not a laughing matter. It is a wonder, and should be treated with the utmost respect.

That said, the video above shows the progress of personal discovery; in the sense of experimental amazement, and in the consequences of such an experiment.

Please do not attempt any such similar experiments without fully weighing the costs and possible, if not probable, risks.

There is no Real Genius here, only a fool with bottled lighting.

The background to this video began in a now defunct electronics shop. Among the racks of used and discounted items I found a peculiar power supply, which I discovered later to be meant for the powering of certain gas lasers. As shown above, the Laser Drive is a power supply that takes a standard 120Vac input and converts it to 2150 +/- 300 DC volts.

Being an intrepid experimenter, I purchased the device and devised a circuit by which to test it.

Here's the basic design. I had to do a fair amount of math in the planning stages.

I chose the two extremes of the rated voltage value, 2450 and 1850, to start the process. I knew that at the maximum voltage, the maximum current output was around 6.5mA. From there I used the maximum values of the voltage rating to determine a range of resistance values that would yield the appropriate current. 2450V/400KΩ ≈ 6mA and 1850V/400kΩ ≈ 5mA. So by this calculation I concluded that, at worst, if the voltage was actually as high as 2450 V, then I would need to control the output current to a limit of 6mA, by using an equivalent total series resistance of around 400kΩ.

I had no guarantee the thing would even work.

The next consideration was component tolerances. A possible max of 2450 V @ 6mA would mean a possible 14.7 watts of power dissipated across the 400kΩ. From this calculation it was painfully clear that the standard 1/4 Watt resistors I had on hand would not work. I had find resistors capable of sustaining the 14.7 Watt power.

I wanted to test the device's maximum voltage, but no DC voltage meter around here could measure that amount. So, to test such a high value, I envisioned a voltage divider network that would, presuming the component R values were the same, drop successive voltage amounts across each resistor to a level which I could measure.

Concerning the theory, I used Kirchhoff's voltage law, which explains how [summarizing] the sum of all voltage drops in a closed loop will equal the voltage applied, and my basic understanding of voltage dividers for my design.

An additional benefit to the voltage divider design is that it spreads the total wattage across the individual resistors; each resistor takes a smaller wattage than the total, while still dissipating the required 14.7 W.

Considering the high wattage of the circuit and the necessity of voltage division, I decided to check out the local electronics shop.

I found a ten pack of these Ohmite Brown Devil high wattage resistors in a value of 40kΩ. This was perfect considering that 10 x 40kΩ = 400kΩ. But as I was shopping, I had another idea.

I found a high wattage 30kΩ multi-turn potentiometer. By replacing one fixed resistor with the variable pot, I could create a controllable, tunable voltage output.

By using the 30kΩ pot, I could theoretically make a variable voltage small power supply out of the larger voltage divider circuit.

Here is a view of the assembled test circuit. Since I used (9)40k resistors and (1)30k pot, my approximate total resistance ≈ 390kΩ.

The total series current should be around 6 mA. The total voltage is speculated to be around 2450 Vdc, as demonstrated in the video. The total wattage per each 40k resistor becomes thus: 2450V/10 resistors=245V @ 6mA per resistor, or 1.47 W per resistor. 1.47 W is well within the rating I found in the Brown Devils.

This is one project I'm not likely to forget for a while. Even now I'm dreaming up different possible uses for it.

Thank you all for watching.

AC Voltage Monitor

2008-04-01T23:52:00.000-07:00

The Rainbow Kits AC Voltage/DC Voltage Monitor.

In summary, this is a simple kit designed to indicate a certain voltage input level by illuminating certain LEDS in a sequential fashion. The kit can be used to monitor DC voltages as well as in-home AC voltages.

To accomplish this with the standard VDC monitor kit, one needs a simple AC to DC rectification circuit. Notice I say rectification and not regulation...the included wall-wart VAC to VDC power supply has only a simple bridge style rectification and filter capacitor circuit. Any necessary voltage regulation must take place in the kit circuit.

The wall-wart transformer steps the input VAC down (10:1), so the VDC+ripple input to the kit is around 12 Volts.

Why the above label reads "110 Volt DC monitor," I have no idea...it seems a little confusing. RainbowKits.com needs a Net Disaster

The basic part layout is shown above. The supplied parts are shown below.

The kit did not include an enclosure, so I fabricated one out of plastic.

Here is a view of the enclosure and nearly assembled board.

I made the enclosure 'on the fly,' without much thought, and due to that lack of forethought I encountered a few obstacles. Specifically, mounting the LEDs would have been easier if I used hot glue. My enclosure drilling was not precise, so some of the inserted LEDs would not remain in place. The lead length of the LEDs was another challenge; as I soldered them in one by one, I found that no matter how hard I tried to keep the lead length constant, there appeared height variations between LEDs.

I slew the birds by taking pliers and bending their leads like so: < >, staggering them left/right so as to eliminate any shorting potential. This way, the variation in height was accounted for by vertical adjustment, and the same vertical adjustment solved the LED snugness issue by angle variations on the newly "formed" LED leads. Technically, after height adjustment, the LEDs angled out in different ways, so the outward variable force of each LED against the side of the enclosure holes summed was enough to hold the whole lot of LEDs in the enclosure panel.

Upon initial power-up, the measured VDC equivalent input was 12.58 V at input. From this value I concluded that the VAC output of the testing facility outlets was a little above 120 V rms.

The kit allows and encourages calibration. Using a small variable resistor (trimmer pot), one can set the approximate voltage points that ignite the LEDs.

To calibrate for input VAC, I needed a stable and defined VAC. For this I needed a Variac, a variable transformer based AC voltage adjuster.

I figured a trade off point between voltage sag and voltage hump out of my home VAC outlets should be around 110 V rms, so I set the output of the Variac to the amount shown above.

I then connected the Variac output to the wall-wart and adjusted the trim-pot so that a certain number and color of lights were illuminated.

Shown above is my calibrated light output for 110 V rms (In). The first Red, Yellow, and Green LEDS are heavily conducting, and the middle Green is lightly flickering (not shown).

110 VAC is about as low as I want to sag in my studio. Given that there's a nuclear power plant down the road, I probably won't be starved for power. Therefore, if I look over and see that at least two Green LEDs are lit fully, I know I have at least 110 Vrms at my wall outlets.

Any loading of the outlets on a circuit will cause variations in the available outlet voltage, and thats a different story. All the monitor is doing is telling me an approximation of the wall outlet voltage for that particular in-home circuit. For example, when the load on the outlet changes (i.e. the power or voltage/current demand changes,) the voltage output will vary accordingly.

When I hooked the wall-wart back to the original test outlet, the measured value was around 121 V rms. Notice below that the first Red, Yellow, and two of the first Greens are conducting heavily, while the third Green is half-way illuminated. This shows that the device has a roughly 10 V rms difference between 110 (2nd Green) and 120(3rd Green).
The kit does do what it is designed to do.

In use I noticed that the ICs run very hot, and I speculate that a more robust design/mod should include some sort of thermal drain or heat-sink. The high amount of heat produced during constant use suggests the possibility of thermal instability and drift. I tried leaving it "on" for few days and found that the loaded and unloaded voltage levels shown are different when comparing cold measurements to warm measurements.

Though the kit does function as intended, I don't recommend this kit as a
cheap replacement for quality (always-on) VAC monitoring. I do recommend it as a simple soldering and theory project.

Thanks for tuning in-

Andrew

The Government Phone

2008-03-26T23:25:00.000-07:00

Here's an interesting, almost assuredly, Government sanctioned communication device. Two points for whoever can tell it's application...

I found this device in a local electronics shop in a high technology area.

Check out these two mammoth mono phono plugs.

A look inside reveals a highly organized, fail-safe design. The leads and phono plugs use crimp style, screw-in terminal ends.

Between the two phono plugs, you can see a connector tension spring. Farther back, you can see a mil-spec resistor connected to one connector sleeve.

The transducers are first-rate, and brand-spankin' new.

So...I wanted to transform this device to a dual function, recording and monitoring device.

My ultimate goal for something like this would be a hand held phone apparatus, capable of receiving a mono headphone feed, (for monitoring,) yet able to send a voice to a mic preamp.

I found that the original configuration ran the tips of the phono plugs to what should be the phone voice pickup end, with both sleeves running to the voice listening end, modified by a small resistor between one of the sleeves and the listening transducer.

So I start disassembling the connectors...

Here is the 300 Ω mil-spec resistor I pulled out.

I decided to unite the white leads (voice capture module) to one phono plug, and the black leads (voice listen module) to the other.

I had to untie this wire knot at one point to allow for enough slack.

I made sure to mark what I "arbitrarily determined" ground to be. Not having the transducer specs, I had to guess that the notch seen above on the (white lead')-voice capture module was a ground of sorts.

Unfortunately, I found that this does not produce a functional capture module. I believe that the capture module requires some additional circuitry, possible a voltage level to function.

Here is a shot of the modified connections, after reapplying the solder-less screw fasteners.

As alluded to above, I failed to make the device work as I wanted. The voice receiver plays line-level audio well-enough, but the voice pickup mic does not work.

Nevertheless, here is a view of the finished product.

Thanks for tuning in.

The Sound Scriber

2007-12-21T07:23:00.001-08:00

A brief history of the Sound Scriber...

The Sound Scriber was recording device that was marketed to businessmen and secretaries. The name "Sound Scriber," was perhaps one of many brand-name dictation-marketed recorders...Edison Voicewriter and Infra Electronic were most likely competitors.

From approximately the 1940's to the 1960's, these machines recorded sounds direct to disc, in a manner similar to the vinyl disc-cutting process. In later years most companies probably adapted their machines for magnetic tape recording, or left the dictation-recording market all together.

Interestingly enough, Joan Crawford appeared in at least one Sound Scriber ad. I even found one Sound Scriber Ad that dates back to 1946.

While doing some work for a man in Gadsden, Alabama, I stumbled across this Sound Scriber microphone assembly. This particular man is a collector of decorative antique sound equipment.

The microphone wasn't being used in the display, so the gentleman allowed me to have it. I revitalized the collectors item by wiring it to modern standards. Here's how:

Here is how the original connector looked, it has six flat pins protruding out. The format seems to be proprietary.

Removal of the rear case revealed the inner workings of a mechanical switch. The switch must have served to operate the recording functions of the Sound Scriber.

Upon removing the plastic switch activator, I found the actual microphone transducer. You can also see an old radial capacitor underneath a copper strap.

Here is a view of the wires inside the cable. Notice that the two audio signal wires are shielded with braided wire. The filler looked like hemp twine and paper.

To install a modern audio connector to the cable end, I hastily soldered the appropriate leads to their respective solder-cups.

I admit that I did a very poor soldering job. You can see how I soldered it together without considering the slack left in the wires, leaving me to coil and cram the excess leads into the connector housing. I was just too lazy to desolder, re-lengthen, and re-solder the connector.

Nevertheless, I tested the mic on one of my external preamplifiers, and it actually works!

The next time I need a really obscure vintage-voice recording sound, I'll make sure to try the Sound Scriber...

Thanks for tuning in-

Circle of Lights, Part Two

2007-11-25T16:17:00.001-08:00

Continuing with the Circle of Lights...

At the end of Part One, I had just dropped the resist laden copper-clad board into an Ammonium Persulphate bath.

Here are some time-elapse photos of the etching process...

Slowly but surely, the solution removed the unprotected copper atom by atom. I had almost expected the Cu to come off in pieces...but I was surprised to find it simply disappeared.

As the copper was removed, the Ammonium Persulphate started becoming bluish-green in coloration. I suppose this is due to the oxidation of copper that must occur in the chemical process of etching.

Here you can see the excess copper almost fully dissolved off the phenolic board. The dark splotches are remnants of the copper.

Here is a view (opposite side) of the etched board. Notice the very bluish chemical in the bottom of the tank, and the etched design showing through the back of the board.

The whole process of etching (time in the tank) took approximately 20 minutes. The solution was most likely at room temperature throughout the process, and the 'agitator,' a small aquarium pump, was run throughout that time.

Here is the still-resist-laden board post tank.

A simple acetone application removed the remaining resist (toner) from the board. The copper traces showed some signs of etching, as if the toner allowed some of the copper to be removed. I checked the traces with a VOM meter for continuity, and all registered minimal resistance.

I used a small drill press and a 1/16th drill bit to make the necessary component holes.

After drilling, I grew tired of worrying about further copper oxidation and decided to tin the traces. At this point I remember I forgot to mention something...small, but perhaps necessary.

When I first opened up the copper-board packaging, and before applying the toner from the TTS paper, I made sure to clean the board with paper towels, cotton swabs, and Isopropanol.

Basically, it is said one should clean the copper of the board with strong alcohol before use to limit oxidation effects in the future, and to remove some of the unseen oxidation present on the board. I made sure to do this at each step: from toner application to pre-tinning, to post tinning flux removal.

I wanted to have the LEDs contained within an enclosure, and use Visual Communications Co's LED LitePipe® to transport the light to the chassis exterior.

This required a fair amount of in-accurate engineering on my part. I remember stuttering to myself in the electronics store trying to guess which PCB standoff size to use....

I had to figure for the chassis depth, the 5mm LED height, the board thickness, the LitePipe length (1/2") on the spot with a scratch sheet of paper and my cell-phone calculator. I ended up using a metric approach, converting and subtracting the combined theoretical internal board-height-limit of the design (the LitePipe + approx. LED height off board +/- a few mm for tolerance) from the total chassis depth. The answer I got was close to 1 1/2". I purchased both 1 1/2" and 1 1/4" standoffs. I ended up using the smaller ones, and as you will see, lifting the LED bases about 4 to 6 mm off the board to "meet" the LitePipe.

I used a copy of the original design to plot out where I need to locate the LitePipe.

By God's grace I managed to drill the holes close enough to where the LEDs would be located internally. Also shown are the On/Off switch, the Reset switch, and the Dimmer Pot.

This photo shows the beginning of through-hole component placement. Above the board you can see the actual bread-board experiment that I used to realize the original circuit.

Above and below show the component side and solder side of the board. As a side note, and in relation to the current measurements listed in part one (in schematic,) I hand-matched each one of these resistors to the closest possible range I could. In other words, out of 45, 5%-tolerance carbon resistors I was only able to find 12 in a range from 588 to 593 Ω.

I'd also like to add that the view show above is not encouraged. I did this to solder the resistors en masse.

Here are my favorite parts of this experiment. The LitePipe.

Each plastic rod extends into the chassis and meets (approximately) with the 12 LEDs shown below. See the topmost photo of this blog to view the inner workings of the device (board, lights, and standoff.)

When finished, I had a rather impractical yet colorful light show.

I thank God for seeing me through this project, and helping me to climb this mountain.

To Him be all glory, honor, and praise.

Thanks for tuning in-

Circle of Lights, Part One

2007-11-25T13:31:00.000-08:00

"I am mesmerized by flashy lights... especially Blue ones..."

Last September I stumbled upon a gas station carrying a brand of cigarette lighters that caught my attention. We've probably all seen them by now, especially if you happen to be a smoker. These particular "Ravin'" lighters have little multi-colored lights inside that illuminate when the flint is triggered. [I don't encourage smoking, I am a former smoker, twice removed...]

I was so intrigued by these flashy things that I splurged and bought 20 of these, "lighters," to play with.

To learn more about how LEDs work visit: HowStuffWorks.Com

The little lights inside are actually a multi-diode, multi-color 5mm Light Emitting Diode, attached to a simple series circuit.

A closer peek reveals the tiny component's secret... a type of semiconductor chip and three different color diodes. The chip and diode islands are enclosed with the 5mm wide capsule.

I don't have any tech specs for the LED, but my observation is that the device can withstand 9 volts series/minimum resistance periodically, (I did destroy a few in RnD). The unit runs very well, around 15 mA, with the Red diode consuming the least current, and the Blue and Green tying for the most consumption.

The chip has some sort of oscillation or timing scheme pre-programmed that makes the lights fire in a certain sequence.

How far we have come technologically.

If you were to try and purchase these LEDs in small quantities you could pay anywhere from $3-5 bucks a piece, and most if not all you could find would not incorporate an internal driver/sequencer. I payed a little over $1.25 per lighter...

So, after a period of tinkering and consulting my instructors at school, I settled upon a design idea. The idea was to arrange the 12 LEDs in a circular fashion, using my eyes to pick and order the LEDs in a fashion that made them appear to fire in a domino-like effect.

Here is a view of the DesignWorks Lite schematic I made to visualize the circuit and arrangement of the components.

I decided to go all-out for this project and design and manufacture my own PCB as well.

This is a hand drawing of the original design (above). I decide against doing a double-sided PCB and intended to make one side a silk-screen side...(more on this to come.)

After the design was finalized, I started assembling the necessary items to create my PCB.

I chose the Toner Transfer System as a means for placing resist on the copper board. It requires a toner-based laser printer and a source of heat and pressure. I got their recommended laminating device, as it applies great pressure and heat, fully transferring all toner to the copper-clad board.

However, my first attempt to transfer toner failed because of user error. As you can see below, when I tried to tear the transfer paper off the board, it pulled up some of the toner as well. I took till next time to do it correctly, a simple small amount of cool running water lifts the transfer paper away effortlessly, leaving the toner where it should be on the copper.

A funny thing happen between re-printing and reapplying the toner after the first failure. I became frightened (all of the sudden) that I was going about it wrong and should have reversed the drawing so that it would be "right." So naturally I reversed the drawing (above,) and reprinted onto the TTS paper.

However, much to my dismay, I found that I had been right to begin with, and that my second attempt, though successful, was actually incorrect in regards to how I originally wanted the board...

I gave up fighting that battle and continued with the board the (reverse) of what I had originally intended.

In order to etch the resist-covered board, I used a "tank," an "agitator," and some Ammonium Persulphate. (I had an aquarium heater too, but I broke it)

Ammonium Persulphate is poison. Take care when handling.

I made about 4 liters of clear etchant solution.

Here is a view of the etching tank and the heater I broke.

To etch, I simply filled the tank with solution, added the resist laden PCB board, and activated the agitator (a small aquarium pump.)

See Part Two to view the etching process and read the conclusion.

-Andrew

Pocket Audio Generator

2007-11-23T13:06:00.001-08:00

I found this Velleman Pocket Audio Generator in my favorite local electronics store.

Its function is that of a basic noise generator, but it also provides a 5 individual sine wave tones along with a pulsed-output option.

The possible usefulness of its features for both school and my work in audio, as well as the occasional need for pure tones at certain (known) voltage levels, lead me to give way to impulse and assemble it hastily.

This photo shows the first few steps of construction...a couple diodes, a few caps and resistors.

This PAG was rather simple to assemble. It shares a common enclosure design with the Velleman Pocket VU Meter.

This was my first-time forming and mounting resistors in this fashion. Getting the transistors and resistors to maintain a steady horizontal plane was difficult...(I had to re-form a couple.)

This photo shows the finished board assembly, minus output cables. Here is where the manual and I parted ways...

I decided I wanted to add a different connection scheme to the enclosure, particularly, a cable that I had made when constructing my last Velleman project (The Charlotte.)

There was little space left inside the enclosure to put a 1/8" stereo mini-jack, but fortunately God granted me grace in my placement guess.

Here is a view of the finished enclosure.

Next I decided to test the device.
My first test was with my handheld DMM. I plugged the 1/8" mini TRS into the PAG, then connected my Fluke to the first and second pins on the XLR end of the custom cable.

I was able to go just over 1 Vrms with the PAG at 1 KHz. Unfortunately, I usually want 1.228v when calibrating audio equipment. As shown above I could still theoretically achieve the 0 VU = .775 Vrms standard...

The next school day I took the PAG to the lab and connected it to one of our O-scopes.

The above photo shows the PAQ generating Pink Noise.

Here is a view of the test method.

Below I am showing the phase relationship between the two XLR outputs.
Each of the cables is being fed by the same output, I have merely inverted the phase of one signal 180 degrees to show the relationship.

All in all, this project was a interesting and challenging experiment...however, like the Charolette, the PAG will most likely see little important use in the future.

Thanks for tuning in,

and since I haven't mentioned it before,

Praise be to Jesus Christ for being Lord and God.
Without Him I could do none of this.
This day after thanksgiving, I thank Him for little things like pocket audio generators and blog posts.

All this stuff is vanity, but in Him is Life eternal.

Patching Up A Pignose

2007-11-11T21:49:00.000-08:00

So I've been working and "schooling" a lot lately...

I am getting to apply the things I'm learning in tech school at my studio job.
Recently the boss had me repair his ol' Pignose Guitar Amp. Here's a briefing:

The Pignose is a small 9 volt battery / 9 volt external DC power supplied guitar amp.

I found that this particular Pignose had major battery terminal corrosion, as well as damage to the DC input jack on the rear of the unit.

Simply put, the rear mounting for the 1/8" minijack had at sometime been completely torn off.
I found the actual jack just dangling in the hole.

I also found that the entire conductor, from (-) negative polarity battery terminal to the small circuit board, was completely oxidized. A few PCB traces were affected as well, as shown by the bluish-green coloration below. I estimate the oxidation began at the batteries, and slowly traveled up the wire, all the way into the PCB. The wire I removed was brittle, and the copper looked blackish.

I removed as much of the oxidation on the PCB as I could, and replace the (-) lead wire from battery terminal to PCB.

Next, I focused on replacing the rear DC power input.

A simple trip to RadioShack was all that was required to find a suitable 120 VAC to 9 VDC converter/transformer. I also picked up a few 1/8" mono mini TS connectors.

Making the cable was simple enough, although I did end up having to switch the leads (tip and sleeve) in order to get the amp working off external DC alone. Apparently they're serious about (-) to sleeve, (+) to positive...

Here's a look at the repair of the rear input jack. I used a couple of washers and some hot glue to stabilize it.

Fortunately, the Pignose survived my tinkering. It now works on either batteries or provided AC to DC power supply.

Thanks for tuning in,

Andrew @ AfaraWayland

"The Charlotte"

2007-08-27T22:21:00.000-07:00

So, this year I'm continuing my education...into a new yet related field: Electronics

I've enrolled in a local tech school to get some formal training, but the wait over the summer proved too much for my curiousity. I found and ordered a Velleman Super Stereo Ear

The Velleman kit, as you can see, is just a simple stereo mic/headphone amp design, that does not include any sort of enclosure, rather the PCB "sits" on the included power supply.

During my limited research into audio electronics, I found that the standard 5% tolerance carbon composition resistors have poor noise characteristics compared to other resistor types. Ultimately I chose to buy some 1/4 watt metal film resistors to replace the carbon comps included in the kit. Unfortunately, I was only able to find 2% tolerances for a majority of them, as 1% and lower tolerances are not usually stocked at your average electronics shops.
I found that my biggest headache was not finding components, but designing and finding parts for a project enclosure.

I also developed an ignorant spark of interfacing the Super Stereo Ear with a small recording device. The idea was to simply run the of the device into an old MiniDisc recorder. Even during the planning stages, I knew that the output of the of the Stereo Ear was mismatched, probably both level and impedance, with the mic input of the MiniDisc recorder.

Nevertheless, being young and gung-ho on making the thing work, I simply wrote it off as, "this is just an exiperiment."

Even before the device was constructed, I had already created a custom cable to transfer the recorded material to my DAW rig. Shown below is my special 1/8" TRS to Dual M XLR MiniDisc output cable.
Its an unbalanced cable by nature.The tip and ring going to separate XLR ends, sharing the same signal ground.

But back to the device...

From the beginning I had a vague idea of what type of enclosure to use, shown above is a standard plastic project enclosure found at any RadioShack. The problem was developing ideas for mounting the PCB based microphones found in the Velleman kit away from the PCB, and essentially on an outer boundary of the enclosure, positioned at a 90^ relationship between the mics.

Originally, I wanted to put a hollow angle piece on the front of the enclosure to house the two Mics. I found such an item unobtainable, even in junk stores and hardware shops. I finally settled on the item shown above: a common PVC plumbing cap.

Although slight more difficult to implement, I designed a way for the cap to sit atop the enclosure. This accomplished the goals of moving the mics out and away from the enclosure itself, while offering an option to mount the mics at a 90^ relationship.

Also shown above is my crude method for marking that 90^ relationship.

Once the idea was clear, I started the assembly process.

Shown above is the top panel of the enclosure.

Since I was going to place the kit inside the enclosure, I had to replace the PCB mount switch and and 1/8" minijack receptacle with panel mount versions.
Here is a shot of the concept:

Strangly enough, it resembles the old Monitor-class warships of yesteryear.
See any resemblance? Even more so, there is the current USS Charlotte defending our borders...

Here are some views of the PCB assembly.

I had to run some wires to and from the PCB/Panel-mount and microphone assemblies.

For the mics I had to proceed carefully. I decided to use a pair of 1/2" o-rings each to help the mic element stay inside the PVC cap.

After they were secure, I added hot glue to further seal and support the assembly.

I used some really messy gloop adhesive to seal the mic assembly to the enclosure.

After that, the project was basically through.

Here was the original idea, materialized.

I found out during testing that I had wired the mics backwards (oops,) so I had to modify it during the first real recording test.

Afterwards, I went back and rewired the mics appropriately.

All in all this was a fun experiment that taught me, not so much about electronics as it did the patience and persistence needed to take an idea from conception to realization. I gained valuable insights into what to do and not do in the future.

The Charlotte now reside in harbor on my computer table. She will probably never sail again.

Thanks for tuning in,

Andrew Wayland

ISA 428, Resurrected From Thermal Death

2007-08-12T13:52:00.000-07:00

Well, my Focusrite ISA 428 Preamp/ADC is back from the dead, thanks to Thermal Relief, Focusrite's only authorized, highly trained, repair service here in the States.

A while back I had posted a few views of my thermally damaged PSU area...

Apparently, the ISA had been sustaining a high amount of heat output both internally, and along the chassis exterior. Eventually, a few components in the power supply area began to experience performance loss, and that loss ultimately ended in the failure of one if not many PSU and ADC components.

Thermal Relief suggested a professional servicing of the PSU and ADC cards. I agreed, having them replace the main PSU capacitors and all of the ADC capacitors (electrolytics). They also replaced/reseated the unit's heatsink with a special thermal-transfer compound, hand-wired the PSU directly to the main PCB, and replaced a ribbon cable connecting the PCB to ADC card.

As you can see, I got shiny new caps all around!

All in all, the repair was quite speedy. It took only 2 weeks to have it repaired, including time spent in transit, shipping across country.

The nice guys at Thermal Relief instructed me to take certain precautions
to avoid a similar problem in the future, as the unit is apparently designed to run very hot.

I'm planning to take those precautions a step further by modifying the unit's top case to incorporate a powered cooling fan...I'll be sure to post my progress on that project when it occurs.

Until the next time, thanks for tunning in-

Andrew @ AfaraWaland

DIY: Calibrating AD/DA, Part Three

2007-08-06T11:58:00.000-07:00

In conclusion to parts One and Two, I have finally settled with a calibration level that I am satisfied with. After a few times of calibrating, using, and re-calibrating, I settled with the manufacturers factory reference of [0vu=+4dBu=-16dBfs]. I ended up trying several different reference levels in the process.

The topmost photo shows the analog voltage level that I chose. I found a chart in Bob Katz' book detailing the dBu to voltage correlations, and given the author, I took his word for it. Needless to say, my research into other websites revealed an identical result, leaving me to conclude that 1.228 volts is a standard.

So, how did I do it?

The first thing I did was to RTFM...and consult the friendly legacy-products support wing of Apogee's website.

Although the manual and the support files differ in their emphasis on which should come first...i.e. should the AD or DA be calibrated first...both list sequences that are relatively the same. I was able to determine that the DA section should be calibrated first from Frank Wells.

In order to calibrate the DA and subsequently, the AD, I needed a standard tone signal that equals 0vu/+1.228v. It just-so-happens that Pro Tools LE, one of my DAWs, has a "plug-in" that generates noises and tones.

Here is (above) PT's standard signal generator.

I created a new "test" session to perform the calibration duties, created a few sets of mono aux tracks to set up identical signal generators proceeding out through digital and analog outputs.

One interesting thing I found was that the internal levels of PT, in particular, the meters...do NOT correlate to accurate measurements at the digital and analog outputs. Such has been the cause for harsh criticism of DAWs in general, and heated debates on popular gear-junkie forums.

In other words, the zero (0.0) level in PT has an elusive meaning...One might think that the 0 level on the PT meters would equal either 0 peak or 0dBfs at the outputs...but they do not.

But before I get into that, I must mention how I made connections between the PT hardware and my multimeter, as well as my new PSX-100 and my multimeter.

First I ran a TRS to XLRm converter-cable from the main analog left output of PT to another XLR cable (for length extension). At the male end of the XLR extension cable, I used my knowledge of the standard pin structure of modern audio gear to determine which pins to hook my Fluke™ up to.

In modern XLR world, the standard is:
Pin 1=Signal Ground, Pin 2 = (+) or Hot, and Pin 3 = (-) or Cold.

So, given that an audio signal is an alternating current signal, I can measure the potential difference between Pin 2 and 3 to record the output voltage.

Shown above and below is the practical application of this knowledge, hooking red to (+) and black to (-).

When I had established the connections, I set the multimeter to detect AC voltage and then went back to PT and experimented shifting level and sine wave frequency to derive a proper and sustainable 1.228v across Pins 2 & 3 of the analog outputs.

Another interesting thing I found was that finding the right relationship of sine wave frequency-to-level was rather difficult.

I did manage to notice in the process that with the PT level constant, say, 0.1+, the output voltage varied relative to the selected sine wave frequency.

I really want to know why this is, so anyone who can tell me gets a free sandwich. As of now, my search for this answer is ongoing...

I finally found a sustainable sine wave frequency to level relationship of 980 Hz @ -13.9 (in the signal generator) with a PT internal fader level of 0.0 (Unity Gain).

This internal setting gave me a consistent 1.228v level at the analog output across Pins 2 & 3.
(Referencing back to the top,) I copied the setting of the same signal generator and applied them to two other mono aux channels. For these channels, I routed their outputs to the ADAT outputs of the PT hardware, and then onward (digitally) to the DA input section of the PSX-100.

Following the instructions, I set the Apogee unit to Calibration Mode, monitoring the DA side of the PSX-100's metering section. I then used a "tweaker" to turn the left/right trim pots so that each side read -10 on the Apogee meters (viewing DA) and 1.228v on the MM connected to the DA L/R outputs.

Then, I took the analog output originally used in setting the +4dBu/0vu reference and fed it to the PSX-100 AD inputs, adjusting the AD trim pots this time to make the Apogee Meter to read -10. [-10 = -16dBfs]

When all the tweaking was finished, I switched from calibration to normal mode and monitored the DA section. Sure enough...

A +4dBu signal equals roughly -16dBfs on the Apogee meters. AD/DA Calibrated.

Thanks for tuning in,

Andrew Wayland

AOTW Records New Single @ Clearwave

2007-07-02T20:56:00.000-07:00

From the other side of the glass...

Ahead of the Wake and I have shared a number of fun times.
This past weekend ranks right up at the top.

A few months back, the band and I, competed against other musical groups at a Habitat for Humanity fund-raising event. Long story short, on that cold, cold, day in Spring, we managed to best the competition and win a free tracking day at a local recording studio.

Ironically, the same studio that supported the event and donated the free time, happened to be a studio owner I had previously spoken with only weeks before. I remember nervously calling Jeremy Stephens up one day after stumbling across his website. On that particular day I wanted to ask him about recording business stuff...such as, what to do when you're fresh out of school and freelancing. Nothing' like being green to make you truly appreciate those with experience...

Jeremy has spent the last 12 years engineering and producing various types of music, various places. He currently owns one of the finest recording studios in the area.

Besides having a decade's worth of experience and fabulous collection of recording equipment, Jeremy is also formally trained. When he's not recording his own projects, he's engineering for his friend and mentor, Johnny Sandlin. He and Johnny go back as far as Gregg Allman's "Searching for Simplicity."

So, on with the session.

Load was at noon, and by 1:00 pm we had thoroughly demolished the sanctity of the main room with guitars, amp, and tangles of cable. Jeremy chose to put Clint, Zach and Eric in that main tracking area, with a view to both the control room and the drum booth, so as to allow them to effectively cue off each other. He also established a handy talk-back or cue mic in the room so clear communication between bandmates and production crew could be sustained.

Jeremy went about routing cables as I connected mics and headphones to the studio tielines. By around 2:00 pm, all the sounds had been set, and all the headphone mixes had been completed.

Then Murphy's law struck (literally) with a vengance.

North Alabama has experienced a very tough drought throughout the first half of this year. As if on cue, precisely before we were to attempt a first take, at least half of the City of Decatur lost electricity. I believe the irony went something like this, "The drought ends on the one day we set out to record..."

Jeremy was quick with contingencies. When the lights went off, the band became stranded in the tracking rooms. The cellphones illuminated the darkness as we all proceeded to stumble around in the dark. Jeremy grabbed a flashlight and some candles and rescued us from the utter confusion.

We obviously couldn't track without electricity, so we were left to marvel in the sounds of thunderstorm looming overhead.

After a period of time, and a few subtle promptings from band members, I called "lunch."
We took advantage of the time in other words...

I went home to grab a backup hard drive, while Jeremy and the Band went for a bite to eat. When we left we noticed that various sections of the town were out of power, traffic lights had ceased to function, and there was mass confusion among the cars in the streets.

Around 3 pm we all returned to the studio to find the power grid back up.
[sigh of relief...]

With the lights and studio power restored, we commenced to record AOTW's hot new single,"Rewind."

We tracked the song a total of 3 times, and collectively decided which of the three we like best. Once that was determined, I called for a certain set of small repairs and instrument overdubs to be performed.

Jeremy was nice enough to let me punch a few using his Nuendo DAW.

Here's a shot of Trey and Zach lounging in the control room, listening back to their accomplishments of the day.

The last thing to receive attention was Trey's vocal part.

Jeremy ran Trey through a Soundelux U195 (now Bock 195 I believe,) into a Langevin channel module that had been removed (and racked) from a Langevin vintage recording console.

The comfort of the studio atmosphere must have set Trey free that day. He performed his vocal part excellently.

My appreciation and gratitude go out to Ahead of the Wake for allowing me to be a part of their musical careers.

Our special thanks goes out to Jeremy Stephens for graciously allowing us his time and giving us a professional experience.

I think its safe to say that we all enjoyed working together.

Contact Information for the parties involved can be found in these links:
http://www.myspace.com/aheadofthewake
http://www.clearwavestudio.com/
http://www.myspace.com/clearwaverecordingstudio
http://www.myspace.com/afarawayland
http://afarawayland.blogspot.com/

Credit List:

"Rewind," was written and performed by Ahead of the Wake.

Tracking: engineered by Jeremy Stephens at Clearwave Recording Studio

Arrangement: Andrew Wayland Hull and Ahead of the Wake

Produced by: Andrew Wayland Hull and Ahead of the Wake

DIY: Calibrating AD/DA, Part Two

2007-06-20T20:36:00.001-07:00

If you remember, I mentioned in the last post, "In the digital realm of Audio, there is a theoretical limit as to how many bits can be encoded within the reference to a specified word length/or bit depth per sample. The maximum digital audio signal level allowable is zero dB full scale, or 0dBfs; it is considered the highest value of digital signal before truncation/distortion."

In other words, O dBfs is the maximum digital signal level allowed by the digital medium.

I should add that zero dB Full Scale is also related to the capacity of the system you are working on. If my understanding is correct, 0 dBfs will occur at different bit summations for different total bit capabilities. In other words, the zero dB full scale level will be higher on a 24 bit system than a 16 bit system. Likewise, a 32 bit system will have a theoretically higher 0 dBfs point than a 24 bit system.

In fact, Rane Professional Audio Reference says,'0 dBFS A digital audio reference level equal to "Full Scale." Used in specifying A/D and D/A audio data converters. Full scale refers to the maximum peak voltage level possible before "digital clipping," or digital overload (see overs) of the data converter. The Full Scale value is fixed by the internal data converter design, and varies from model to model.'

Jim Price, Understanding dB and Wiki say similar things.

I guess the major point is that Zero dB Full Scale relates directly to the individual system, and means nothing without a reference. My Apogee PSX-100 has a set internal 0 dBfs point. It also allows calibration, so that one can adjust the input and output levels of AD/DA to better serve its internal limits.

Frank Wells has this to say about AD/DA calibration.

Frank's article says, "Precisely calibrated converters are important to maximize headroom while ensuring you operate within the capabilities of your analog gear."

Some of my analog gear is capable of producing voltages up to +32 dBu...well beyond the capabilities of my converters...and well beyond most 0dBfs set points.

Enter reference levels and headroom...to be continued in Part Three

DIY: Calibrating AD/DA, Part One

2007-06-09T07:14:00.000-07:00

Recently, my main analog to digital converter and preamp unit, the Focusrite ISA 428 , experienced some sort of power supply failure.

Since Focusrite is based out of England, and there is only one authorized Focusrite gear tech stateside, I will be without 4 preamplifiers and 8 analog to digital audio converters for a considerable period of time.

My main production service depends on quality preamplification and analog to digital audio conversion to function efficiently.

In order to counter delays in my production schedule, I set out to find a replacement for the ISA 428. I ended up choosing Apogee's (out of production) PSX-100, a two channel AD/DA converter, along with a new API® Lunchbox-compatible mic preamp, the Buzz Audio Elixer.When I purchased the ISA, it came pre-calibrated with certain A/D input and output levels, specified at the factory.

Getting a used converter, one can never be assured as to what reference level the previous owner used (if any). Further, the PSX-100 has both A/D and D/A sections...

I referred back to my audio education at MTSU, and embarked to brush-up on AD/DA calibration. The resources I used included Mastering Audio, by Bob Katz, Desktop Audio Technology, by Francis Rumsey, and www.gearslutz.com among others.

In the digital realm of Audio, there is a theoretical limit as to how many bits can be encoded within the reference to a specified word length/or bit depth. The maximum digital audio signal level allowable is zero dB full scale, or 0dBfs; it is considered the highest value of digital signal before truncation/distortion.

How this relates to my system, (being part analog, part digital,) is that it is very possible for me to send an analog voltage of such intensity that my A/D converters cannot encode all of the information. The result of the overload of info is essentially bit truncation, which takes on an audible form of distortion.

Those of you who have experienced digital distortion know that it is far from pleasing.

More to come in Part Two!

Major Gear Failure: ISA 428 ADC goes down : [

2007-05-24T15:20:00.001-07:00

DOUH!!!

Today, I had my first real pro audio equipment failure...

My Focusrite ISA 428 unit's ADC card failed at the very beginning of a session today.

The 428 has been a pillar of reliability and usefulness ever since I bought in 2005. Alas, it is no more...perhaps.

When I ignited the unit's power, everything activated except the ADC card (lights won't light, DAW reads "lost clock signal"). The mic channels seem fine.

Not only did I have to cut the session short, I am down a great set of converters.

Bummer.

Nashville still has that magic

2007-05-17T16:17:00.000-07:00

It has been at least a year since I visited Nashville.

I moved my production operation to North Alabama mid-year 2006, and ever since then, my mind has constantly returned to the sights and sounds, the experiences and emotions of Nashville.

For anyone unfamiliar with the town, this historic metropolis is known for its magic.

Nashville has always had a mystical bloom of spirit and vibe surrounding it. It is a magnetic draw for music talent from all over the world.

Walk the streets of 2nd Avenue at night to find prospects performing street-side.

Nashville's promise of fortune and fame are just a dream away.

I had wondered if it would still be the same, having moved away to a more mechanical burg.

Much to my pleasure, Nashville still has that sensation of electricity.

I can attest: Nashville still has that magic.

Recording acoustic instruments, to mic or not to mic...Part Three

2007-03-25T22:00:00.000-07:00

Hi all,

I've rounded up some pictures of acoustic-instrument-micing I have been a part of in the past. Enjoy!

This photo was taken when I was still in MTSU, for this particular situation, we used a Neumann KM 184 and a Sennheiser MD 421 in a spaced stereo configuration.

Here's a shot of a miced solo violin. The mic shown is a Neumann KM 84.

This picture shows a mono "lap-dobro" micing technique. The mic in the photo is a MicroTech Gefell UMT 70.

When Labeling Deloris and I worked on their 2006 EP "For the Sake of Deloris," I used this stereo configuration. It is shows a pair of AKG C414B-XLS in an over-the-sholder mock-up.

This dobro shot was part of mine recent release, Hotcakes on the Griddle. It shows a 'knee-dobro,' with a spaced pair of microphones. One mic is an AT 4060 tube mic, the other is a Josephson C42.

This shot shows a standard XY stereo configuration using AKG 451 or AKG 460's...I can't remember which.

Here is a banjo player, shown with one Gefell UMT 70 and one KM 84.

How bout acoustic bass you ask??? LOL, here is a close-up of a Shure KSM 44 and a Neumann KM 184, here the "full-bodied-ness" of the bass is captured with the 44, and string slap or other noise is accentuated by the 184. Notably, I've heard wonderful results using a RCA D or DX-77 paired with a KM 84 in a similar configuration.

For the final shot, here is a session I attended at Oceanway Nashville during one of my college internships. The two mics shown are the coveted Neumann U-47 (on the left) and the famous AKG C-12 (on the right).

Thank you all for tuning in, please send me your comments or questions by clicking the "email wayland," button in the upper-right-hand side of the blog.

Andrew Wayland @ AfaraWayland

Recording acoustic instruments, to mic or not to mic...Part Two

2007-03-06T20:25:00.000-08:00

Continued from Part One:
What are some of the draw backs associated with each type of equipment/technique?

Direct recording techniques/equipment:

As I stated before, I've never heard a direct pick-up system that could match a studio-microphone perspective. However, with modeling technology and hybrid direct pick-up systems, who is to say the future does not hold better, perhaps more natural direct instrument pick-up options.

I own a Dean Markley contact pickup, which I bought for direct pick-up of my classical guitar...buyer beware, the contact pick-up works so well that it picks-up not only the guitar sounds, but every movement of the body, and cable, and flesh...to sum it up, I found the contact style pretty useless for serious work.

The "Blender systems," and "under the saddle," style pick-ups sound artificial.

Out of all direct recording types, I'd probably prefer the internally mounted condenser mic, linked to a hybrid blender system.

External Microphone techniques/equipment:

The drawbacks are: choice of microphone(s) and preamp(s), the room or booth acoustic characteristics, followed closely by where to place the microphone in respect to the instrument. Remember from part one that there are many variables which contribute to how we perceive an instrument's sound. Even the grandest of performance halls must fall before the physics of sound...

The most important things...
[1.]Of the ones mentioned, room and environment spectral response should be one of your greatest concerns. (Forget about the HRTF, ITD, and ILD stuff. These concepts help explain what makes a natural sound, sound...natural.)

Room size, construction materials, and presence/lack of acoustical absorptive materials, will determine the amount of (constructive/destructive interference) comb-filtering present in the room and ultimately, the recording. The combination of the room's acoustic characteristics and the instrument spectral projection make an instrument "sound," the way it sounds.

A real-life example of this: playing a guitar in a small bedroom versus playing that same guitar in an enclosed auditorium.

The environment in which it is played has everything to do with the way an instrument sounds.

So, although "comb-filtering" is a natural part of HRTF, and our everyday lives, certain versions of "CF" in a recording space make for unnatural instrument sounds.

If we can control or alter the room's acoustical characteristics, chances are that we can achieve a recording that accurately portrays the instrument.

After maneuvering around the room's acoustic characteristics, the next most important element of capturing the instrument in the most natural fashion is IMO [2.] mic placement.

Certain instruments project sound in ways we might not expect. It is the job of the engineer to find the place in the room which most accurately conveys the natural sound of the instrument. Much literature has been devoted to the study of how instruments project sound.

However, opinions on the "best way," to mic an instrument are numerous. I believe that someone who desires to know what the "best" way is should research the standard techniques found in any basic recording manual, and then apply those techniques to their recordings. With trial and error, that someone will learn what he/she likes and dislikes concerning mic placement.

I have encountered many instances where my first choice of mic placement failed, either in capturing the sound I wanted, or more specifically (when using stereo techniques,)in leaving overlaps and/or gaps in the captured instrument's stereo perspective. Phase cancellations, comb-filtering, and blurred stereo image can ruin a recording. Yet, as I suggest above, I endeavor to find what is best, through many successes and failures.

Also, when using stereo mic microphone techniques, it is very easy to use too-many stereo-tracked instruments...to get carried away in the stereophonic bliss of recording...

Critiques of a recent recording I did, where I tracked many acoustic instruments and used mainly stereophonic procedures, have revealed a blurred sense of instrument placement within my mix.

The listener struggled to discern [between stereo monitors] where the elements of the mix were placed or coming from. My failure as a mixer...

To hear an example of such a production, listen to Hotcakes on the Griddle on the AfaraWayland Soundclick page.

The moral of the story: the combination of monophonic and stereo recording techniques could help clear up spatial perception within a mix.

Hints and tips:

At the top of the original post, the top picture shows a dobro player and the mic technique I used to record him.

I started out by setting up the mics in a familiar fashion, the famed "over-the-shoulder," technique. Then I brought each mic up to a good amount of gain, listening to each one in center-panned mono (many would suggest single speaker mono, however).

Once each mic's gain was relatively set, I panned the mics out in a stereo configuration in the monitoring system. However, I didn't pan them very wide, since I found doing so created a "hole in the center." [This is due to this particular micing technique, other stereo techniques can be panned wide]

Using two channels on my ISA 428 preamp, I trimmed the gain of each channel until the image of the dobro fell into the spot I wanted it.

From that point on, I made sure to monitor the image after each pass, adjusting the balance whenever I found that the performer had shifted position in his chair. When close-micing a performer, even slight movements can skew the stereo image and perceived sound of the instrument. [This is not so much a problem with distant micing.]

The result of this technique can be heard by clicking "Hotcakes on the Griddle"

Come back for Part Three, and see some example of stereo-micing techniques!

-Andrew @ AfaraWayland

Recording acoustic instruments, to mic or not to mic...Part One

2007-02-22T20:06:00.001-08:00

"_____, writes: 'So, is a built-in guitar mic a no-no most of the time, or is it just another example of "whatever sounds right to you"?'

This is another issue best left up to individual preference, de gustibus non disputandum est...

Yet, for those of us trying to capture acoustic instruments as they sound 'to us,' the issue of what methods and/or equipment should be used to do so is quite valid.

What methods can we consider when choosing how to record an acoustic instrument?

Referring back to a previous discussion, the two ways we can capture acoustic instruments are:

1) Use a transducer of some type that turns acoustical energy into electrical energy. (i.e. Microphones)

This type of reproduction equipment can be located internally or externally in relation to the instrument body, and may be combined with other functions such as the surrounding acoustical environment, microphone placement technique, and signal processing.

2) Use a transducer of some type that turns mechanical energy into electrical energy. (e.g. bridge pick-ups, contact pick-ups, etc...)

This type of reproduction equipment is usually fastened directly to the instrument body. Its principle method of transduction works by capturing string or body resonances and vibrations. This method may also be combined with signal processing, but given the placement and operation of this type of equipment, acoustical environments place little if any part in sound characteristics.

How can we achieve the most natural acoustic instrument recordings?

With microphones, you can use one or many.

I've noticed that monophonic techniques can help instrument tracks 'fit' into your mix, even broaden the production value of certain songs. However...

In my humble opinion, the best way to capture an acoustic instrument (so that it is perceived as sounding natural,) is to use two high quality microphones, tactfully placed in some type of stereophonic configuration.

The reasoning behind this deals with the fact that we have two ears. Head-related-transfer-functions, interaural time differences, interaural level differences, spectral room and environment response, to name a few...all affect how we perceive sound.

When capturing an acoustic instrument to a recording medium using two microphones, one can manipulate the sources to reflect how the instrument sounds at that time and place, mixing the two so that it seems to be natural...(although, in fact, it is not,) and so that it occupies a perceivable yet intangible space.

I personally prefer using either tube based or solid-state based condenser microphones. Yet, you may find that certain instruments sound "better," to you when using ribbon mics or even quality dynamic mics.

I also personally recommend the API 3124+ for microphone preamplification. Millenia, Neve (duh), John Hardy pres are also equally as excellent.

I have had good results with my Focusrite ISA 428 pre's.

Microphones I have heard, used, and favored for acoustical instruments:

AKG C 414 X-BLS, SolidTube
Audio Technica AT-4060
Coles 4038
Josephson C 42
Neumann KM 84s, M-147, TLM-170
Microtech-Geffel UMT-70S, M300
Milab DC-96
RCA Type 77
Schoeps CM5
Soundelux U-195

Nevertheless, sometimes microphones don't work well in certain situations.

For their simplicity and functionality, especially in live sound reinforcement situations, mechanical transducers or internal instrument body mics can be a measureless blessing.

Direct instrument pick-up systems continue to advance in sophistication and quality every year. Here are some examples:

www.kksound.com

www.frets.com

I've personally tried some of the more dated types of this technology, mainly magnetic soundhole pick-ups, "Blender" systems, and contact pick-ups, but have experienced others' "under-the-saddle," pick-ups and internal body microphone systems when running live sound reinforcement operations.

I can say that I've never heard an internal instrument pick-up or mic that sounds like 'real-life,' or rather as good as my preferred stereo pair techniques.

I can also say that I'd never prefer to mic a live-performed acoustic instrument if a direct pick-up system could be used...nobody likes acoustic feedback during a concert.

End part one...

Continued in part two.

Thanks for tuning in!

Low Frequency Response and Headphone Mixing?

2007-02-09T22:51:00.000-08:00

"_____ wrote: If I move an inch in my room my bass mixes sound different."

Here's my response:

Well, sans mix error/problems, your room should respond equally to program music, whether it be your own mixes, or CDs you play through the same system.

Eliminate all the variable reasons why there may be frequency balance differences between your mix and a professionally mixed/mastered CD, and what you have left is your room.

Room modes, resonances or cancellations, in lower frequencies are going to exist in your room.

There's practically no way to avoid it. There are ways to detect room modes and the like. Many acousticians use complex measurement techniques and software programs to uncover the smallest of details. For those of us with shoe-string budgets, other methods are available that can let us "hear" some of the room's problems.

Here's something I tried when setting up my mix room/studio.

I tried playing sine wave tones through my system. I started at say 180 Hz, and moved around the room. I tried to note the differences in intensity when moving around the room, comparing the levels to the mix monitor "sweet spot." I continued this for each main frequency band down to 20 Hz. I made sure to keep the level the same for each frequency.

I noted that some of the lowest frequencies are harder to hear (if at all) when played at the standardized level. In this instance, what I was experiencing has to do with the human bodies' response to sound.

I think I also tried a pink noise signal. Again, I got a reference perspective in the sweet spot and then moved around the room.

I found that many places in the room had heavy resonances while others had severe cancellations. At the lowest of frequencies, from 60 Hz down to 20 Hz, even the sweet spot had cancellations.

In any untreated square room, low frequency response will vary greatly. The fact is that the shape, constructive materials and room occupants (things or persons) will contribute to that room's spectral balance.

[I am purposefully avoiding the variable of monitor placement, which can also greatly influence monitor/room response.]

So, at that point I realized that even with the treatment I had previously installed, my mix room was still greatly under par with professional mix spaces.

Again "_____ wrote: If I move an inch in my room my bass mixes sound different."

And yes, the science of acoustics is this precise. When you dissect it to its root, sound is a constantly changing variable.

This situation should NOT happen in a professional environment to the degree that an experienced and trained engineer could tell. (the frequency fluctuations at such a narrow range)

But for those of us with bedroom studios, small and constructed of drywall, this situation is very much a reality.

So what can we do about it?

To counter the lack of low end accuracy in my room, I've been making critical EQ decisions using my Grado headphones.

Earlier this year, I came across an article in the Sound on Sound trade publication that discussed the same sort of situation.

I'd suggest checking January [07's] Sound on Sound article concerning headphone mixing, to anyone struggling with a similar low end frequency problems. Though I do stress the importance of monitors, and their permanent place and function in the studio, for those of us who want spectrally balanced mixes, yet can't afford a spectrally balanced room, reference class headphones present a good option.

I'm not talking about 100-200 dollar headphones. I'm talking Grado RS-1's/2's, Senn. 650, or AKG's new 700-something.

I found the article very helpful.

Headphone junkies beware:

There are several critical issues with headphone mixing.

Self-respecting engineers will tell you that headphones fail to give the same picture as professional monitors, and that headphone mixing generally shouldn't replace monitor mixing. I agree with this mentality.

I own and use RS-1's on a daily basis, but have fall back plan, my Alesis Monitor II three-ways, and other consumer grade speakers. As suggested in the article, I use the speakers to check stereo balance and placement, but go to the headphones for critical frequency evaluation.

Suddenly, "if I move an inch in my room my bass mixes sound different," no longer becomes a terminal problem.

Thanks for tuning in,

Andrew @ AfaraWayland

DIY: Acoustic Treatment: Tube Traps

2007-01-18T12:13:00.000-08:00

First off, the real-deal tube traps are made by Acoustic Sciences Corporation, they can be found here: ASC.

Here is the ASC link to Acoustic knowledge: Studio Acoustic, The Basics

I do not know how these awesome units are made, nor do I know what materials they are made from.

For everyone out there on a home-studio budget, here's another DYI experiment for you.

Hello all,

Here's another "do-it-yourself," home studio acoustics project. I first came across this idea when researching Owens and Corning 703 panels.

I've found two similar websites devoted to sound absorbing 703 tubes both of which may provide you with greater detail on the applications and the reasoning behind them .

DIY Tube Traps

JTGale,Cylindrical Bass Absorber

The first link comes from an audiophile perspective, it's practically a basic instructional guide. I found the second link more helpful. It includes a some technical information about frequency response, etc...

However, I cannot substantiate any the technical facts presented in the article, as there is no practical way to test the tube traps.

A closer look at the links above will show that I have deviated slightly from their plans.

So heres my version of DYI Tube Traps.
When I was up in Nashville last year to get my mother-load of 703, I happened to ask about the 703 pipe insulation that I had heard about through the above sites.

The man at Nashville Insulation Supply showed me a wall to wall selection of tubes in sizes from 6 inches to 20 inches diameter. Above shows standard 15 inch rigid fiberglass pipe insulation.
At this point I should give you some details...

Required materials:
703 pipe insulation tube(s)
1/2" or 3/8" single pane sheetrock
Hot glue gun with glue-sticks
Acoustically transparent fabric (or similar)
Latex gloves
My first task was to cut out the base and top for each tube. As you can see, I chose to use the advice of J. T. Gale, who suggested using sheetrock for the end caps rather than wood or particle board. With a Roto-Zip®, and a ceramic tip, the sheet rock shapes up easily.I used the tube to trace a ring on the sheet rock, then used the Roto-Zip® to cut out the pattern. Once the sheetrock was shaped, I set the tube atop the circle, and put generous amount of hot glue around the bottom edge of the fiberglass. I was careful to put as much hot glue as the 2" of 703 could stand, so as to give the unit more stability.You can almost tell from this picture that there are gobs of
glue surrounding the bottom of the tube.
Here is a view of the top cap.I made two presumable errors when constructing these units, that possibly affect their ability to absorb sound.My first blunder was to experiment with loose cellulose insulation. Here you can see that I filled the 703 tube cavity with the cellulose.

Another possible mistake I made was choosing to leave the kraft paper facing on the tubes. If I had left out the cellulose, the paper could have acted as a membrane through which lower frequencies could be attenuated. However, it is possible, that by filling the air gap inside the tube, I canceled out the benefit which the air-gap/kraft paper might have made.

The idea was to increase the mass of the tube. Perhaps, increasing the mass combined with an absence of the paper membrane would allow the sound wave to enter into that mass, and disperse as heat more efficiently.

Unfortunately, I have no way to test the factual response of the tubes I made.

So here's how they turned out.

Thanks for tuning in,

Andrew Wayland @ AfaraWayland

AOTW and AfaraWayland

2006-11-19T16:35:00.000-08:00

Hi all,

AOTW and I have been playing out a lot recently, here's some shots from our travels.

Trey
Gilby
Zach
Eric
The crowd
Me

-Andrew @ AfaraWayland

Where have I been?

2006-11-19T15:33:00.000-08:00

Apologies to all for the hiatus in posts.

So what's been going on at AfaraWayland?

As most of you know, I've taken a job at a local TV News station. I do broadcast audio, camera work and teleprompter work...but recently I've taken some new tasks.

My newest role at WAFF 48 is: internet producer. I manage the National, and International news pages at WAFFcom.

Here are the associated links. I update these daily during the week, usually by 2 p.m. each day. Check them out for your latest US national and international news.

WAFF_National News

WAFF_International

I try to post stories, pictures, and videos that I think are pertinent to today's U.S. citizen. Some of which you won't find in your usual mainstream media.

Happy Happy, Joy Joy!

-Andrew Wayland

Live Performance: Ahead of the Wake, Big Springs Jam 9/24/06

2006-09-26T18:52:00.000-07:00

Hi All,

This is a special post for Ahead of the Wake and I. In the last post, I eluded to a special performance at an annual festival in Huntsville, AL. This annual festival is none other than Big Springs Jam: 14.

Big Springs Jam boasted over 90 groups this year, with headliners including: Blood Sweat and Tears, Boyz II Men, North Mississippi All-Stars, Jo Dee Messina, Sugarland, The Wailers, Joan Jett and the BlackHearts, Clint Black, 38 Special, Nickel Creek, KC and The Sunshine Band, to name a few.

In fact, when we arrived and began load-in, we were quickly ushered to the rear of the stage to make way for Joan Jett and Blackhearts.Apparently the big stars demand that no one can be on "their" stage during sound check. I'm told that this is standard practice. Here is the group waiting for their turn.After an hour and a half of waiting, we finally got the word to rush the stage and setup our gear. Fortunately, Otis, one of the more friendlier stage techs helping us, had already made most of the preparations.
Being the most time consuming element to perfect, the stage tech set Trey's drums up first. Here is some profile shots of the band after setup, taken by our friend Melissa.Here are some views from the front of the stage. AOTW began playing as the gates of the festival opened, and people trickled in as the show progressed.So what about me? Well, after helping supervise the stage setup, I was introduced to the FOH engineer, Mark. He graciously allowed me to run the faders during the performance. Here's some shots of the FOH tent:The FOH console, I used was the Yamaha PM5D, one of Yamaha's premier Live and Broadcast sound digital consoles. On a side note, a few weeks before, I was able to demo this console at my day-job. The standard unit has 48 mic inputs, 32 bit processing, and sample rates up to 96 KHz. The rest of the PM5D specs can be found here.This is a really neat console, and seems to be quite popular among live sound engineers.

We really enjoyed out time at this year's Big Springs Jam. Thanks to all of you who came and gave your support. Keep checking back for more updates on the band's progress.The weekend ended in fine colors.

~ Andrew @ AfaraWayland -- The graphic below courtesy of Randy.

Live Performance: Ahead of the Wake, Side Pocket Lounge 9/23/06

2006-09-24T11:28:00.000-07:00

Hi All,

Well, Ahead of the Wake has resumed their tour schedule for Fall 2006, kicking off the return at the Side Pocket Lounge in Florence, AL.The Side Pocket boasts a pretty large performance area. It has a large stage and dance area, an upper level with pool tables, and a centralized beverage distribution center.

Given the expanded size of the performance area, I decided to reinforce some additional elements of the drumkit. Instead of the usual overhead/kick mic setup, I added some Tom and Snare mics.
This busy night was accompanied by the excitement of a following busy day...

The next post will detail a special performance at an annual festival in North Alabama....

-Andrew @ AfaraWayland

DIY Acoustic Treatment: Standing Mineral Wool Panels

2006-09-17T13:15:00.000-07:00

Hi all,

It's been at least a month since the last post -- [insert excuses here]. I'd like to start by giving a shout-out to Brandon @ RecordingReview.com. Brandon commented on my earlier blog concerning 703 absorptive panels. Since then, Brandon has welcomed me to the RecordingReview.com Forum. I suggest that anyone looking for some interactive discussions on audio related topics to look up RecordingReview.com.

On to the business of today,
DIY-low frequency absorbers and the applications of rockwool/mineral wool. Shown above is a bundle of what is know as rockwool or mineral wool. This material, though used as a type of insulation in housing, has become widely accepted in the DIY acoustic treatment circles for its usefulness in controlling sound.

Check out info on Mineral Wool here: Wikipedia

I started by theorizing a frame system to hold the 2'x4' panels suspended in the room. Beside hanging the panels from the ceiling, a simple frame design proved a more practical solution for a particular area of the room.I constructed the frames with 2"x4"x8' wood. Two 8' pieces make up the outer frame, with 2' sections making up the supports. I decided to mount six feet of panel from the top of the frame, adding a 2'x2' square to the full 2'x4' panel, leaving roughly two feet of frame left at the bottom.At first I tried just wedging the insulation into the frame. I found that the heavy mineral wool would need an additional support to keep it in place. My father suggested that I use vinyl mesh.Here is a view of the mesh wrapped frame. I had to remove the panels to complete the job, re-installing them after fastening mesh to one side.The next step was to add supports (feet) to the frame, to keep it standing upright. In the future I can add casters to make a rolling frame if necessary.The last step is to cover the framed panels in fabric.For fabric, I chose quality polyester dress-liner. Its hard to go wrong with Black.
So here is the final product! The room shown at the top of the post shows two frames, aka, low frequency absorption standing panels.

Stay tuned for more posts on DIY acoustics!

-Andrew Wayland @ AfaraWayland

DIY Acoustic Treatment: 703 Rigid Fiberglass "Soundboard" Absorbing Panels

2006-08-22T16:16:00.000-07:00

Hi all,

Sorry for the lapse in posting here at AfaraWayland, I've been exceedingly busy trying to get this new mixing room finished! Right now, I've installed all my acoustic panels and am moving on to wiring and calibration.

BTW, I'd like to send a shout out to my viewers overseas and out in the west coast area...From Oz to Japan, and from France to California, Hello!
The topic today concerns Owens and Corning 703 Rigid Fiberglass panels, and using 703 panels for the purpose of sound absorption. There are many resources available on the internet concerning acoustic physics, truths about controlling and manipulating sound by the implementation of acoustic physics, the theories and principles of using sound absorptive materials, and most of all, sites that can tell you why a person might want or need sound absorption in the first place.

Before I begin, I should mention that I am not an Acoustician. I have borrowed from the designs of others, and utilized my own understanding of the physics of sound and studio design to create a plan for treating a particular room. Please understand that not all rooms used for audio production have the same acoustic needs, and if one intends to treat a room right, one needs to understand the principle behind it. In other words, before you rush out to buy 703 or rock-wool, know at least what you want it for, and whether or not you really need it. [If you'd like to contact me about the treatments I'm using, I will gladly answer any questions you have about them or perhaps give you primer on the theory behind them.]
Case and point, playing around with insulation can be an unpleasant experience. For those of you who don't recognize this, this is what we call insulation rash...

A few posts from now, I'll clue you into the particular resources I used to develop a solid plan for treating my new room. For now, I'll try to stick with a step-by-step explanation of how-to for you DIYers out there.
So, on with the post, the first materials one needs to construct these panels are: 1"-2" 703 Rigid Fiberglass, 1/4" plywood or luan, and few cans of spray adhesive. The amount of materials is obviously proportionate to the scope and scale of your intentions.

Owens and Corning 703 and 705 insulation cannot usually (if at all) be found in your local chain hardware store. One would need to find an industrial/commercial insulation supplier for these items.

I started out by contacting Nashville Insulation Supply for my 703 needs. Before I moved the HQ to AL, I bought 3 bundles of unfaced 2" 703 (enough for 36 2" panels....) Unfortunately, in North Alabama, my new location, I haven't been able to locate any retailer of 703, 705, or rock-wool. The closest place to buy rock-wool in AL is about 1 1/4 hours away. I've had a satisfying relationship with Nashville Insulation Supply (formerly Breeding Insulation), so when I came to needing more insulation, I decided to make the trip back to Nashville.

So, once I obtained North Alabama unobtainium in Middle Tennessee, I focused on the next materials- 1/8"-1/4" Luan or plywood and adhesive spray. I was able to find these popular building supplies at a neighborhood Lowe's. I purchased a few sheets of 1/4" 4'x8' plywood, and as you can see above, I then proceeded to rip the 4'x8' sheets into 2'x4' sections (perfect for the 2'x4' panels!)Depending on the type of method you use to "rip" the plywood, it is very likely that the saw blade will leave a picture just like this...shards of splinters in every direction!In the likely event that this happens, just use a basic sanding block to scrape away the roughness. Do use gloves though, splinters underneath your fingernails can be very painful-not to mention a certain risk of infection.

Once the "panel backing" is finished, next, you simply apply a generous coat of spray adhesive to one side, and lay a 2" insulation panel onto the adhesive-treated surface.
After the panel sets, (allow at least 30 minutes,) you are then ready to cover the insulation/plywood backing in fabric.A word on fabric...

I found out after making 12 panels that certain types of fabric weaves and textures are disruptive to the panels sound absorption. The source I used (*See Below for Link to this cited work) explained that shiny fabric, usually having a limited stretch (tight weave,) will actually impede high frequencies from entering the panel insulation. The smallest wavelengths will simply reflect off of the stiff and shiny fabric.

The fabric you see above is relative shiny polyester weave. After learning the new information, I found that during the "*blow test," I had to put a moderate amount of air pressure to the fabric to get the air flowing through. However, spending as much as I did on the first batch of fabric, I decided to use it in combination with other panels (with better fabric,) that would absorb the high frequencies that are possibly reflected.

I am glad that I learned before I finished them all...
To cover the panel, measure out some appropriate size fabric pieces. Make sure to account for the panel surfaces, plus some overlap on each of the sides in your measurements.
Layout your fabric on a flat surface, then place the panel insulation-down onto the fabric. Adjust sides to make them even as necessary. A good test is to folder the overlaps over the wood. That way, you can be sure to have enough overlap on each side, and adjust positioning further if needed.

I choose to fasten the fabric with a common staple gun. Use whatever works for you.
My method for tacking down the fabric is to start with the top and bottom 24" sides. I prefer to stretch the fabric a good bit, but the downside is that is makes irregular "wave patterns" on the sides of the panel where the fabric and staple are fastened. One way to avoid this is to build 1"x2" frames to surround the outer edges.[You may notice that this is a particularly thick panel. It is in fact a panel made from 2 2" panels glued together. I made three 4" thick panels.] After tacking down the 24" sides, I work on the 48" sides. The most challenging part of this is getting a flat and uniform corner-to get mine, I used a simple folding method.When Finished, trim of any excess overlap, and whalla! You are now ready to devise a hanging strategy.

Well that's it for now, still a lot more to come.

Here is a link to a very useful site, and the work * cited * above! This is one of the resources I've found that will benefit anyone new to the topic:

http://www.ethanwiner.com/acoustics.html

More posts coming soon!

-Andrew Wayland @ AfaraWayland

A Word On Music Performance: (Original) Song Introductions...

2006-08-18T12:17:00.000-07:00

Hi all,

This is an excerpt from a letter I wrote to a contemporary of mine, a singer-songwriter who I witnessed perform one day. All of you songwriters and groups should take this letter to heart, should you not already be aware of this!

"Hey ____

You may remember me from ----- earlier this summer with ----...

Just a quick word on your performance,

I noticed that when you introduced the song, you said something to the tune of, "this is an original..." and I can almost be certain that you usually announce your songs this way.

Cotton-gloves off...

This is IMO a bad habit, and I think that you should consider introducing your original material in a different fashion.

Put it this way, if someone likes a song you write, they will come to you and ask what it's called. Even if it's smack in between cover songs... The point at which people desire to know what that mysteriously wonderful song they just heard was called, is the exact point when you can know that you've actually written a song that connects with the audience, a.k.a. a good song.

I don't care who it is, you'll never hear a professional artist get on the mic and say "this is an original..."

Since we strive to become those people eventually, naturally, it is best to imitate them as much as possible right now.

Many bands and artists I come into contact with reproduce this same bad habit over and over. It is something that unless a person is told, he or she is likely to continue in ignorance. So...

That said, please don't take this as a jab on you or your art. I had to learn the same thing, in the same fashion. Someone had to tell me. "Stop doing that!"

I suggest that you refrain from announcing any original song until its published, yet if you absolutely must introduce new material, say something along the lines of "this is a track from our next/current album...or, this songs called..."". It is just sounds better. Here's why:

Mentioning the "originality of a tune" can be perceived negatively by the audience.

When an artist says that...what they are really saying is, "hey look at me! I wrote a song!" Anyone can write a song.

Good songs need no introduction.

If people are generally intolerant of those who display narcissistic qualities, consider then, that when a song intro subconsciously displays intentional or unintentional hints of narcissim, an audience member is consciously or unconsciously turned off to the performance.

Narcissim of such note is reserved for those who think highly of themselves, yet are going nowhere, e.g. many garagebands."

In conclusion,

Those of you out there trying to better your craft, you deserve to hear this. I'm not trying to step on your toes.

[BTW, It is very good and useful for you to have at least one person who will give you a non partial opinion of yourself/your group. Friends and family usually cannot give you this sort of constructive criticism.]

For all you writers out there...Keep on shooting for that hit song, keep on shooting for the song to touch millions!

Until then,

Andrew @ AfaraWayland

Where have I been? AfaraWayland HQ has Relocated!

2006-08-08T05:35:00.000-07:00

Hi all,

Sorry for the gap in posts, a few really busy weeks have passed recently.

On top of working with a few of my clients, I've moved my studio/HQ from Murfreesboro/Nashville to North Alabama. The reason for the move is that I have just taken a part-time gig working for WAFF 48 News TV Station, in their production department. WAFF is an NBC affiliate.

Aside from the transition into a life-paced TV news production position, I'm currently modifying a room in my new residence into a mixing/light-tracking room. I've been working day and night, and pretty soon, I might have some photos for you to view.

Also, expect a post concerning room treatment, IOW, how I treated my room. Panels are still being built, in fact, I'm traveling to Nashville today to buy some more materials. I'll show you a how-to DIY for absorbing panels, and discuss why I'm using them.

Until then, or rather, until I can find the time to write, keep checking back for more posts!

~Andrew Wayland @ AfaraWayland

Ahead of the Wake: Dancy-Polk House Recordings: Tracking: Recording Guitars

2006-07-20T21:59:00.000-07:00

Hi all,

Here's the break down on how I recorded guitars...
Here is Eric, the bassist, and Clint, playing a nifty Les Paul.

By now you know that I set up the drums in the stairwell of the Dancy-Polk House Inn. At least 25 feet above the drums, I had the guitarists split out into separate rooms on the third level. Since the bass guitar was running direct into my preamp/interface, I had him share a room with Clint (shown above).
Here is Clint's tracking rig that includes various pedals and a Marshall half-stack. I chose to use Clint's Audix i5 dynamic microphone on the cab during live tracking, yet even at that point, I knew I would almost certainly replace the live guitar tracks with overdubs at a later date. Even with the Audix, isolation from the drums (through the cab mic) was not attainable. Look for another post to come concerning other reasons why I might want to replace original live tracks with overdubs.
Shown here is Zach, the other AOTW guitarist. I set him up in the room across the hall from Eric and Clint. Zach's rig consisted of a few pedals, and a nice Carvin-Marshall combination half-stack. Zach's Beta 57 was put in front of the cab for live tracking as well.

Notice the row of chairs set up in front of the speaker cab. I deliberately set them up this way to absorb some of the sound pressure energy coming from the cab. In short, this is an improvised sound absorber.
After finishing the live-tracking portion of the recording, we tried to focus our remaining time on guitar and bass overdubs. Here is a shot of Zach on one of our final days. Shown in front of the cab now is an Audio Technica 4060 Tube Condenser mic.

Unfortunately, the heat of the day had congregated on the third floor by six o'clock, and naturally we began experiencing guitar intonation problems. It seemed, no matter how often the guitarists would tune up, the very next minute (or measure for that matter,) the guitars were again out of tune. This is the reason why one shouldn't expose guitar wood to extreme heat and coldness: wood expands and contracts in relation to the surrounding temperature. Needless to say, after a few hours of attempting overdubs, we called it quits.
Here is Eric's Bass rig. It consisted of two bass preamps of presumable well renown, and two basses, a custom 5 string, and a custom 6 string.As with the other guitar tracks, I decided that I could get a better sound overdubing the bass guitar tracks. Above is a shot of Eric, Zach and Trey, as Eric is preparing for more overdubs.

Since I had not mentioned it yet, here is the spot I was in the moderate portion of the week:
This sizeable bathroom is located on the third floor, between the two far tracking rooms. Notice the old-fashioned yet luxurious toilet-tank and shower/bath. I was able to fit myself, a card table, computer, recording gear, and a folding chair in this space [surprisingly].

During our numerous session breaks, the group and I took time to lounge in the formal room and flesh out two new tunes.

Thanks for tuning in! I apologize in the delay of this post, I've got several more in the works. Stay tuned to find out what's happening in AfaraWayland!

~Wayland

Ahead of the Wake: Dancy-Polk House Recordings: Tracking: Recording Drums

2006-07-18T21:59:00.000-07:00

Hi all,

Here's the breakdown of how I tracked drums during the week. Shown below is how I initially setup the drums.

After listening to the sounds, I decided to change configurations, as you will see later... I chose this particular spot, not because it made getting in and out of the front door a big hassle, but because it is one of the few rooms with irregular symmetry and very high ceilings.

The worst room shape for evaluating/recording sound is a cube or a rectangle with solid surfaces. Most modern professional studios are built ground-up with the understanding that parallel surfaces are bad. Like it or not, a good/bad room will greatly affect your recording/mixing accuracy.

This stairwell was shaped as three odd rectangles, one flat, one on end vertically (up the stairs), and one across the 3rd story floor, kind of like a capital "C." The unique shape directed sound away from the drums and up the stairs, most of which was probably trapped reverberating on the third floor. The stairs, surrounding furniture, and my own hand-built sound absorbers (shown below), helped to diffuse and absorb direct and reflected sound at drum level. I knew as soon as I saw the stairs during my research that I wanted to try the drums there, and my instinct then paid off very nicely during tracking!

Here is the initial setup:Notice the two Oktava MK-012's out in front of the drums. They are configured in a near-coincident arrangement, aiming across the front cymbals. I've tried this technique before on other kits with success, but the tight floor space and early reflections from surrounding walls captured a sound that I did not want. So, I thought about what I did want to hear, mainly, more of the bottom snare and the hihat. I then moved one of the Oktavas underneath the snare, and replaced the EV Cobalt vocal-condenser above the hihat with the other Oktava. [See farther below]On the toms, you can see Sennheiser 604e(s). These performed excellently in this application. Sennheiser drum products like the 604e, the 602, and the MD 421 get my approval on a regular basis. On a side note, the 441 model is stunning on certain vocals.

A week or so before the tracking date, our options on kick microphones were running thin. The drummer and I had been pondering whether to build a home-made kick mic out of an old speaker. The common ideal is to use a speaker, perhaps 6-12" in diameter, mount it with the speaker face facing the beater or head, and to then run the +/- connections to a preamp. The speaker is basically turned into a microphone.

After much research and thought, I decided that the one spare (car-audio) 10" woofer we had available was too heavy to use. The next idea was a using "kick tunnel," combined with a beater mic. The dilemma was, being limited on kick mic options, how to capture the attack of the beater (that sound that is so prevalent in Speed Metal) and still maintain the body and fullness of the shell resonance? There are many good ways to do this, yet I chose this method.
Here is how I captured the kick drum sound. Seen above is a "kick tunnel." It is comprised of merely an extra kick shell with both drum heads removed, running flush with the shell of the main kit kick shell. Inside the tunnel, roughly six inches off the beater, is a Shure Beta 57. At the rear of the tunnel, I placed an AKG SolidTube. I placed a spare drummer's rug over the top of the tunnel as well.

The function of the tunnel is to channel the sound pressure energy from the beater and shell. The "boom," or "deepness" of the kick is more focused in this fashion (more so than removing the main shell only), because low frequency sound wave cycles are very long in length, (some even need more than ten feet to fully develop!) The kick drum sound, if allowed to resonate and develop when channeled through the kick tunnel, produces what some describe as a full and defined tone.

One thing to remember when using such a configuration is always check the phase relation between the beater mic and the tunnel mic. This is because the sound of the beater will be delayed in reaching the tunnel mic by a few milliseconds. If you were to combine both signals when mixing, certain frequencies of the combined signals would resonate with each other and other frequencies would cancel each other out. The result of which will be what we audio guys call a "comb-filter effect," or a rather "hollow-sounding," kick drum. In order to avoid this comb filtering effect, one must try moving the rear mic around and/or inverting the phase of the tunnel mic in relation to the beater mic. Fortunately, inverting the phase of the tunnel mic, via my Focusrite ISA Preamp, eliminated most of the hollowness that the configuration produced.

Here is an updated view of the drumset, after sight modifications were made:Notice that I placed an AT 4033 in front of the kit, and moved the Oktavas to different locations.Here is the updated snare photo, notice I chose to place mics, both on top and bottom. The bottom mic captures a more of the actual snare mechanism sound than the top mic, but as with the kick configuration, one must also check the snare drums phase configurations when combining both top and bottom snare signals. Again the simple solution is to invert the phase of one of the microphone signals.Here is the actual setup, as seen from the half-way point up the stairs. Shown in the back-ground are my hand-made studio sound absorbers. They are made from the same material that many professional studios use, but perform better and cost less in comparison to Auralex or other absorptive foams. Their construction is quite simple, and I plan to show you a how-to demonstration in the weeks to come.

I placed these so that I could absorb as much high-frequency early-reflections as possible from all critical sides of the drum set. Long story short, I didn't want the drum sounds bounce off the near wall and returning into the microphones (...another example of comb-filter producing phase irregularities.)

As far as a final mic list:

Ovhs: AKG C414 B-XLS
Kick Beater: Beta 57
Kick Tunnel: AKG SolidTube
Snare Top: SM 57
Snare Bottom: Oktava MK-012
Hihat: Oktava MK-012
Toms: 604e
Room: AT 4033
Stairwell Room: SP LSD-2

As far as the recording project goes: Drum tracking is complete and the next phase is recording instrument overdubs.

Thanks for tuning in, more to come soon!

~Andrew Wayland

Ahead of the Wake: Dancy-Polk House Recordings: The Trains Don't Stop

2006-07-17T09:59:00.000-07:00

So I did some research into the area's train schedules. I mentioned earlier my doubt of being able to obtain solid information about them, but as it turns out, I was actually able to find a few helpful resources! As it turns out (a fact previously unknown to me,) there are many people out there whose hobby it is to monitor and watch trains. It appears to fall into a category similar to bird watching. On the web you can find small networks of what are called, "Railfans."Railfans are railroad enthusiasts with a passion for all things train. One Railfan in Decatur played a vital part in helping me get a grasp on the rail usage near our recording site. This Decatur citizen hosts this site, AlabamaRailfan.com. It was hear where I stumbled onto a gem of information!

If you preview his site, you will find a link to a webcast server connection, which is fed by a local rail frequency radio scanner. The man has set up his own radio antenna and scanner to listen to dispatcher and train engineer communications in Decatur. What this meant for me was that I could monitor the train traffic approaching the Decatur train bridge(see below), the central entry and exit point for the rail line in the area, and the adjacent line running by the Dancy-Polk House! After listening to the feed for a while, I found that having a train schedule would be all but useless. Here is why: trains, like planes and buses, can fall behind schedule due to factors beyond operator control. One factor that figures in is the Norfolk-Southern Tennessee River Bridge.As you can see, it is an older steel levitating lift bridge that crosses the Tennessee River. When large pleasure boats and river barges make wake coming through the area, the captain of the vessel enters a request to the bridge/rail dispatcher, and the rail dispatcher "schedules," in a spontaneous rail delay. Sometimes the watercraft must wait for a train to pass, and sometimes a train (or trains) must wait for a vessel to pass. Other delays are caused by the bridge itself, or heavy train traffic in the area. The bridge has problems pretty regularly it seems, and Decatur sees an estimated 50-60 trains pass through every 24 hours. Listening to the AlabamaRailfan.com MP3 stream, I heard each of these schedule-breaking occurrences.

In conclusion to the topic of trains, I found that obtaining a solid schedule of the rail line was next to impossible without listening in on the rail communications, and when listening to them, even then it is difficult to decipher the lingo used by the dispatcher and engineer. One thing is for certain...the trains don't stop...

As for the recording, thus far we've been fortunate enough to escape having any major train/recording issues.

Thanks for tuning in today, we're spending some long days in the Dancy-Polk House this week. Cheers!

~Andrew Wayland

Ahead of the Wake: Dancy-Polk House Recordings: Setup

2006-07-16T21:50:00.000-07:00

Hi all,

Today I traveled to Decatur, AL to start setting up for recording an album with Ahead of the Wake. In my previous post I explained that through a networking connection, Ahead of the Wake secured a seven day slot to record in one of Decatur's oldest buildings. This has to be one of the most unique on-sight recordings I've ever attempted.Two of the group members and I arrived around 8 PM tonight and began unloading gear. Stay tuned for more information on what gear and techniques we used.Also mentioned previously, one of the major obstacles of this production will be dealing with the active railroad line directly across from the front of the site. As sound travels better through solids than gases, I found that the entire house is filled with train engine and friction noise when trains pass. I tried explaining this to one of the band members' girlfriends.

The train tracks are located in a concrete ditch across the street. The buildings foundations are down some feet below the ground and are coupled to the earth and possibly the concrete surrounding the ditch. When a train proceeds across the tracks, the friction from the tracks and train iron, and the noise energy from the engine itself, project through the concrete and earth outward, sequentially making contact with the foundations of the surrounding buildings.In acoustics, vibrations that travel from one physical state to the next do so by what we call "coupling." Its the same principle that makes an acoustic guitar sound big and full. The strings are stretched between the bridge and the nut, and the strings vibrate when strummed or plucked. The vibrations travel to the bridge, which in turn is connected to the guitar body, this transfer of the energy between the bridge and the body is called "coupling," or "acoustical coupling." By the process of coupling, the vibrations/energy are transferred through the solid bridge into the solid and hollow body, and then resonate in the body (bringing out the big and full sound). Perhaps you've seen guitars without soundholes? Soundholes are not necessary for the sound to resonate or radiate from the body. In fact, acoustic sound will radiate out from a solid into the surrounding air, simply by coupling with air.

Of course this is not an in-depth discussion of what actually is taking place, its just a summary of the basic physics of sound.

Try this experiment: get an electric guitar, find a wall, hold the guitar against the wall (wood/paint to wood/paint/drywall), and play.

Result...Instant amplifier!So back to the train problem, not only do vibrations and energy transfer through the solid states in the ground, but vibrations in the air arch over the ditch and into the surround surface areas. Especially low frequency sound waves, such as that created by train and other machines. Vibration/energy at sub-20 Hz is inaudible, but one can definitely feel the effect. On a side note, that's what make true sub-woofers so popular with the mall-cruisers or home theatre enthusiasts...you can feel the sound energy. High frequency sounds are more directional in nature, low frequency sounds seem to go where ever they feel like.

A note about sound:
While it is true that sub-20 Hz frequencies are inaudible, sound is very rarely made up of one frequency. The sound as we hear it, (voices, car horns, and locomotives) are made up of many frequencies. Our human hearing ranges (at best) from 20 Hz to 20 KHz. A passing train creates noise from deep in the sub-20 Hz range to high above the 20KHz range. The noise we hear are the frequencies within our audible range.

Unfortunately, I cannot isolate the recorded signals from the train noise, should we be tracking during the passing of a locomotive. Further, I fear that the Department of Homeland Security might have tightened the reigns on diverging cargo train schedule information to ordinary citizens (it would seem appropriate to do so). I say this in all seriousness, terrorists threaten us even in North AL.

I will try to obtain the train schedules, but in the event that I am not allowed the information, or cannot find it, the band and I will have to try tracking a cut between trains.

So here are some shots from the initial setup. Finalization and perhaps even tracking will begin tomorrow.

Thanks for viewing, and look forward for more posts-

Andrew Wayland @ AfaraWayland

Gearing up for another project: The Dancy-Polk House Inn!

2006-07-10T00:30:00.000-07:00

Hi all,

For some time now, I've been doing FOH live sound gigs with Ahead of the Wake. In an earlier post I mentioned that the band had asked me to consider doing some studio recording for them. As it turns out, the plans are now final, and tracking begins next week.

The most interesting part of this engagement has got to be the location!For a period of about seven days, this historic mansion in downtown, Decatur, AL will be converted into a recording studio. Built in 1829, this massive structure withstood the Battle of Decatur in 1864, when Union troops burned Decatur and it's strategic railroad bridge to the ground, leaving only the Dancy-Polk Inn, the Old State Bank Building, and the McEntire House unscathed. It is reported that Union troops actually used the inn as a base of operations and temporary lodging.

The house itself is approximately 4818 square feet. Here are the dimensions of the interior: Foyer, 11 X 21; Living Room, 17 X 21; Dining Room, 17 X 21; Kitchen:, 15 X 21; Great Room, 20 X 29; Utility Room, 13 X 16.8; Study, 17 X 21; Bedroom #1, 16.6 X 20; Bedroom #2, 14.7 X 14; Bedroom #3, 17 X 21; Bedroom #4, 17 X 21. As you will see in the photos below, all of the ceilings are fairly high...perfect for the situation.

A few people have even asked if we feel like Led Zepplin recording in a castle...well, yeah!The house is actually a fully functional Bed and Breakfast establishment. Further, it is an officially accredited historic site by the Nation Register of Historic Places.

As you walk into the house, the first thing you see is the main stairway. It ascends at least 20 feet and joins the second and third levels together. I think this area would be a really neat place to try the drumset. In fact, I'd like to mount some room mics from the top of the third level, looking down onto the drumset on the second. There are plenty of other big rooms to track drums in, should the stairwell prove ineffective.Here are some shots of the actual rooms, as we witnessed on the day we went in for research.Shown are Trey Allred (drums) and Chris Haddock (saxophone), helping to give me some perspective on scale.
One other interesting feature of the house is the false door...
Well, in all truthfulness, it is a real door, it just doesn't go where you might hope it would, should you want to exit the building late at night. I can imagine that there are some funny stories surrounding this. Apparently, the caretakers use this quick exit to dispose of impolite guests and household waste... ; )

Unfortunately, one of the few problems associated with this location happens to be its historical location. Directly adjacent to the property: The Courtland and Decatur Railroad. Ironically, the feature that most likely saved the building from arson 142 years ago will possibly present a considerable problem for the band and I as we attempt tracking.

Nevertheless, expect another post concerning this location in the next few weeks, as we record Ahead of the Wake's first studio album.

Thank you all for tuning in!

~Andrew Wayland

Happy 4th Everyone!

2006-07-04T21:57:00.000-07:00

All Photos © Andrew Hull 2006

Bonnaroo!

2006-06-19T16:43:00.000-07:00

Hi all,

At the last minute, a relative of mine graciously passed on some tickets to the annual weekend festival in Manchester, TN. Here are some of the sights I saw there.The one thing you can spot in every photo is...people. Lots of people at Bonnaroo this year, over 70,000. Also, tens of thousands of people walking around in dry dirt produces massive amounts of dust, I found.Welcome to Planet Roo.

This was the least of the strange things I saw at Bonnaroo...here is possibly the strangest...

Can anyone tell me what is going on here? I'm too afraid to ask...This gal's got a plastic tank stuck on her bum...And the peacock feathers?

And my favorite moment of them all, the "hippy grinder." Lured by the odor of patchouli oil and free balloons, unsuspecting hippies are lured to their doom...
Of other interesting note, I did see a glass blower at work. Here he is creating a fine beverage mug.
Here is Robert Randolph, performing at the Other Tent.

The highlight of Friday night was Tom Petty and the Heartbreakers. I was especially suprised when Tom brought out the band's "little sister," Stevie Nicks. They put on an awesome show!
"Hippy Grinder," redux...Actually, these are mobile mist machines, the salvation of those stuck in the sweltering heat...
Perhaps my favorite performance of the weekend was early Saturday evening, Beck! The show included a live puppet show! Actually, while most other bands had the live video feeds of their performance pasted to the two "megotron" screens flanking the stage, Beck had master puppeteers mimick their stage show, then fed the puppet show live to the two screens. The coolest part about the puppets was that they were dressed exactly alike the real stage performers. Sweet!

Later in the evening, in the Comedy Tent, I saw the Upright Citizens Brigade preform "Air Conditioned Acid Trip."

I pity da fool who smirk at my oversize "ok"...
Enter the Dragon...
And of course, to finish the night...Radiohead...

Sunday, the only show I attended was Bela Fleck and the Flecktones.

This was my first Bonnaroo experience, despite having lived in Middle Tennessee for a few years.

Thank you all for viewing!

~Andrew Wayland

Live Performance: Labeling Deloris @ Ziggy's 6/15/06

2006-06-15T11:41:00.000-07:00

Hi all,

After attending Labeling Deloris' CD release party, I followed the band to Winston Salem, NC to see their follow-up tour performance at Ziggy's.Here is a mural I saw driving to the show.
Ziggy's is quite a nice establishment compared to the dives I've been patron to. It has several spacious subdivisions (return of the Rush song...). The main room is tiered thrice, with professional sound and lighting.

FInd Ziggy's on the Net here: Ziggy's
The band rocked the house that night! So well, that they received an invitation to return again in July.Special thanks goes out to Justin Dionne, of Salisbury, NC, for running lights during the show.

Check out Labeling Deloris's website for more info on their summer tour: LD Tour Dates

~Andrew Wayland @ AfaraWayland

Labeling Deloris: CD RELEASE PARTY, Wilmington, NC

2006-06-13T11:40:00.000-07:00

Hi All,

3...2...1...Blast Off! Labeling Deloris has official kicked off on their summer tour! Last Monday, I traveled a fair distance to the Wilmington, NC on the Atlantic coast, to attend the "For the Sake of Deloris, CD Release Party." I stayed with the band for a couple of days, then followed them up to Winston-Salem on my way home Thursday. For those of you who haven't heard, LD has officially released the EP we produced this spring, called, "For the Sake of Deloris." Sales at Awarestore, For the Sake of Deloris, reached all the way to number 1, and spent 3 weeks in the top five. Presale orders were sent out June 12th.

So, what does the album look like?

The headline photo is a compilation I put together to display LD's new EP and related promotional materials. Shown at the top, their new poster; in the middle, the EP's CD label/face; on the bottom, For the Sake of Deloris EP, front and back covers. Here are some links to articles about the new EP in the Currents and Encore.The Soapbox is a combination bar. Can you guess what the bar is combined with?...

Give up? Beer, pool tables, video games, internet cafe', and yes, washing machines.

The establishment is located on Front St., in the popular tourist district, on the direct opposite shore of the historic Battleship North Carolina memorial.A few miles down the road stands the Atlantic ocean, which at the time, had been unsettled by TS Alberto, but fortunately, I only saw light rain during my stay.
Here is the band as they set up for the show. Above you can see Jason, Jen, and Brad checking monitors and getting sounds. Below, shows Andrew and Deloris, in front of their new banner. At the last minute, the band requested that I ride the faders (mix) their show. What a treat! Here is a quick shot from the mixing station. The band played with a group from Athens, GA: Last November.

What an awesome night! Tomorrow, LD @ Ziggy's in Winston-Salem, June 15th, 2006.

Thanks for tuning in!

~Andrew Wayland @ AfaraWayland

Live Performance: Ahead of the Wake 6/09-6/10 2006, Fastlane

2006-06-10T11:25:00.000-07:00

Drama with the DPD...

Hi all,

After a week of traveling here and there, I finally have some time to catch up on blogging. Last weekend I engineered yet another live performance for Ahead of the Wake. Their most recent performance kicked off the grand opening of the Fastlane Bar, formerly known as the Dugout, located in Decatur, AL. Fastlane is a run-of-the-mill sports bar, equipped with pool tables, trivia machines, and of course, beer.
For some reason, I got on a black and white kick this time. Here is shots of the scene: at top, the bar; at center, the setup; at bottom, the performance. I used my newly acquired 100' audio snake for this gig, and of course my usual setup for the musicians. Of special note, a few anomalies took place over the weekend. One of the surprises the band put together, besides adding some new material, was a special acoustic section to one of their sets.
Shown at top is Clint, playing an interesting resonator guitar. Below is a shot of another special feature to the performance. Our friend, Chris Haddock, accompanied the group during "Josie," a popular Steely Dan song. How many bands do you know who can pull off Steely Dan tunes?

And finally, what small town weekend in North AL would be complete without a police drama sequence.
After the show on Saturday, the band and I went to the Waffle House for mid-morning nourishment. Zach, has the duty of pulling the trailer, and as he was trying to enter the WH parking area, a local police officer swooped in an signaled him to stop. The picture above is actually what was happening in the far end of the WH parking lot. Apparently, the weekend brings out the worst in small town nightlife, and one particular citizen found himself being examinated as to the alcohol content in his person. As we watched from a distance, one, not of our party, was arrested for D.U.I.
The band and I, excluding Zach, who was being examinated himself for "failing to yield," sat on the curb and pondered the situation. Zach was not intoxicated, nor had he been drinking that night. Unfortunately, it is unwise to argue with authority in respect to traffic law. John Mellencamp knows...

You can vaguely make out the truck, trailer, and police cruiser behind the row of cars.

After about ten minutes of waiting, the officer finally let Zach go, but not without a traffic citation. Needless to say, the rest of our evening together was spent trying to figure out why Zach had been stopped, and the righteousness of it all.

The headline picture shows the aftermath

Despite this one unpleasant anomaly, the weekend performances were very good. I'd like to thank the DPD for making our nightlife safer...especially from people who fail to yield.... ; p

Thanks for tuning in!

~Andrew Wayland @ AfaraWayland

Payne Live @ Sweetwater Video!

2006-06-07T00:04:00.000-07:00

Get this video and more at Payne, Live Hip Hop
Oh Snap!

One of the producers working with Payne and I on his upcoming record, has released this video of one of the shows we did at Club Sweetwater in Murfreesboro, TN. Look at my previous post of his first show at the club to view photos of this performance .

Apparently uploading videos onto a Myspace page allows you to get pre-written code for inserting videos and things of that nature into say...Blogger. Cool!

This particular video shows an audience perspective of what the show was like that night. Can you sense the energy? I was there at the mixing station, and my earlier blog description of the night does no justice compared to this video. The sound quality of the clip is not excellent, but it does express the necessary parts.

Now that I have a trick to insert video into my blog, I plan on doing this more often.

Thanks for stopping by San Diego.

Andrew Wayland @ AfaraWayland

Live Performance: Ahead of the Wake, Featuring Special Guest, Wayland; Alabama Jubilee, 5/27/06

2006-05-31T15:49:00.000-07:00

Hi All,

Here's the scoop on the Ahead of the Wake Alabama Jubilee Performance last Saturday. (The night before, the band had performed alongside special guest Pete Car.)

This is a shot of what we saw driving up. As you can see, the Jubilee spares no expense for a professional sound setup. I don't recall the name of the company involved, but I can tell you that they did look the part.

Looking into the sound mixing tent one can see one of their consoles, a Midas to be exact. The days of the festival are full of musical performances. These guys don't get much of a break.

The band in the slot before us was onstage at the time of this photo. Next up, Ahead of the Wake!

Behind the stage lies a 500 yard string of food and trinket vendors. Its the only place in town where you can buy a polish sausage, gyros, fried catfish, cotton-candy, dippindots, onion petals, turkey legs, (and of course) chicken on a stick in one place.

So here we are! Ahead of the Wake and I rocked out to classic rock tunes such as Jessica, Two Tickets to Paradise, Cajun Moon, and Shakedown Street. We also brought out a few songs from Ahead of the Wake's upcoming album. As stated in a previous post, I'm not a musician of Ahead of the Wake. I merely filled in for Clint, as he was out of town that day. See the original members here: Ahead of the Wake.

Here we are rocking out. I am located on the right, across from Eric, the bassist, and Zach, the guitarist. Trey the vocalist and drummer is hidden in the back. I should mention now that Clint allowed me to used his gear during the performance. He has a very nice Fender Deville amp, accompanied by some kick-a$$ pedals, and when I combined them with my American Strat, rock happened. I was definitely spoiled that day.

A little known fact about me in Middle Tennessee/Nashville is that I was actually a musician before getting into producing and engineering. The best producers and engineers are usually those with musical backgrounds. Back in my home town in AL, I did the garage band thing, played in a few bands, etc... This experience is very helpful to me when trying to communicate and relate to the folks on the other side of the glass. It has been said that music is a kind of language. I guess it may be as hard to learn as say...Chinese?

I am extremely thankful to Ahead of the Wake for considering me worthy enough to play with them that day. Truth be told, since I stopped playing in the last band I was in late last year, I haven't really played much...I've been too busy recording others. When the band asked my thoughts on the idea the week before the show, I felt a familiar excitement. The "Limelight" is definitely a drug of some kind. When your in it, its ecstasy, when your away, you want to be in it. Since I decided to take the path of engineer and producer, I have had to come to terms with the meager allowance of lurking in the background. It is times like these that I get to actually poke my head back through the curtains and live it up just one more time. Thanks guys. The week of scales and finger blisters was worth every minute.

Sunset at the Jubilee. As you can see, the festival is quite popular. It draws hot-air balloon pilots and fans from all over the country.

The band and I played a short one hour set, packed up our stuff, and headed home. I'd like to thank all of you who came out for their performance. You are why we do what we do. My special thanks to Clint and the rest of Ahead of the Wake for another go at my 15 minutes of obscurity. Playing with you guys was a blast.

To all those who missed my last post, look for Live Performance: AOTW, featuring special guest, Pete Car.

The band and I are heading to the dive in T-town again (Harry's) this weekend. This time I'll remember to bring my own soap and towels... ; )

Thanks for reading, everyone.

Andrew Wayland @ AfaraWayland

Live Performance: Ahead of the Wake, Featuring Special Guest, Pete Car; the Brick, 5/26/06

2006-05-28T17:27:00.000-07:00

Hi All!

Its been a busy Memorial Day Weekend! On Friday, my friends, Ahead of the Wake, performed with with a very special guest. The man on the right (above) is none other than Pete Car. If you've listened to any southern/classic rock in the last 30 years, chances are high that you've heard some of Pete's extensive contribution to popular music. Keep reading for more info on Pete.

Saturday, I performed with Ahead of the Wake at the Alabama Jubilee, an annual hot air balloon festival. Clint, one of Ahead of the Wake's guitarists, could not attend the show, so the band asked me if I could possibly dust the rust off my guitar strings and play with them. Needless to say, I was very excited to receive such an invitation! In a day or so, I'll show you more about my recent limelight experience.

Only now, on Memorial Day, do I have the time and energy to let you all in on the fun we're all having at AfaraWayland.

Ahead of the Wake played to a full house Friday night. Fortunately, I escaped the night without having any intoxicated bar patrons fall into me or our gear. The reason for the house being packed is simple: Ahead of the Wake pleases the crowd. Where they employ me (as the live sound engineer), I help the band on stage sound the best it can to the people in the audience. With out them, my job would be useless, but together, we make quite a powerful team.

Bar owners only want bands to do a few things. Entertain their patrons, encourage the bar's business end, and do so in a professional manner. If the fact that the band is playing a club every weekend isn't enough to make you a believer in their worth as entertainers, ask some of their fans how they feel about the band. These guys are musicians, striving to make music that people enjoy. Apparently, their formula for rock is striking up some attention.

Shown in the pictures above is the pre-show stage setup and the pre-show crowd. Once the music starts playin', everyone migrates to the stage area and packs in pretty tight.

Here again is Pete Car, shown on the right, jamming away at a popular J.J. Cale tune, Cajun Moon. The shot below it is a picture of the crowd, swarming the stage that night.

Now for some info on Pete Car. He's been playing music professionally since the 60's. He has worked with some groups you might recognize.

The Hour Glass
Duane Allman
The Allman Brothers Band
Willie Nelson
Rod Stewart
Bob Seger
Bob Seger and The Silver Bullet Band
Art Garfunkel
Paul Simon
Hank Williams, Jr.
Cat Stevens
Joe Cocker
Joan Baez
Barbra Streisand
Wilson Picket

Find his allmusic.com credits here: allmusic.com-Pete Car

Read his interview with Izotope, a popular music software company here: Izotope-Pete Car

Visit his website here: playthatguitar.com

I consider myself truly blessed to have worked with such a legend of the studio recording world. Its not every day that we get to witness their talent first hand! Shown above is Pete on the right, me on the left, during one of the set breaks. Pete is an awesome guy.

"The Hungry Hippo Kick Resonance Fluxuator."

The Ahead of the Wake show has evolved somewhat since I started working with them. I don't think this evolution has happened on my account, but rather, the band itself has come up with some new ideas for an entertaining stage performance. When you go to a rock and roll show, don't you want to be entertained?

I'm not talkin' glam rock by any means, but if the performers look like they are having fun, then chances are, the audience will have fun with them. Shown above is an "improvised performance enhancement." The Hungry Hippo Kick Resonance Fluxuator. Little stuff like this can go a long way when you add a happy atmosphere and good music.

Every now and then I get a most extreme challenge of my own. This one, thankfully, did not involve a Japanese obstacle course.

Recently, the band and I have been recording live to 2-track mixes of their live performances with a CD-recorder. One of the band members owns the Zoom stand-alone multitrack recorder shown above. A few days before the show, the band suggested that I use it to record this particular show, and I agreed to do so, reserving the CD-recorder as a back-up.

Long story short, I had downloaded the manual earlier in the week (nearly 200 pages), yet it wasn't until Friday that I was able to play with the unit. I had to learn the basic recording functions before the show began and managed to capture over 3 hours of audio. Challenges and pressure are your friend, but learning how to learn is priceless.

After the barkeeps ushered everyone away, and the equipment had been stowed, we took our traditional after gig band photo.

We all had a wonderful time that night. I especially want to thank my client Ahead of the Wake, and of course Pete Car, for allowing me to take part in such an event.

Thank you all for tuning in, stay tuned for a review of our show Saturday at the Alabama Jubilee!

Andrew Wayland @ AfaraWayland

Live Performance: Ahead of the Wake 5/13/06; Pascagoula, MS

2006-05-16T19:37:00.000-07:00

Hi All!

Last weekend Ahead of the Wake rocked a high school prom in Pascagoula, MS. I went along with the band as their sound engineer. Pascagoula is located in South Mississippi. It's an eight hour drive from my home in Middle Tennessee. Fortunately, I met the band in North Alabama a few days before the trip, so the driving time was actually six hours.

Here's a shot of the folks in my car on the way down to MS. Shown is Clint, one of Ahead of the Wake's guitarists, riding in the back; Trey, the band's drummer and lead singer; and me in the driver's seat, sporting what one of my brothers calls "baby killers." They are extraordinarily large, no?

Along the way, during one of our many gas stops, we took the time for a multifarious band photo. Many bands use brick walls or train tracks. Not us baby!

The private high school that booked Ahead of the Wake paid for our stay at (in their words,) one of the finest hotels in the area. I humorously refer to the lodging as the "LaFonda" Inn, (N Dynamite reference). As you can see, the LaFont Inn is a relatively above-par accommodation. The soap there was extra moisturizing too, but alas, they did not provide for us complementary chapstick!

Despite the major damage the area suffered during Katrina, we did not see much of the aftermath in the area we stayed/performed. Here is the building, Scranton's, in which the prom was held.

The upstairs portion, rented out for the dance, was quite spacious. It seemed to have a reverb time of at least 1.5-2 seconds. As you can see, it had fairly high ceilings, a spacious extension across, extending over 100 feet to the back wall. Two problems that I had to deal with was a lack of convenient power outlets, and an active railroad line directly out the back door. To make matters worse, the building was located adjacent to a railroad/road crossing, so, every train that whizzed past was preceded by very loud locomotive horns. To give you a visceral image, the rear of the building vibrated when the trains passed.

Here is the stage setup after 3 hours of stowing gear and checking the sound. I used the band's PA system in combination with some of my microphones. I ran two guitar mics, an Audix i5 and SM 57, the bass direct to a dBX 286 compressor I had lying around at home, the kick mic into another 286, two Oktava MK-012's on the overheads, and three vocal mics for the singers. Quite frankly, the 286's really appreciated getting some use. I honestly don't ever consider using them for my recording work. The bassist's Mackie CFX mixer did all the additional preamplification and mixing.

Here is another shot of the band on stage. (I have never been able to fit them all the same frame!)

The band really pleased their crowd during their performance. Of special note, upon a strange request by a party-goer, the band pulled out a rather spontaneous rendition of the RHCP's "Under the Bridge." No matter who your are, no matter how strange, awkward, or unprepared you feel, take every opportunity you get to help others. What may seem like a small and insignificant trifle to you, may in fact be something very significant to another. In the long run, helping others will give you a much more fuller life than money, success, or other fleeting things. There's my $.02 on life for anyone, performer or patron.

And finally...after the fruit punch is gone, and the guys in "Dumb&Dumber," inspired tuxedos have disappeared...the time comes for the band to pack up and head home. Before we left, I snapped on final shot of the band.

My thanks goes out to the Ahead of the Wake, for giving me an interesting mini-vacation, and being all-around fun guys and friends. You can check out some of their tunes by visiting their Myspace account here: aheadofthewake.

Cheers to all, and I hope you return to view more of my posts. Heck, why not start another audio discussion? I just got an informal clearance from Lynn Fuston, of 3D Audio in Franklin, TN to post my final installment on the analog summing debacle. Yay! Any ideas for the next topic?

Andrew Wayland @ AfaraWayland

"For the Sake of Deloris" Its Done!

2006-05-08T23:06:00.000-07:00

After a few months of hard work, my client, Labeling Deloris, has arrived at what is perhaps their finest work! At the end of this month, Aware records will start distributing their new EP, "For the Sake of Deloris."

For Mastering and Duplication, the band chose to use my studio partnership with Disc Makers. I recommend Disc Makers services to all my clients. A great sounding record deserves a great looking package, and they are professionals at what they do. As a bonus, using a referral from a Disc Makers studio partner, the band receives special discounts on their purchase. It is a great deal all around!

This project has been the story of my life from January 15th to May 8th this year, and both the band and I are extremely happy with the way it finally came together.

***The official release date is now June 13th, 2006. Aware Records will start taking pre-orders at the end of this month. You can look the EP up here: Aware Store.

I want to thank the band, my parents, my supporting friends and family, and most of all my Holy Father for helping us make this happen!

Andrew Wayland at AfaraWayland

Melodyne: Sweetening a Vocal at 33,000 feet.

2006-05-02T11:42:00.000-07:00

Can 33,000 feet of air make your vocals sound better? I don't know, but maybe that is why some mic preamps have an "air" function dial included in their design.

The story for today is that I just returned from Monterey, California. I'm currently finishing up my EP project with Labeling Deloris, and I took the long return flight from San Jose to Dallas as an opportunity to sweeten up some additional vocals for one of the EP's tracks. I imagine that only a few engineers and producers can say that they tuned a vocal in flight. I mean, lets face it, "ecomedy class" is not your usual studio atmoshpere. It helps to have some headphones that isolate your work from the ambient noise, not to mention the limited space involved.

The program I use for tuning is Celemony's Melodyne. I can't express how much time and money it saves me and my clients. Having Melodyne, I can do tracking, focusing on the performance, without worrying about bad notes here and there. The days of doing take after take just to get perfection are over. It can be said, however, no one could really tune like this 30 years ago, and likewise today, talented vocalist still make the magic happen. I'll take an excellent performance over an over-tuned vocalist anyday.

For me, Melodyne eliminates the need for extra days in the studio. For my clients, it means less overall cost, and less frustration trying to get that one perfect take. All the vocalist has to do is sing the parts through a couple times, focusing on emotion, phrasing, and enunciation, and I take care of missed notes. I call it "sweetening," for obvious reasons.

Here's some more shots of me in the plane. The flight from San Jose to Dallas is roughly 3 hours or so. Plenty of time to tune a few tracks.

You all can check out the benefits of Melodyne here: Celemony.

Thank you all for reading, stay on the look out fro more posts!

Live Performance: Ahead of the Wake 4/20/06

2006-04-21T16:09:00.000-07:00

Clint, Trey, Eric, and Zach___All photos, Â© 2006, Andrew Hull

Hi All,

Yesterday, 4/20/06, Ahead of the Wake, recently noted in an earlier post, performed at the Brick in Decatur, AL. This time their plan was to simply rock the house rather than to record..Soo that they did! You can find Ahead of the Wake on myspace here: Ahead of the Wake.

Here is the basic setup. Amps, Drums, the works.

The band's mobile sound rig consists of twomammothh JBL Dual 15"+Horn mains, two 402 Crownpower ampss and a Mackie CFX mixer as shown below.

For the drums, Trey brought in a simple PZM condenser mic and place it on a pillow inside the kick drum. I put up a pair of Oktava MK-012's over the drums, in a near coincident pair configuration. Though the room was a small one, it actually helped to reinforce certain parts of their sets.

Here is Trey at the drums. I used a Beta 57 for Trey's vocals, as shown in the photo.

So, once again, Ahead of the Wake rocked the house! An interesting feature of the night was the presence of a local record label owner. This goes to show that Ahead of the Wake is making waves.

On a side note to anyone working in an environment that servesalcoholl...watch out! After the show I had a face to face encounter with pain. One of the bar patrons, exceedingly intoxicated, fell into my nose as I was rapping up cables. OUCH!
My nose is still soar, though myinitiall feeling is that there is no serious schnozel infracture. Anyway, to everyone who is soberish in a bar, keep your head on a swivel! You never know when a drunken clown will fall on you. No hard feelings for that guy, though, it was an accident,that'ss all.

The band has let me know their desire to continue our relationship, and even mentioned bringing me along for a road date this May. I'll be sure to keep you all posted as this band progresses and I gain some more interesting experiences with them.

Thanks for tuning in!

Andrew Wayland

Live Performance: Lackluster

2006-04-17T15:34:00.000-07:00

Hi all,

The day after I mixed Payne's Live performance at Club Sweetwater, I went across town to mix another live performance. Lackluster is a local Christian Rock group, that delves into Latin and Blues. Here is a shot of the room farther back.

As you can see, this is not a very large space, nevertheless, at show time 50 people crowded the front of the stage. The venue is an upstairs Christian Ministry/Coffee Shop called Bonhoffer's, located in Murfreesboro, TN, right across the street from my Alma Mater, MTSU. It seems to be a fairly popular hangout for young 20-somethings, especially when musical performances are involved.

The sound equipment for the night was lent to us by Ben Williams, a mutual friend and live sound engineer from Huntsville, AL.

A scary thing happened before sound-check though...

Shown here is John Warner, a Mufreesboro native and percussionist for Lackluster. About an hour before sound-check, I called Andy P. Scott, the Lackluster lead guitarist, to check on whether the sound-check would proceed as scheduled. As I spoke with him he informed me that John had just been involved in a motorcycle accident. Yikes! Fortunately, he escaped with minor road-rash and torn clothing.

The mixing board used was a standard Allen and Heath GL3 as shown below:

The speaker setup included giant Community Cabs and Subs:

The Mic's used were standard SM 57's, SM 58's, a Beta 52, and two Oktava MK-012's.

...on a lighter note, DARN THOSE PESKY SQUIRRELS!!!!

Unfortunately, I was too busy doing mixing during the performance to stop and take pictures of the crowd in front of the stage. This is a trade off that is inescapable sometimes. The bottom line for me is that everyone enjoys the show, and that I do my job well, so that no one is distracted from the musicians. Taking photos must always come in second or even third.

After mixing the Hip Hop act on Wednesday, and the Christian Rock act on Thursday, I settled down to finish editing and mixing my client Labeling Deloris. Their new EP should be in stores at the end of May.

Thank you all for tuning in!

Andrew Wayland

Live Performance: Payne at Club Sweetwater...Again

2006-04-13T23:45:00.000-07:00

Hi all,

Its been a busy few days here @ AfaraWayland!

4/13/06 (last Wed. night) Payne performed again at Club Sweetwater, in the continuation of last weeks music competition.

The show was a success! Here I am in a shot taken after the show. All I can say is that I (honestly) was not under the influence, but my eyes sure don't tell that story do they...Working under the influence of alcohol is never a good idea in any profession, so I stayed soberish that night. I'll admit I had three beers between the hours of 9 p.m. and 1 a.m. But remember, even in the club environment, a sound engineer must be clear headed and alert. So I make it a point not to tie one on until after the important stuff is finished.

I had a lot of fun that night and saw a few firsts for my lifetime. This was of course the second Live show I mixed for Payne and his crew. The setup for this show was the same as last week, the bar hired a DJ and a Live Sound Reinforcement company. Tim Veach of 2 Sound Productions setup and performed the general sound engineering role, and was nice enough to step aside and let me mix when my client performed.

Here are some shots from before the show. In the top photo, you can a pattern emerging, can't you? Well, the entourage/crew that my client is now affiliated with is called "Da Movement." This is a close-knit group of artists and producers pulling toward their unified goal of success. Notice the similar air-brush t-shirts...they have had different shirts made for each performance. The front-line performers are all wearing an elaborate version of the shirt (see pictures for example).

Can you spot Payne above? He is on the left.

Ever since Payne advanced from last week's semifinals, Payne, his producers and I all knew that Wed. would be an important night. Unlike last week, Payne's set was required to contain more material, so naturally they hired me to mix, remix and polish three more backing-tracks. They retained "9mm," from the last week, and had me prepare "Call Me," a sexy ballad from our Darkhorse sessions, "Snap It Up," and "Get A Clue," two highly commercial Hip Hop tracks. The performance also featured the song, "I'm So High," a song that I had previously mixed for Payne.

So, all in all, a little over half of the show CD came from mixes that I have done recently, and it was an extra special treat to see people moving to tracks that I helped make! The show CD was organized by Ishmael Jones, one of Payne's producers. Ishmael collaborated with me on what parts he wanted in the songs, i.e. beats, music, hooks, and ad-libs, and then he took my mixes and combined them into a 35 minute show.

The show itself was fantastic. Spectators saw organized yet fluent movement on stage at all times. There was nothing awkward or insecure in their stage presence. These guys came to play. These guys excited the crowd.

The show was treated very seriously by all of us involved. This photo shows Payne in a back room, preparing to do his thing and discussing the final performance plan with the crew.

Once the group took the stage, the seats that had been filled in the back of the bar earlier became empty. Everyone was making their way to the stage area. FYI, Payne received the short end of the stick that night, and had to perform last. That didn't stop 50+ people from sticking around for the show.

One of the firsts I mentioned earlier I will detail now. Sometime during the second song I looked up to the bar's second floor surrounding the stage to see two women gyrating to the music. As I suspected, this was part of Payne's show. Tim and I giggled quietly behind the console...I did not take a picture of these gals, but try to image any popular Rap video of the day and you'll get the picture. When "Call me," came up in the show, somehow a lady from the balcony had worked her way onto the stage, where she and several other females were greeted by roses and Moet.

This is an example of how well Payne and his crew had prepared their show.

Unlike last week, I was able to steal away from the console for a moment and snap few pictures. Later I got one of the performers to take some pictures from the stage.

Unfortunately, a talented R&B singer duo won the competition. What matter to us is that we had fun and gave it our best shot. By no means will "Da Movement," stop. Hence its name. I truly believe these guys are going to succeed.

The guy on the bottom row could give even Arnold a run for his money, don't ya think...; ]

Well, thanks to all who helped make the show happen, and thank you for reading!

Look for more posts to come!

Andrew Wayland @ AfaraWayland

Studio Recording: Payne @ Darkhorse Studios 11/19-21/05

2006-04-11T03:21:00.000-07:00

Hi all,

Last week Payne, my Hip Hop client, managed to eliminate the local rap competition and move on to the next round (see last post)! I've been working hard for the last 9 hours now trying to polish off some more tracks for this weeks follow up performance. Right now I'm bouncing some EQ'd tracks, so I figured I take this opportunity to tell you a little more about my history with Payne.

I must say that Darkhorse is one of the finest studios I've been privileged to work with. It offers three gorgeous studios with top-of-the-line gear, and a beautiful farm to boot! This is a picture from the loft on the third level. It looks down into the Lodge.

So Payne and his producers come to me early last fall and ask for my help in engineering their tracking project. They pooled up some cash and set the plan in motion, and ended up settling on Nov. 19th through 21st as a do or die weekend session. All in all, we tracked vocals for 11 songs, and had a wonderful time.

I was not the only engineer on this session, given the fact that the clients planned to make as much use of their time as possible...this meaning that they wanted to record around the clock for the entire weekend. My colleague, Keith Moyers shared the engineering responsibilities with me. I would record a couple songs, and then Keith would take over and run things for a bit. Recording around the clock also alludes to the next picture.

Here I am at the Pro Tools Station. As you can see in the background, we used two Amek 9098 Pre's and a Tube-Tech Compressor. This picture shows me on the 20th, wearing a bandanna, in true Hip Hop recording fashion.

Here is Keith at the station. Keith even recorded me while I laid down some classical guitar tracks at the suggestion of Ishmael Jones, one of the producers.

Here is me with the guitar:

Here is a shot of Payne being attended by one of his friends. In the background you can see Ishmael, shooting video and supervising the session.

After the session came to a close, we all took the opportunity to take some photos in the Lodge, the grand room of Darkhorse. Shown are the Producers, flanking Payne, and Keith and I seated at the console.

Thanks to all for reading!

Look for more posts to come!

Andrew Wayland @ AfaraWayland

Live Performance: Payne at Club Sweetwater

2006-04-06T03:28:00.000-07:00

Hi all!

An aspect of AfaraWayland Productions that you may not be aware of is it's diverse clientele. Last year, I began working with James Payne. Payne is a Memphis native and a Hip Hop artist with big talent and extreme potential. After we did a few demo tracks for his upcoming album at my place, Payne graciously tapped me to be his engineer, and our relationship has been solid ever since. Look for more posts on my work with Payne in the future.

A mural of Payne, painted by a friend.

Long story Short...

Last night was the culmination of a few days of intense work. Payne called me up Monday and told me he needed a song prepared for a Wednesday night performance. Basically, a track we recorded last year at Darkhorse Studio, needed to be mixed and polished so it could serve as a track he could rap over during the performance.

The song I mixed was "9mm." This track features a wide range of Payne's vocal abilities, from his sweet and melodic bass hooks to his raw "Twista-fast" rhymes, along with, you-guessed-it...gun blasts.

Besides the basic music tracks, I mixed in Payne singing and ad-libbing to thicken up the sound. If you catch any commercial Hip Hop these days, then you know the "hooks" or choruses are big and layered. I've heard of mixes that have up to and over 12 layers of the same vocal take, but for this application, I only used 3 layers on hooks the and then a double layer on the ad-libs. One must be very careful when layering anything if the information contained in the layers are phrased slightly differently on each pass. If they are not all "together," in time, the product of the combined layers will be an indiscernible mush, or at best have a quirky chorus-type of sound. The worst case scenario may be what we engineers call a comb-filter. Taking this into consideration, I made sure to match and phase align all of the vocal passes used during the mix.

Here is Payne with a few friends. He is a very popular guy.

He is shown circled above.

Tim Veach of 2 Sound Productions was hired by the club for the live sound set up. There was also a DJ spinning records, as shown in the picture below. This was the sound mixing station.
At Payne's request, Tim graciously allowed me to mix Payne's performance. When Payne went on stage, I took over the sound mixing responsibilities.

The crowd got very thick by the end of the night. When Payne and his entourage took the stage, they took it in a force of at least 14 people strong...What can I say but...Wow! This was my first Hip Hop live mixing situation, yet the energy coming from the stage and the crowd made my job all the more enjoyable.

In my honest opinion, Payne and his producers have a winning combination. They have the talent and they have the drive to succeed. Even more so, they are good people who don't fall into the popular culture stereotypes. If you've happened to flip past a music video channel on your way to ESPN or the Cartoon Network you understand what I mean. These guy's are just as real as anyone I've met.

So, until next time... YEeeAAAAHHHHH!

Check back to hear more about my work with Payne and his producers, as well as a final word on my analog summing discussion.

Thanks!

Andrew Wayland

Analog Summing? Necessary or Not? Part Four, The Answer To The Question

2006-04-02T03:29:00.000-07:00

Continued from Part Three,

If we escape the all this mess, the differences between analog and digital audio, the subjective opinions, and get down to why we're um "hear,"- the summing aspect, we can arrive at a simple answer.

The answer to our question is this:
Analog summing is only necessary if your client desires it, or if you personally prefer it.

What I have heard from my colleagues is that the components of analog devices add a sonic elements that are subjectively pleasing. Those added elements give the analog summing mixers and consoles their appeal.

Digital summing does not add those elements, but who is to say (objectively) that an analog summed mix sounds better than a digitally summed mix?

One thing is for sure- psychoacoustic perceptions can be skewed.

In my conclusion I will lead you to the men who have settled many of the insecurities I had about digital summing, and whose work has greatly refined my view of whole analog summing discussion/argument.

Again, thanks for tuning in!

Andrew Wayland

Updated Posts! Analog Summing? Necessary or Not?

2006-04-02T01:42:00.000-08:00

Hi All,

I've been swamped with several projects lately, and I finally have gotten back to our discussion of the necessity of analog summing. In the last month I've actually changed some of my own personal feelings about the topic.

On a side note, SPRING IS HERE!

This taken into account, I have recently edited all previous posts on the subject. Here they are in bulk:

Part One

DAWs
Digital Audio Workstations are now commonplace in the professional recording studio. Large and small studios that ran miles and miles of tape have, for the most part, surrendered to the trend and secured their own high quality DAW rigs. DAWs are small, less expensive to maintain than large format analog gear, and moderately compatible with existing analog systems. Further, the DAWs utility, automation and ease of use has caused it to become so popular that the technology is already an industry standard.

Topic
This series of posts is meant to examine and discuss the necessity and purpose, if any, of summing audio in the analog state, e.g., through a console or line mixer, rather than within a DAW. I do most of my work using Pro Tools, a common DAW, so the topic hits close to...work.

Industry professionals generally say that analog summing just "sounds better." They argue that summing "In the box," or inside DAWs, takes away clarity, depth, punch, definition, and/or overall quality from a mix. Suppose that analog summing actual adds elements to a mix during the summation process, would this account for the lack of these elements in digitally summed mixes? As far as DAW summing quality goes, many more myths are believed than actual truth, and I will discuss some actual research by fellow engineers that disproves some of these myths later on.

Back on track, one of the problems with using analog summing methods is that many people cannot afford "Class A," analog consoles.

It is also quite possible that the average person does not hear a recognizable difference between digitally summed and analog summed audio, and therefore decides that analog summing is not "necessary."

Gear manufacturers have responded to market demand recently by releasing small rack mountable analog summing units labeled as "summing mixers." These units offer the analog circuitry that professionals adore at a price the modest studio owner can afford. (Users simply make a mix with their DAW and then run the mix through the summing mixer.)

Unfortunately, De Gustibus Non Disputandum Est, so anything I say concerning my own opinion or the opinion of others in this matter is not useful. I must then focus on why analog summing may or may not superior based on an objective standpoint. Ultimately, apart from objective fact, one must make his or her own decision.

Part Two

Continued from Part One...

So, does it really matter? Can a person actually hear more clarity, etc... with either summing method? If so, why?

Before I attempt to answer these questions-let's get some common ground.

Let us agree that any audio signal began as some form of energy, and that to hear sound our sensory organs had to translate fluctuations in air pressure. What our ears hear is an infinitely variable analogous signal, which (strangely and amazingly) is transformed into bioelectrical impulses and then sent to the brain to be processed and perceived as sound.

For now, let us escape a long discussion of the theory and concept of analog and digital audio, and agree that analog signals are completely and continuously variable electrical signals and that a digital signal is an imperfect picture of an analog signal, encoded into a strand of 1's and 0's (using pulse code modulation). [This is a very short and narrow view of the anatomy of analog and digital signals.]

Summing, In the Analog and Digital Realm

When I talk about summing audio, I am also talking about mixing audio, but not in the sense of mixing a song. The term summing is appropriate for this reason. Summing basically means adding or combining things together, (e.g. combining 32 tracks/channels into 2 tracks/channels).

The "summing" or adding of analog signals seems simple enough, one voltage is added to another voltage and Wal-la! A summed analog signal. In the digital realm summing requires math: the addition of those 1's and 0's, adding numbers together to make bigger numbers.

Depending on variables within the analog chain, the original analog signal exits a summation process in the fashion that it entered, that is, it comes out relatively the same as it went in, only now it is combined with other analog signals. Technically, analog summing takes place through a series of resistors. Please understand that the technical side of analog summing is much more complex, and in real life, analog circuits will actually add and/or delete elements to a signal in trace amounts as it travels through the signal path.

Summing in the digital realm is actually a series of mathematical computations.

Digital audio is hindered by limitation. For example, a mix bus with 24 bits of resolution, or a 24 bit word length, reveals a maximum reachable sum of 2²⁴, or 52,980,000,000 discrete levels or a theoretical dynamic range of 144 dB. (As a side note: the actual dynamic range of component utilizing a 24 bit word is going to be less than 144 dB! My Digi 002 R is spec'd out to reach up to only 108 dB, Pro Tools HD 192 I/0 reaches to 120 dB. The higher the component's dynamic range, the more resolution it has, the better it is.)

Because of this digital audio sums are limited to how much resolution a system uses. Further, what happens if the mathematical equation is wrong in some way? As we know, math is unforgiving, and should one variable be wrong, the whole computation will be wrong. Coming up in our discussion is some objective data about current DAW programs and their ability and inability to produce solid math, and the results may surprise you!

Part Three

Continued from Part Two...

More on DAWs and Digital Audio

Digital audio signals are not perfect pictures of the analog signals they represent. Un-encoded analog signals that are summed will produce a equally relative imperfect picture of the original signals also! That is to say that in a way, digitized audio leaves out elements through lossy coding, and analog audio adds or deletes elements to itself during circuit travel.

What's wrong with digitized audio?

DAWs are digital audio workstations. They must process digital signals. All signals must be converted from their original analogous state in to a codified digital state, via analog to digital conversion or "A/D."

One thing we do know is that A/D conversion does in fact leave out information. This left-out audio information is called Quantization Error. Once lost (or left out, in this case) it can never be regained.

Strictly analog audio never goes through this process, so naturally, it wouldn't lose the information if it would if it went through A/D. Even then, arguing about analog audio being relatively less imperfect than digital audio would be pointless.

Waking up we realize that hardly anything is completely analog anymore. To get to a CD, it must go through A/D.

Once it is captured and converted, digital audio can theoretically stay constant in a medium, be consistently and accurately reproduced, and all for a much much greater period of time than analog tape ever could. So, an argument for people in the digital camp is that digitized audio, with a high enough sample rate and bit resolution, though imperfect, can theoretically stay in its 'original' state, without being changed, for a longer than the life span of a single human being.

Analog audio processing is analogous and always changes. Tape deteriorates and capacitors go bad. But I digress...

Continued in: Part Four

Thanks for your time, as always, feel free to post your questions, comments, and corrections.

Andrew (Wayland) @ AfaraWayland

New Addition! AT 4060 Tube Mic

2006-04-01T10:07:00.000-08:00

Hi all,

I recently procured a new microphone for my collection.
Its an Audio Technica 4060 Tube Mic.

Notice the nice case! Your local Lowes has tool cases just like these that you can customize to fit your needs. The brand is Task Force, though, you will not find the brand easily on the web. Make sure to to get one with the foam inserts in the case interior. I have seen cases that do not have the foam.

All I can say is...sweet!

Here is a synopsis from AT's site:

© 2005 Audio-Technica U.S., Inc.

"Vintage tube sound with the versatile performance necessary for the most demanding studio applications"

"Wide dynamic range, low self-noise and high max SPL capability"

"The 2-micron-thick, vapor-deposited gold diaphragms undergo a five-step aging process so that the optimum characteristics achieved remain constant over years of use"

"Dual-diaphragm capsule design maintains precise polar pattern definition across the full frequency range of the microphone"

"Precision-machined, nickel-plated brass, acoustic element baffle provides enhanced element stability and optimal sensitivity"

"Regulated and heavily filtered heater current prevents noise contamination of audio signal chain"

"Hand-selected tubes are individually tested and aged to maintain peak performance"

"Large coupling transformer provides superior low-frequency linearity"

"Custom shock mount provides superior isolation"

Thanks for tuning in!

Andrew Wayland

Live Recording: Ahead of the Wake 3/11/06

2006-03-20T18:00:00.000-08:00

© Andrew Hull 2006, CD design
Ahead of the Wake-Live at the Brick 3/11/06.

Hi All,

The same weekend I finished recording the two choirs, I got a last minute call to record a the band, Ahead of the Wake. Ahead of the Wake is one of the finest musical acts Decatur has to offer, so naturally I jumped at the chance to partake in their evening performance.

The band performed a wide array of popular crowd-pleasers for the patrons of the the Brick, and kept everyone dancing, even after the lights went up and the barkeeps started to rustle everyone out.

Here are some shots of the mic setup I used:

Notice the interesting manner which they are fastened to the ceiling. A special thanks goes out to fellow engineer, Chad Henderson, of Decatur, for providing his custom mic hanging devices. He made these custom mic hangers specifically for this venue, but one can see how such a design could be useful in other applications involving low ceilings and "acoustic" ceiling tiles.

The mics shown are two AKG C 414 B-XLS, set to OMNI, filtered at 80 Hz, and padded -6dB. The zero axis of each mic is angled toward each PA speaker. I had to adjust the vertical alignment of each mic to account for the fact that one PA speaker was on a stand, and the opposite was on a table.

Note also the obvious concerns one should have when setting up expensive equipment in a public setting, especially one involving the sale of alcohol. Despite the advantageous position of the mics, the crowd factor still made me uneasy. I looked back a few times to find that one of the mics had been moved, and I had to reposition the mic. If I had been using a more directional polar pattern, the effect of off-axis coloration would probably be very apparent. Omni patterns allow for less off-axis coloration, so a slight movement away from the mic-to-source zero axis will not greatly affect the captured sound.

The two cables were run along the ceiling tiles over to the sound mixer station where I was seated. I cannot express how much of a benefit the ceiling mount system provided as opposed to a floor stand system. When the band broke into song, the dance floor filled with slightly to moderately intoxicated patrons, so a mic stand in the middle of the fray would be a very cumbersome and dangerous liability.

The band hired Wayne Thrasher, of Decatur, to provide and run the PA equipment. Let me take this opportunity to suggest that anyone trying to coordinate a live recording, should make sure that the sound guys are all on the same page. This means making sure the live-sound engineer and the recording engineer know what each other is doing, and how they are doing it. Wayne wasn't informed about the performance recording until he arrived, after which I had already finished setting up, so naturally there followed confusion.

Fortunately Wayne is an experienced sound engineer as well, so he was able to adapt to the situation.

Here are shots of my station:

Above, I am seated at my station, tending the recording process.

After testing the mics during the sound check, I noticed that the the vocals were not clearly translating through the microphones. I solved this problem by taking the a balanced vocal send out of Wayne's mixer into my ISA 428 line inputs. Since no instruments were being sent through the PA (only vocals), this allowed a direct vocal mix that could supplement the vocals being picked up by the microphones.

However, a problem one encounters when trying to mix such a combination is that the microphone (with PA) mix will fight with the direct PA vocal mix for occupation of the frequency spectrum. What happens is that the direct vocal mix arrives at the preamps faster than the vocal mix coming out of the PA speakers that is captured by the microphones, and the result is a phase (time domain) difference. When the two signals are combined, the phase difference creates what engineers call a "comb filter." Comb filters are undesirable and make the combined signals sound hollow.

To counter this effect, one may choose to delay the direct signals to make up for the phase difference. To do this one would need to measure the electronic delay somehow. I chose to bring the vocal mix out in front, so as to mask the vocal mix coming from the PA speakers and the effects of the comb filter.

All in all, I recorded over 30 songs that night. During the next few days, I mixed and rendered the songs to stereo files, chose 12 of the tracks to be included on a finalized master, designed a disc face, which is shown at the top, and burned a number if CDRs for the band to market and sell. (The CDs can be purchased through the band for $10 at the link I provided at the top.)

My thanks goes out to Chad, Wayne, and especially Ahead of the Wake.

Thank you for your time as well! Stay tuned for a long awaited conclusion to our discussion of Analog vs. Digital summing.

Andrew Wayland

Recording A Church Choir: What I Learned

2006-03-17T21:30:00.000-08:00

A difference you can see from the layout at the top is that one of the choirmasters is stationed out front.

One thing I would have done differently was put a few more spot mics up.

I suggested that the choir move the piano from the far right to the front center. However, this was not agreeable with everyone, so the piano stayed where it was, lid open and facing the back wall. This left the captured sound clear in the choir but muddy and distant in the piano.

I just had to accept the disagreement and move on. What I should have done at that point was stop the session and place a stereo pair or additional right flanking mics on the piano. Of course, this is not optimal, since it does introduce phase relation issues when mixing, it destroys the spatial balance and perspective, and it kills tracking time like a power outage.

Another thing I would do differently is try to move the mics closer to the choir.

The way things worked was that I had about twenty minutes to position the mics, set levels, test and start recording. So basically the recording happened on the spot with no adjustments.

In hindsight, I should have stopped the session and made adjustments as necessary. Unfortunately, stopping the session is usually not a pleasing option for the client. In most cases clients do not have a clear understanding of what is necessary for sound engineers, and therefore, will not understand what the "hold-up" is. Some clients will be more understanding than others.

The captured sound had plenty of room ambiance, but left me wanting to hear more of the choir.

In the end, I enjoyed recording this choir very much, and I look forward to doing more choral recordings.

Thanks for viewing,

Andrew (Wayland) @ AfaraWayland

Why Do I Have Two names?

2006-03-16T15:57:00.000-08:00

Well, the answer to this question is almost confusing, so I am attempting to explain.

My last semester at MTSU, I had an internship position with one of the finest mastering houses in Nashville, Georgetown Masters.

The chief mastering engineer their had the same first name as I, so to eliminate any confusion, I started going by my middle name "Wayland." Wayland is a pretty nifty name in these days, at least, its more "original" than plain Andrew.

So, since my internship made the name normal for me, I debated whether or not to use middle name as my studio name, or credit name. Unfortunately, my previous credits had all been "Andrew." So, upon graduating college and getting out on my own, I thought up the name AfaraWayland, and here we are.

So, I respond to both names now!

Thanks for tuning in!

Andrew "Wayland" @ AfaraWayland

Recording A Church Choir, Step Four: Evaluate, Mix, and Print

2006-03-16T15:18:00.000-08:00

When the tracking was finished, I packed up, headed home, and set out to mix the captured performance.

Above is a CD cover that I designed for the group. It shows what print shops call "bleeds." In actuality, I simply print this image direct to a CDR face. Discmakers sold me a printer that does just that. Cool!

Designing your own CD cover is simple with a good design program. However, beware that some software programs are not compatible with certain direct to CDR printers.

For large duplication runs, it is advisable that one contract a professional duplication house. Burning and printing CDRs takes a fair amount of time.

After the mixes were finished, I rendered the mixes to CDR, and printed the CDR face-allowing a 24 hour period for the ink to dry.

In Conclusion

A professional product will always look better with a great design and color ink. For those of you who have great mixes already, I can design CD covers for your project at a very competitive rate.

Stay tuned for more CD design examples and a continuation of current topics!

Andrew (Wayland) @ AfaraWayland

Recording A Church Choir, Step Three: The Fun Part

2006-03-15T11:39:00.000-08:00

After I finished all the research I could do on the site and the choir itself, I made a formal agreement with the choirmasters and laid out the project from start to finish.

Now comes the fun part.

I made a list of things I needed to have for the gig. Then I rounded up the gear and traveled to Alabama. Insert barefoot joke here.

On the day of the "taping" I arrived at the church two and a half hours prior to show time. (I tried to document as much as possible.) Here are shots of the floor plan again:

I found the best spot for me was in the back left, so I proceed to set up my own little corner.

I had to wait until after and evening service to set the mic placements, but the format I used was similar to my original guesses.

I used three microphones in this setup. Three, along the same parallel, a center figure-eight and two flanking omni mics. I ran the mics into my ISA 428 pre with A/D card installed, and then ran via lightpipe to a 002R unit.

The whole process took about five hours, with one and one half of those hours being actual tracking.

Thanks for viewing! Stay tuned for the conclusion of how one can record a church choir.

Andrew (Wayland) @ AfaraWayland

Recording A Church Choir, Step Two: Reach an Agreement

2006-03-12T18:54:00.000-08:00

Above is an example of a recent recording agreement. It was four pages in length.

I cannot stress enough how important it is for all parties involved in an honest business transaction to fully understand the agreement.

A simple and effective agreement will make sure everyone is "on the same page."

The more complex the arrangement, the more necessary it may be to have an attorney review or even author the agreement. This is why labels employ droves of entertainment lawyers.

If you are planning to do something simple like on-site sound or live recording, a simple agreement may be all you need.

Here are some suggestions as to what should be in a recording agreement.

Issues that you definitely want to discuss:

Include but are not limited to:
Who is involved?
What is each party doing for each other?
What are the benefits of doing business?
How will the business be done?
When will business be done?

Issues that I suggest you discuss:

Include but are not limited to:
What are the rights of each party?
When are those rights satisfied?
What recourse does a party have if the opposite party does not fulfill their agreement?
How long do the rights last?

This is by no means everything, but you get the point: be as thorough as possible!

Thanks for tuning in, stay tuned for more posts!

-Andrew (Wayland)

How To Contact Me

2006-03-12T18:02:00.000-08:00

Hi ALL,

Many of you may wonder how to contact me for AfaraWayland's production services...

You can contact me via Email here: e-mail Wayland

Send your name and a short description of what sort of service you require.

Include your phone number if you prefer to be contacted by phone.

Thanks as always,

Andrew (Wayland)