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Build Your Own Guitar Special Effects
Analog Audio Delay Systems
by Spike Tsasmali, Lupine Systems

Introduction

Audio Delays come in a wide variety and can produce an array of effects depending on what type of delay you are using. Here is a breakdown of some of the more popular types of delays built over the years:

Mechanical Delay

Mechanical delay is done by recreating the sound with a small transducer at one end of a solid mechanical media, such as a spring or steel band. The transducer physically "vibrates" the media and the sound travels mechanically along the length of the media from one end to the other. The longer the media, the longer the delay. Once the sound has traveled the length of the media it is then picked back up using a small pickup device, usually a small coil similar to a guitar pick-up. The sound is then amplified and then re-mixed with the non-delayed original sound to create reverb.

Mechanical delays suffer from several inherent problems. First, the mechanical media is subject to outside vibrations from speakers, stage movement and sounds from adjacent instruments. They can also be "banged", which creates a nasty noise (have you ever bumped a reverb amp and heard the "spring" sound?)! Also, depending on many factors, mechanical delays can alter the bandwidth of the sound, filtering off high frequency and leaning towards bass. Although troubled with many factors, mechanical delays are very reliable, dependable, inexpensive, easy to implement and time tested. Mechanical delay is the world's most popular form of delay.

Tape Delay

Tape delay systems utilize an endless loop of tape, usually in a cartridge, much like an 8-track tape cartridge to perform the delay function. These delays are actually no more than a complicated tape recorder. Audio is recorded onto the tape using a magnetic head placed to one side of the cartridge. Once the tape is recorded the tape then passes through a distance then is dragged across a playback head. The audio is then amplified and re-mixed with the original non-delayed signal to form reverb, chorus, echo, sequencing and sampling functions. Varying the distance between the heads and the speed of the tape increases and decreases the delay time and a handy mixer allows the user to either mix the delayed signal with the original or use the delayed signal all by itself.

Tape delays come in many sophisticated forms, but all of them work exactly the same. The only features offered between the models are selective features, multi-track and very long sampling periods. Although very reliable, tape delays suffer from the occasional "broken tape", and some lesser expensive models suffer from tape drag. Also, the tape wears out quickly and the heads filth easily so maintenance is required on a regular basis to maintain decent sound quality.

Bucket-Brigade Delay

Bucket-Brigade delays were the first form of analog audio delay systems to be fully solid-state with no moving parts. Bucket-Brigade Devices (BBD's) are silicon microelectronic integrated circuits that contain hundreds or thousands of capacitive "bucket-brigade" stages. These chips work by taking in the audio on one end, charging a capacitor to "sample and hold" the audio, then a clock pulse shifts this charge from capacitor to capacitor along the chain of BBD stages while charging a fresh sample with each shift. When the audio shifts from one end of the delay to the other the shifting and clocking functions delay the audio. The audio is then recovered in its original analog form at the opposite end of the BBD chain and is then amplified and mixed with the original undelayed signal to create reverb, chorus, and time-frame modulation (sounds like a "robot" talking).

Bucket-Brigade delay has a slew of problems that must be overcome before it can be used as a decent form of audio delay. The most important of these issues is "clock noise", an unwanted artifact of the clocking function used to shift the audio through the BBD stages. Cleaning up the clock noise is usually done by using a zero-balance mixer and a dual-phase clock. The clock, being dual-phase will cancel itself out if balanced. Other techniques incorporate filtering, but filtering may affect the overall bandwidth. Other issues include power consumption (these devices are half-digital, half-analog so they are either CMOS at 12v or TTL logic at 5v, whereas most guitar pedals are 9v battery powered), sensitivity, PC board layout specific and limited delay times. Another issue is cost, for these chips are quite expensive.

Bucket-Brigade delay does offer features not capable from other delay forms. One of these features is time-frame modulation. Time-frame modulation takes into consideration the fact that the audio coming into the BBD will be "sliced-up" by the sample clock. If the sample clock rate is in the same ballpark as the incoming audio's frequency, the two will "collide" and harmonize. The result is defined in Hetrodyne's Theory where the result frequency from mixing two frequencies is A-B. The cancelling effect creates a sound that has been used to create the voice for science-fiction robots, machines and music. A very nice effect indeed!

Digital Delay

Digital delay is the most complex of all forms of delay. Although it appears to be quite simple, it takes a lot of electronics to pull it off with any reasonable sound quality. Until recently, all digital delay systems used discrete TTL (Transistor-Transistor Logic) chips to do the job. Nowadays, with modern microprocessor technology these delays have been drastically simplified using high-speed processors. But back in the late 1970's processors did not run fast enough to actually do the task, so I won't get into using processors for digital delay.

Digital delay works by first sampling the incoming audio using an Analog to Digital (A/D) Converter. The sampled audio is converted to a digital (numeric) value and then saved in digital memory. Delay is created by playing back the sampled audio starting at a memory address less than the current address incoming data is being saved at. The more this difference is the longer the delay. Audio is recovered by reading data from memory and then writing it into a Digital to Analog (D/A) Converter. Once converted back to analog, the audio is then amplified then mixed with the original non-delayed signal to create reverb and other effects.

Digital delays offer the widest range of effects but come at a huge cost of complicated circuitry and price. Digital delays have several drawbacks as well. The greatest benefit from digital delay is the availability of very long delay periods, extreme accurate sampling, repeat, sample and sequencing functions, all at the touch of a button. Digital delays can actually be a musician's partner, playing background tracks while the musician takes over lead roles. It can harmonize, making one guitar sound like many. So versitile, digital delays have become a staple part of many guitarist's pedal collection.

The main drawback of digital delay is cost. Some of these delays cost as much as $8000! It all depends on how many features the delay offers and how much "memory" it has. Memory is always one of the priciest parts, but it is not the only part that can be expensive. Digital delays, being digital, can come in 4-bit, 8-bit, 16-bit and 32-bit. The number of bits can make a difference. 4-bits is virtually useless unless you are doing voice only (no music). 8-bits is quite standard and works very well. 16-bits does not offer anything 8-bits can't offer and 32-bits is only a selling point to make you think it is so much better. So stick with 8-bits. That is all you really need.

The type of A/D and D/A converters used also matters. There are many types. Early models used a technique called "Successive Approximation", where a slew of electronics "guessed" the digital equivalent until it actually "matched" the incoming signal -- a "trial and error" way of doing it. Later, more modern techniques use "Sample and Hold", "Delta-Sigma" (change in current phase) and "Flash". For music the best I've found is Sample and Hold conversion. This technique "grabs" the analog input, then samples the sample. A nice stable conversion. Other techniques split the frequency and amplitude components of the incoming signal apart and sample them separately using different sampling methods for each, but this is a very costly method only found in extreme studio-quality top-end equipment costing thousands of dollars.

Sampling rate is the final major factor in digital delay. This is complex, so bear with me...

When audio signals are converted to digital, the audio is "sliced" into "samples" and then saved in memory, much like the memory used on your computer. The number of samples taken per second determines how much "resolution" the sound will have when converted back to analog, kinda like the resolution of a digital camera picture -- the more pixels, the sharper the picture. In audio, the highest frequency commonly reached is about 16KHz. But you can't sample AT 16KHz! There is a NASTY rule of physics in the way! It's called "Algol's Theory". And Algol was a hum-dinger!

Algol realized that if you chop up an analog signal into pieces, the chopping of that signal will mix with the signal itself and therefore, you have TWO signals -- the analog signal being sampled and the sample clock (and/or the artifact of the slicing effect). When these two signals mix, like any other set of signals, the Hetrodyne Theory applies. Hetrodyne states that when there are two frequencies, let's say A and B, and you mix them, you will get:

  • A
  • B
  • A+B and
  • A-B.
    So if you had a 16KHz signal and you were sampling at 16KHz, you might just end up with ZERO!

    You also might end up with a BUNCH of unwanted signals. These unwanted signals are called "aliases" and represent harmonics created by the mixing of the sampling and the original signal.

    To avoid aliases, there are two things that can be done. First, use filters. Stop any signal from reaching the A/D converter that is over the sample rate. Or you could simply sample at twice the highest incoming frequency plus ONE. Or you could do both just to be sure.

    So to avoid aliases and keep the audio clean and give the best bandwidth, 32KHz is chosen as a classic sample rate. This is twice the typical incoming frequency. Filters are usually placed on the output of the signal input buffer amp to stop any frequencies over 16KHz from passing through to the A/D converter as well.

    So now you see where digital delays can get complicated. I've just begun to explain it! So I will save that explination for the next installment of these articles.

    Building Your Own Delay

    In this series of articles I will explain how to build two different audio delay systems -- a simple, but quite effective Bucket-Brigade delay and a much more complex, full featured Digital Delay which will rival delays costing thousands of dollars! The Bucket-Brigade project is easy and fun to build and won't set your pocketbook back as much as you might think, and if you build-it-yourself, you'll learn how it works, how to use it and how to modify it to fit your personal sound effect needs. The Digital Delay project is very complicated and is recommended for advanced experimenters and professionals only as it uses three large PC boards and over 60 chips!

    Lupine Systems will offer pre-etched, pre-punched PC boards for all of these projects for those who cannot make their own PC boards. In this series of articles, all PC board layouts will be presented at the Lupine Systems Download Site in both EasyTrax format and as Gerber files. Schematics will also be posted in the usual EasyTrax format and as large JPG files. I recommend that you print out these schematics and use them as reference throughout the presentation.

    Building a Bucket-Brigade Audio Delay    Difficulty=2 Howls

    One of the easiest delay systems to build is a Bucket-Brigade delay. A simple but quality Bucket-Brigade delay is pictured to the left. This delay offers variable delay time, depth and decay and will allow you to choose whether you wish to mix the original incoming signal with the delayed signal or use just the delayed signal stand-alone. You will be able to do reverb, echo, chorus and time-frame modulation using this board. All parts are redily available except for the BBD chip, which can still be purchased but not from the classic distributor sources. I will provide more information on where to get this chip later on in the article.

    ...

    How It Works

    Refer to the Schematic Diagram (below, click image to enlarge). Audio is supplied to the board through one half of the dual RCA input/output connector CN1. Audio is then split into two feeds, one to the input buffer amp and the other to the output mixer. The input buffer amp consists of one half of a LF353 Dual BiFET op amp. This type of op-amp was chosen for its high slew rate, excellent gain and overall audio performance. Gain on the buffer amp is set to 1 and is maintained through a balance between the overdrive current of the output of the BBD chip and supply voltage. This threshold is set via potentiometer R28. The output of the buffer amp is then fed to the input of the first of two stages of the SAD1024 BBD chip.


    Bucket-Brigade Delay Schematic Diagram (click to enlarge)

    The SAD1024 is a dual 512 stage Bucket-Brigade device. To get longer delays, both stages are cascaded together. The output of the first stage is fed into the input of the second stage and clocked through both stages at the same rate. To nullify clock noise from the first stage the audio is first sent through a zero-balance mixer consisting of potentiometer R7 and resistors R12 and R13 before being sent into the second stage. Final delayed audio is recovered from the BBD at pins 11 and 12 and once again are fed to a zero-balance mixer to remove unwanted clock noise. The audio is then boosted back up to line levels by the remaining half of U4 and presented to the output mixers along with the remaining split feed from the original input splitter for depth, decay and mix functions. After mixing the audio appears at the other half of connector CN1 for output back into the world.

    Clock is provided by U3, a 555 timer running as an astable multivibrator. The clock is split into dual phase for each section of the SAD1024 by U2, a 4013 CMOS dual flip-flop. Clock pulses are then sent to the SAD1024 to shift the analog signal through the BBD.

    Smooth power is provided to the entire board via 7812 regulator U5. Power input comes in via power connector CN2 and is either rectified if AC or polarity corrected if DC by bridge rectifier Z1. Capacitor C3 provides ripple filtering before the power is sent to U5 for final regulation. To keep out digital noise from the clock divider from reaching the buffer amp stages, power is fed through choke coil L1.

    Construction

    Construction can be easily facilitated by using a Printed Circuit, but you can use point-to-point pad-per-hole board as well. If you choose to use pad-per-hole board, be sure to observe good grounding techniques and keep the digital clock section powered and grounded separately from the analog sections. A full PC board layout is available in EasyTrax format as well as in Gerber plot format from the Lupine Systems Download Site. The download contains the PC board layout, bill of materials, parts placement diagram, hole drill guide, Gerber plot and EasyTrax file. You can view and print using a laser or inkjet printer Gerber files by using GC Prevue.


    Bucket-Brigade Delay PC Board Parts Placement Diagram

    Parts List
    Bucket-Brigade Delay

    DesignatorQuantityDescription
    C1,C2,C5310uF 16v Radial Electrolytic Capacitor
    C312200uF 25v Radial Electrolytic Capacitor
    C4,C924.7uF 16v Radial Electrolytic Capacitor
    C61100pF Ceramic Disc Capacitor
    C7,C8,C13,C144.1uF Axial Conformal Capacitor
    C10,C12,C173.001uF Axial Conformal Capacitor
    C1111uF 16v Radial Electrolytic Capacitor
    C15,C162220pF Ceramic Disc Capacitor
    C18147pF Ceramic Disc Capacitor
    CN115mm Power Jack
    CN21Dual RCA Jack
    D11Red LED, Size T1
    D2,D321N4001 Diode
    J1,J2,J3,J44Zero Ohm Jumper (use bare wire)
    L11100uH Conformal Coil
    R11240 ohm, 1/8 watt Metal Film Resistor
    R2,R8227K, 1/8 watt Metal Film Resistor
    R3,R11,R14,R18,R205100K, 1/8 watt Metal Film Resistor
    R4,R15,R25,R29422K, 1/8 watt Metal Film Resistor
    R17,R19,R32310K, 1/8 watt Metal Film Resistor
    R911.8K, 1/8 watt Metal Film Resistor
    R12,R13,R16,R22,R23568K, 1/8 watt Metal Film Resistor
    R21,R26,R313220K, 1/8 watt Metal Film Resistor
    R24139K, 1/8 watt Metal Film Resistor
    R271150K, 1/8 watt Metal Film Resistor
    R7,R10,R28310K Subminiature Trimmer Potentiometer
    R51100K Linear Taper Potentiometer
    R6110K Linear Taper Potentiometer
    R301500K Linear Taper Potentiometer
    S1,S22Miniature Toggle Switch, SPST or SPDT
    U11SAD1024 Bucket-Brigade Delay Chip
    U21CD4013BCN Dual CMOS Flip-Flop
    U31NE555V, 555 Timer
    U41LF353N Dual BiFET Op-Amp
    U517812 12-volt Positive Voltage Regulator IC
    Z111A, 200PIV Bridge Rectifier
    --1Heatsink for U5

    All components are available from Mouser Electronics, Digi-Key or Jameco Electronics, except for the SAD1024. The best place to find this chip is Ebay. There are several sellers who operate Ebay Stores that carry this chip regularly.

    Setting up The Delay

    The delay must be calibrated before it can be used. Calibration is easy. Simply hook up the delay to your stereo using two RCA patch cords. Hook it up to the TAPE monitor connections. Apply power to the 12v power input jack and turn on the delay. Adjust your stereo for normal volume, start some music. Adjust all three control pots (DELAY, DEPTH, DECAY) to maximum rotation and put the MIX switch in the NO MIX position. There should be NO MUSIC come through the stereo. Slowly adjust R28 until you can hear the music come through. Fine adjust this control for the best volume.

    Next, adjust R10 and R7 to eliminate clock noise. Disconnect the INPUT from the delay unit before you begin this adjustment. You may have to adjust these one at at time, or both interchangeably. Start off with both controls centered then tweak off center from there while listening. Once you tweak these two pots for the best noise reduction, connect the INPUT back up, then flip the MIX switch. Play with the DEPTH, DECAY and DELAY controls to get echo, reverb and time-frame modulation effects!

    Also on the Lupine Systems Website...

    Build a Digital Delay

    Back to the Lupine Systems Homepage
    Lupine Systems Download Section
    How to Make PC Boards
    How to Use EasyTrax CAD Software

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