Amplifier Classes from A to H

Engineers and audiophiles have one thing in common when it comes to amplifiers. They want a design that provides a strong balance between performance, efficiency, and cost.

If you are an engineer interested in choosing or designing the amplifier best suited to your needs, you’ll find columnist Robert Lacoste’s article in Circuit Cellar’s December issue helpful. His article provides a comprehensive look at the characteristics, strengths, and weaknesses of different amplifier classes so you can select the best one for your application.

The article, logically enough, proceeds from Class A through Class H (but only touches on the more nebulous Class T, which appears to be a developer’s custom-made creation).

“Theory is easy, but difficulties arise when you actually want to design a real-world amplifier,” Lacoste says. “What are your particular choices for its final amplifying stage?”

The following article excerpts, in part, answer  that question. (For fuller guidance, download Circuit Cellar’s December issue.)

The first and simplest solution would be to use a single transistor in linear mode (see Figure 1)… Basically the transistor must be biased to have a collector voltage close to VCC /2 when no signal is applied on the input. This enables the output signal to swing

Figure 1—A Class-A amplifier can be built around a simple transistor. The transistor must be biased in so it stays in the linear operating region (i.e., the transistor is always conducting).

Figure 1—A Class-A amplifier can be built around a simple transistor. The transistor must be biased in so it stays in the linear operating region (i.e., the transistor is always conducting).

either above or below this quiescent voltage depending on the input voltage polarity….

This solution’s advantages are numerous: simplicity, no need for a bipolar power supply, and excellent linearity as long as the output voltage doesn’t come too close to the power rails. This solution is considered as the perfect reference for audio applications. But there is a serious downside.

Because a continuous current flows through its collector, even without an input signal’s presence, this implies poor efficiency. In fact, a basic Class-A amplifier’s efficiency is barely more than 30%…

How can you improve an amplifier’s efficiency? You want to avoid a continuous current flowing in the output transistors as much as possible.

Class-B amplifiers use a pair of complementary transistors in a push-pull configuration (see Figure 2). The transistors are biased in such a way that one of the transistors conducts when the input signal is positive and the other conducts when it is negative. Both transistors never conduct at the same time, so there are very few losses. The current always goes to the load…

A Class-B amplifier has more improved efficiency compared to a Class-A amplifier. This is great, but there is a downside, right? The answer is unfortunately yes.
The downside is called crossover distortion…

Figure 2—Class-B amplifiers are usually built around a pair of complementary transistors (at left). Each transistor  conducts 50% of the time. This minimizes power losses, but at the expense of the crossover distortion at each zero crossing (at right).

Figure 2—Class-B amplifiers are usually built around a pair of complementary transistors (at left). Each transistor conducts 50% of the time. This minimizes power losses, but at the expense of the crossover distortion at each zero crossing.

As its name indicates, Class-AB amplifiers are midway between Class A and Class B. Have a look at the Class-B schematic shown in Figure 2. If you slightly change the transistor’s biasing, it will enable a small current to continuously flow through the transistors when no input is present. This current is not as high as what’s needed for a Class-A amplifier. However, this current would ensure that there will be a small overall current, around zero crossing.

Only one transistor conducts when the input signal has a high enough voltage (positive or negative), but both will conduct around 0 V. Therefore, a Class-AB amplifier’s efficiency is better than a Class-A amplifier but worse than a Class-B amplifier. Moreover, a Class-AB amplifier’s linearity is better than a Class-B amplifier but not as good as a Class-A amplifier.

These characteristics make Class-AB amplifiers a good choice for most low-cost designs…

There isn’t any Class-C audio amplifier Why? This is because a Class-C amplifier is highly nonlinear. How can it be of any use?

An RF signal is composed of a high-frequency carrier with some modulation. The resulting signal is often quite narrow in terms of frequency range. Moreover, a large class of RF modulations doesn’t modify the carrier signal’s amplitude.

For example, with a frequency or a phase modulation, the carrier peak-to-peak voltage is always stable. In such a case, it is possible to use a nonlinear amplifier and a simple band-pass filter to recover the signal!

A Class-C amplifier can have good efficiency as there are no lossy resistors anywhere. It goes up to 60% or even 70%, which is good for high-frequency designs. Moreover, only one transistor is required, which is a key cost reduction when using expensive RF transistors. So there is a high probability that your garage door remote control is equipped with a Class-C RF amplifier.

Class D is currently the best solution for any low-cost, high-power, low-frequency amplifier—particularly for audio applications. Figure 5 shows its simple concept.
First, a PWM encoder is used to convert the input signal from analog to a one-bit digital format. This could be easily accomplished with a sawtooth generator and a voltage comparator as shown in Figure 3.

This section’s output is a digital signal with a duty cycle proportional to the input’s voltage. If the input signal comes from a digital source (e.g., a CD player, a digital radio, a computer audio board, etc.) then there is no need to use an analog signal anywhere. In that case, the PWM signal can be directly generated in the digital domain, avoiding any quality loss….

As you may have guessed, Class-D amplifiers aren’t free from difficulties. First, as for any sampling architecture, the PWM frequency must be significantly higher than the input signal’s highest frequency to avoid aliasing….The second concern with Class-D amplifiers is related to electromagnetic compatibility (EMC)…

Figure 3—A Class-D amplifier is a type of digital amplifier (at left). The comparator’s output is a PWM signal, which is amplified by a pair of low-loss digital switches. All the magic happens in the output filter (at right).

Figure 3—A Class-D amplifier is a type of digital amplifier. The comparator’s output is a PWM signal, which is amplified by a pair of low-loss digital switches. All the magic happens in the output filter.

Remember that Class C is devoted to RF amplifiers, using a transistor conducting only during a part of the signal period and a filter. Class E is an improvement to this scheme, enabling even greater efficiencies up to 80% to 90%. How?
Remember that with a Class-C amplifier, the losses only occur in the output transistor. This is because the other parts are capacitors and inductors, which theoretically do not dissipate any power.

Because power is voltage multiplied by current, the power dissipated in the transistor would be null if either the voltage or the current was null. This is what Class-E amplifiers try to do: ensure that the output transistor never has a simultaneously high voltage across its terminals and a high current going through it….

Class G and Class H are quests for improved efficiency over the classic Class-AB amplifier. Both work on the power supply section. The idea is simple. For high-output power, a high-voltage power supply is needed. For low-power, this high voltage implies higher losses in the output stage.

What about reducing the supply voltage when the required output power is low enough? This scheme is clever, especially for audio applications. Most of the time, music requires only a couple of watts even if far more power is needed during the fortissimo. I agree this may not be the case for some teenagers’ music, but this is the concept.

Class G achieves this improvement by using more than one stable power rail, usually two. Figure 4 shows you the concept.

Figure 4—A Class-G amplifier uses two pairs of power supply rails. b—One supply rail is used when the output signal has a low power (blue). The other supply rail enters into action for high powers (red). Distortion could appear at the crossover.

Figure 4—A Class-G amplifier uses two pairs of power supply rails. b—One supply rail is used when the output signal has a low power (blue). The other supply rail enters into action for high powers (red). Distortion could appear at the crossover.

Simple Guitar Transmitter (EE Tip #102)

You need a guitar amplifier to play an electric guitar. The guitar must be connected with a cable to the amplifier, which you might consider an inconvenience. Most guitar amplifiers operate off the AC power line. An electric guitar fitted with a small transmitter offers several advantages. You can make the guitar audible via an FM tuner/amplifier, for example. Both the connecting cable and amplifier are then unnecessary. With a portable FM broadcast radio or, if desired, a boombox, you can play in the street or in subway.

Source: Elektor 3/2009

Source: Elektor 3/2009

stations (like Billy Bragg). In that case, everything is battery-powered and independent of a fixed power point. (You might need a permit, though.)

Designing a transmitter to do this is not necessary. A variety of low-cost transmitters are available. The range of these devices is often not more than around 30′, but that’s likely plenty for most applications. Consider a König FMtrans20 transmitter. After fitting the batteries and turning it on, you can detect a carrier signal on the radio. Four channels are available, so it should always be possible to find an unused part of the FM band. A short cable with a 3.5-mm stereo audio jack protrudes from the enclosure. This is the audio input. The required signal level for sufficient modulation is about 500 mVPP.

If a guitar is connected directly, the radio’s volume level will have to be high to get sufficient sound. In fact, it will have to be so high that the noise from the modulator will be quite annoying. Thus, a preamplifier for the guitar signal is essential.

To build this preamplifier into the transmitter, you first have to open the enclosure. The two audio channels are combined. This is therefore a single channel (mono) transmitter. Because the audio preamplifier can be turned on and off at the same time as the transmitter, you also can use the transmitter’s on-board power supply for power. In our case, that was about 2.2 V. This voltage is available at the positive terminal of an electrolytic capacitor. Note that 2.2 V is not enough to power an op-amp. But with a single transistor the gain is already big enough and the guitar signal is sufficiently modulated. The final implementation of the modification involves soldering the preamplifier circuit along an edge of the PCB so that everything still fits inside the enclosure. The stereo cable is replaced with a 11.8″ microphone cable, fitted with a guitar plug (mono jack). The screen braid of the cable acts as an antenna as well as a ground connection for the guitar signal. The coil couples the low-frequency signal to ground, while it isolates the high-frequency antenna signal. While playing, the cable with the transmitter just dangles below the guitar, without being a nuisance. If you prefer, you can also secure the transmitter to the guitar with a bit of double-sided tape.

—Gert Baars, “Simple Guitar Transmitter,” Elektor,  080533-1, 3/2009.

Dual-Channel 3G-SDI Video/Audio Capture Card


ADLINK PCIe-2602 Video/Audio Capture Card

The PCIe-2602 is an SDI video/audio capture card that supports all SD/HD/3G-SDI signals and operates at six times the resolution of regular VGA connections. The card also provides video quality with lossless full color YUV 4:4:4 images for sharp, clean images.

The PCIe-2602 is well suited for medical imaging and intelligent video surveillance and analytics. With up to 12-bit pixel depth, the card  provides extreme image clarity and smoother transitions from color-to-color enhance image detail to support critical medical imaging applications, including picture archiving and communication system (PACS) endoscopy and broadcasting.

The card’s features include low latency uncompressed video streaming, CPU offloading, and support for high-quality live viewing for video analytics of real-time image acquisition, as required in casino and defense environments. PCIe-2602 signals can be transmitted over 100 m when combined with a 75-Ω coaxial cable.

The PCIe-2602 is equipped with RS-485 and digital I/O. It accommodates external devices (e.g., PTZ cameras and sensors) and supports Windows 7/XP OSes. The card comes with ADLINK’s ViewCreator Pro utility to enable setup, configuration, testing, and system debugging without any software programming. All ADLINK drivers are compatible with Microsoft DirectShow.

Contact ADLINK for pricing.

ADLINK Technology, Inc.

Low-Cost, High-Performance 32-bit Microcontrollers

The PIC32MX3/4 32-bit microcontrollers are available in 64/16-, 256/64-, and 512/128-KB flash/RAM configurations. The microcontrollers are coupled with Microchip Technology’s software and tools for designs in connectivity, graphics, digital audio, and general-purpose embedded control.

The microcontrollers offer high RAM memory options and high peripheral integration at a low cost. They feature 28 10-bit ADCs, five UARTS, 105-DMIPS performance, serial peripherals, a graphic display, capacitive touch, connectivity, and digital audio support.
The PIC32MX3/4 microcontrollers are supported with general software development tools, including Microchip Technology’s MPLAB X integrated development environment (IDE) and the MPLAB XC32 C/C++ compiler.

Application-specific tools include the Microchip Graphics Display Designer X and the Microchip Graphics Library, which provide a visual design tool that enables quick and easy creation of graphical user interface (GUI) screens for applications. The microcontrollers are also supported with a set of Microchip’s protocol stacks including TCP/IP, USB Device and Host, Bluetooth, and Wi-Fi. For digital audio applications, Microchip provides software for tasks such as sample rate conversion (SRC), audio codecs—including MP3 and Advanced Audio Coding (AAC), and software to connect smartphones and other personal electronic devices.

The PIC32MX3/4 family is supported by Microchip’s PIC32 USB Starter Kit III, which costs $59.99 and the PIC32MX450 100-pin USB plug-in module, which costs $25 for the modular Explorer 16 development system. Pricing for the PIC32MX3/4 microcontrollers starts at $2.50 each in 10,000-unit quantities.

Microchip Technology, Inc.

Microcontroller-Based Markov Music Box

Check out the spectrogram for two FM notes produced by FM modulation. Red indicates higher energy at a given time and frequency.

Cornell University senior lecturer Bruce Land had two reasons for developing an Atmel AVR micrcontroller-based music box. One, he wanted to present synthesis/sequencing algorithms to his students. And two, he wanted the challenge of creating an interactive music box. Interactive audio is becoming an increasingly popular topic among engineers and designers, as we recently reported.

Land writes:

Traditional music boxes play one or two tunes very well, but are not very interactive. Put differently, they have a high quality of synthesis, but a fixed-pattern note sequencer and fixed tonal quality. I wanted to build a device which would play an interesting music-like note sequence, which constantly changed and evolved, with settable timbre, tempo, and beat… To synthesize nice sounding musical notes you need to control spectral content of the note, the rise time (attack), fall time (decay), and the change in spectral content during attack and decay.  Also it is nice to have at least two independent musical voices. And all of this has to be done using the modest arithmetic capability of an 8-bit microcontroller.

Land’s students subsequently used the music box for other projects, such as an auto-composing piano, as shown in the following video.

In early 2013 Circuit Cellar will run Land’s in-depth article on the Markov music box project. Stay tuned for more information.

Prevent Embedded Design Errors (CC 25th Anniversary Preview)

Attention, electrical engineers and programmers! Our upcoming 25th Anniversary Issue (available in early 2013) isn’t solely a look back at the history of this publication. Sure, we cover a bit of history. But the issue also features design tips, projects, interviews, and essays on topics ranging from user interface (UI) tips for designers to the future of small RAM devices, FPGAs, and 8-bit chips.

Circuit Cellar’s 25th Anniversary issue … coming in early 2013

Circuit Cellar columnist Robert Lacoste is one of the engineers whose essay will focus on present-day design tips. He explains that electrical engineering projects such as mixed-signal designs can be tedious, tricky, and exhausting. In his essay, Lacoste details 25 errors that once made will surely complicate (at best) or ruin (at worst) an embedded design project. Below are some examples and tips.

Thinking about bringing an electronics design to market? Lacoste highlights a common error many designers make.

Error 3: Not Anticipating Regulatory Constraints

Another common error is forgetting to plan for regulatory requirements from day one. Unless you’re working on a prototype that won’t ever leave your lab, there is a high probability that you will need to comply with some regulations. FCC and CE are the most common, but you’ll also find local regulations as well as product-class requirements for a broad range of products, from toys to safety devices to motor-based machines. (Refer to my article, “CE Marking in a Nutshell,” in Circuit Cellar 257 for more information.)

Let’s say you design a wireless gizmo with the U.S. market and later find that your customers want to use it in Europe. This means you lose years of work, as well as profits, because you overlooked your customers’ needs and the regulations in place in different locals.

When designing a wireless gizmo that will be used outside the U.S., having adequate information from the start will help you make good decisions. An example would be selecting a worldwide-enabled band like the ubiquitous 2.4 GHz. Similarly, don’t forget that EMC/ESD regulations require that nearly all inputs and outputs should be protected against surge transients. If you forget this, your beautiful, expensive prototype may not survive its first day at the test lab.

Watch out for errors

Here’s another common error that could derail a project. Lacoste writes:

Error 10: You Order Only One Set of Parts Before PCB Design

I love this one because I’ve done it plenty of times even though I knew the risk.

Let’s say you design your schematic, route your PCB, manufacture or order the PCB, and then order the parts to populate it. But soon thereafter you discover one of the following situations: You find that some of the required parts aren’t available. (Perhaps no distributor has them. Or maybe they’re available but you must make a minimum order of 10,000 parts and wait six months.) You learn the parts are tagged as obsolete by its manufacturer, which may not be known in advance especially if you are a small customer.

If you are serious about efficiency, you won’t have this problem because you’ll order the required parts for your prototypes in advance. But even then you might have the same issue when you need to order components for the first production batch. This one is tricky to solve, but only two solutions work. Either use only very common parts that are widely available from several sources or early on buy enough parts for a couple of years of production. Unfortunately, the latter is the only reasonable option for certain components like LCDs.

Ok, how about one more? You’ll have to check out the Anniversary Issue for the list of the other 22 errors and tips. Lacoste writes:

Error 12: You Forget About Crosstalk Between Digital and Analog Signals

Full analog designs are rare, so you have probably some noisy digital signals around your sensor input or other low-noise analog lines. Of course, you know that you must separate them as much as possible, but you can be sure that you will forget it more than once.

Let’s consider a real-world example. Some years ago, my company designed a high-tech Hi-Fi audio device. It included an on-board I2C bus linking a remote user interface. Do you know what happened? Of course, we got some audible glitches on the loudspeaker every time there was an I2C transfer. We redesigned the PCB—moving tracks and adding plenty of grounded copper pour and vias between sensitive lines and the problem was resolved. Of course we lost some weeks in between. We knew the risk, but underestimated it because nothing is as sensitive as a pair of ears. Check twice and always put guard-grounded planes between sensitive tracks and noisy ones.

Circuit Cellar’s Circuit Cellar 25th Anniversary Issue will be available in early 2013. Stay tuned for more updates on the issue’s content.





Q&A: Andrew Spitz (Co-Designer of the Arduino-Based Skube)

Andrew Spitz is a Copenhagen, Denmark-based sound designer, interaction designer, programmer, and blogger studying toward a Master’s interaction design at the Copenhagen Institute of Interaction Design (CIID). Among his various innovative projects is the Arduino-based Skube music player, which is an innovative design that enables users to find and share music.

The Arduino-based Skube

Spitz worked on the design with Andrew Nip, Ruben van der Vleuten, and Malthe Borch. Check out the video to see the Skube in action.

On his blog, Spitz writes:

It is a fully working prototype through the combination of using ArduinoMax/MSP and an XBee wireless network. We access the API to populate the Skube with tracks and scrobble, and using their algorithms to find similar music when in Discover mode.

The following is an abridged  version of an interview that appears in the December 2012 issue of audioXpress magazine, a sister publication of Circuit Cellar magazine..

SHANNON BECKER: Tell us a little about your background and where you live.

Andrew Spitz: I’m half French, half South African. I grew up in France, but my parents are South African so when I was 17, I moved to South Africa. Last year, I decided to go back to school, and I’m now based in Copenhagen, Denmark where I’m earning a master’s degree at the Copenhagen Institute of Interaction Design (CID).

SHANNON: How did you become interested in sound design? Tell us about some of your initial projects.

Andrew: From the age of 16, I was a skydiving cameraman and I was obsessed with filming. So when it was time to do my undergraduate work, I decided to study film. I went to film school thinking that I would be doing cinematography, but I’m color blind and it turned out to be a bigger problem than I had hoped. At the same time, we had a lecturer in sound design named Jahn Beukes who was incredibly inspiring, and I discovered a passion for sound that has stayed with me.

Shannon: What do your interaction design studies at CIID entail? What do you plan to do with the additional education?

Andrew: CIID is focused on a user-centered approach to design, which involves finding intuitive solutions for products, software, and services using mostly technology as our medium. What this means in reality is that we spend a lot of time playing, hacking, prototyping, and basically building interactive things and experiences of some sort.

I’ve really committed to the shift from sound design to interaction design and it’s now my main focus. That said, I feel like I look at design from the lens of a sound designer as this is my background and what has formed me. Many designers around me are very visual, and I feel like my background gives me not only a different approach to the work but also enables me to see opportunities using sound as the catalyst for interactive experiences. Lots of my recent projects have been set in the intersection among technology, sound, and people.

SHANNON: You have worked as a sound effects recordist and editor, location recordist and sound designer for commercials, feature films, and documentaries. Tell us about some of these experiences?

ANDREW: I love all aspects of sound for different reasons. Because I do a lot of things and don’t focus on one, I end up having more of a general set of skills than going deep with one—this fits my personality very well. By doing different jobs within sound, I was able to have lots of different experiences, which I loved! nLocation recording enabled me to see really interesting things—from blowing up armored vehicles with rocket-propelled grenades (RPGs) to interviewing famous artists and presidents. And, documentaries enabled me to travel to amazing places such as Rwanda, Liberia, Mexico, and Nigeria. As a sound effects recordist on Jock of the Bushvelt, a 3-D animation, I recorded animals such as lions, baboons, and leopards in the South African bush. With Bakgat 2, I spent my time recording and editing rugby sounds to create a sound effects library. This time in my life has been a huge highlight, but I couldn’t see myself doing this forever. I love technology and design, which is why I made the move...

SHANNON: Where did the idea for Skube originate?

Andrew: Skube came out of the Tangible User Interface (TUI) class at CIID where we were tasked to rethink audio in the home context. So understanding how and where people share music was the jumping-off point for creating Skube.

We realized that as we move more toward a digital and online music listening experience, current portable music players are not adapted for this environment. Sharing mSkube Videousic in communal spaces is neither convenient nor easy, especially when we all have such different taste in music.

The result of our exploration was Skube. It is a music player that enables you to discover and share music and facilitates the decision process of picking tracks when in a communal setting.

audioXpress is an Elektor International Media publication.

Modify & Test a Phase Meter Calibrator

Charles Hansen described a DIY phase meter calibrator using all-pass, phase-shift filters in a November 2006 article published in audioXpress magazine. Being able to measure phase angle is often helpful, so I’ll begin by quoting from the beginning of his article:

“A phase angle meter is useful in audio work to determine the phase angle between a reference signal and a phase shifted signal, both having identical time periods. Typical uses include: Finding the phase angle between voltage and current to determine the phase shift and impedance of a loudspeaker over its frequency range. Finding the phase shift between the input and output of a tube amplifier to establish the HF (high frequency) and LF (low frequency) cutoff points needed to avoid instability in feedback amplifiers.”

In addition to these, there are other uses—for example, measuring the phase shift through any active or passive filter which includes equalization networks.

In his design, he chose a set of five calibrations frequencies: 10 Hz, 100 Hz, 1000 Hz, 10 kHz, and 100 kHz. He relied on an external oscillator to drive the calibrator at these input frequencies. I first built the calibrator as described, and then I made some modifications that better suited my needs. But first I will describe how the calibrator works. I think it’s best to just provide a bit more  from Hansen’s article:

“The Phase Angle Calibrator makes use of an op amp filter circuit called the all-pass circuit, which takes a sine-wave input and produces a constant amplitude phase-shifted sine wave output. The lag output version was used in Fig. 1. The theory behind the all-pass filter is available in many reference books and texts, but I found one by Walt Jung [1] that I believe is the easiest for a novice to understand. The phase shift angle is varied by the parallel combination of R3 and R9 through R19 with C3 through C7 in accordance with the formula:

θ = -2 arctan (2ΠRC)

where θ is the phase angle, and f is the frequency. After selecting a suitable value for C, you can solve for R by rearranging the formula:

R = tan(-θ/2) / 2ΠfC

This is hardly a linear relationship. Large changes in resistor value produce very little change in phase angle as you approach 0 or 180 degrees. It’s much easier to apply the input signal to both inputs of the phase angle meter for zero degrees, and use an op-amp inverter to generate the 180 degree signal.”


I added an internal Wein-bridge oscillator to simplify using the calibrator and I changed the set of frequencies to cover just the audio range: 20 Hz, 100 Hz, 1000 Hz, 10 kHz, and 20 kHz. (This range is also easier to cover with a single-range oscillator, the capacitor values stay reasonable.) The actual frequencies, shown in Table 1, vary somewhat from the ideal frequencies because I used standard 1% resistors and 5% capacitors.

Table 1: These are phase calibrator phase-shift measurements. The column labeled “305” refers to the Dranetz model 305 phase meter with 305-PA-3007 plug-in. The column labeled “5245L” refers to the Hewlett-Packard model 5245L frequency counter with a model 5262A time interval unit plug-in. Phase shift measurements at 19.6 kHz are not useful from the HP-5245L counter because the 10-MHz timebase does not provide enough significant figures.

Selecting and matching the capacitors would give closer results, but it’s more important to know what the frequencies are. Because I had already built a circuit board for the calibrator circuit, I used a second circuit board for the oscillator. Figure 1 and Figure 2 are the two circuit diagrams.

Figure 1: Phase angle calibrator using all-pass phase-shift filters. This is a Charles Hansen design, 2005, with circuit board design by the author (PHASECAL.PCB).

Figure 2: Wein-bridge oscillator with lamp amplitude stabilization. Adjust R6 for minimum harmonic distortion. (MAIN115.PCB)

Tables 2 and Table 3 are the parts lists. Please note that the calibrator circuit is unchanged from Hansen’s design, except for the values of the capacitors C3 and C7. In the original, C3 was 470 nF (for 10 Hz) and C7 was 47 pF (for 100 kHz). I put both circuit boards and a ±15-VDC power supply (any regulated supply will suffice) in a Wolgram MC-9 enclosure.

Table 2: Calibrator parts list

Table 3: Wein-bridge oscillator parts list

The completed calibrator is shown in Photo 1 with a Dranetz Phase meter. (More about this later.) The unlabeled knob, lower left in the photo, is an oscillator output level control (R8 in Figure 2), which I added after making the front panel label.

Photo 1: A Dranetz automatic phase meter, model 305, is at the top. The phase meter calibrator is below. The calibrator, a Charles Hansen design, is not a TDL product, but construction details are included in this article.

Wein-bridge oscillator theory is discussed in many textbooks and is rather mathematical. I will describe it as simply as possible. In Figure 2 the oscillation frequency is set by the value of R and C connected between the op-amp non-inverting input (pin 3), the op-amp output (pin 6), and common. For a frequency of 1,000 Hz, C = 22 nF (C3 and C10) and R = 7235 Ω (the series combination of R1 + R2 and R3 + R4). The equation is:

f = 1/(2ΠRC) = 1/(2Π(22 x 10-9) (7235)) = 1000 Hz

For amplitude-stable oscillation to occur, the gain of the op-amp circuit must be 1/3. This is set by the impedance of the RC network and resistors R5, R6 and R7 and the incandescent lamp. The lamp is important because it stabilizes the gain at 1/3. If the output voltage (pin 6) tries to increase, the lamp’s resistance decreases and the output voltage decreases. This works very well, but it takes the output amplitude a small of amount of time to stabilize, especially at low frequencies. The CM6833 lamp is very small, so its thermal time constant is very low and stability happens very quickly. The trimmer pot, R6, is adjusted for minimum distortion in the output signal. You can get rather close by looking at the waveform with a scope, but it’s better to use a distortion analyzer or spectrum analyzer. Spectrum analysis software on a PC is fine, just adjust R6 to minimize the height of the sidebands or use a program that directly displays harmonic distortion.

At 1000 Hz, TrueRTA shows the second harmonic (2000 Hz) down 80 dB (0.01% distortion) with the higher harmonics even lower. AudioTester shows a total harmonic distortion of 0.0105% using the first ten harmonics.

TrueRTA is a spectrum analysis program available from True Audio. Demo versions and a free version (level 1) are available on its website. AudioTester is another spectrum analysis program.


The calibrator should be reasonably accurate when built using the 1% resistors and 5% capacitors in the parts list. But as with any other piece of test equipment, it would be satisfying to make some measurements to be sure. I will describe two methods that I used: all the measured values are presented in Table 1. As you can see, the calibrator is very satisfactory.

One method is to use a calibrated phase meter with an accuracy better than the calibrator. I used a Dranetz model 305 (five-digit phase angle display) with a model 305-PA-3007 plug-in.(The Dranetz phase meter is no longer manufactured but used units may be found on eBay or from used electronic instrument dealers.) This plug-in provides automatic operation for input amplitudes of 50 mV RMS to 50 V RMS and frequencies from 2 Hz to 70 kHz. Automatic operation means there are no operating controls. The plug-in scales the input voltage to the mainframe and provides the correct frequency compensation.

Another method is to use a time interval counter to measure the time between an amplitude zero crossing of the reference signal to the amplitude zero crossing of the phase shifted signal. Phase shift can be calculated from the time interval as:

θ = 360τf/1000

where θ is the phase shift in degrees, time delay τ is in milliseconds, and f is the frequency in hertz.

I used a Hewlett-Packard (HP) model 5245L frequency counter with a model 5262A time interval unit plug-in (see Photo 2).

Photo 2: Hewlett-Packard model 5245L frequency counter with a time interval plug-in unit below and my dual zero-crossing detector above.

The 10-MHz counter timebase gives a time resolution of 0.1 ms. The time interval plug-in has trigger-level controls for each channel but they are not calibrated and can’t accurately set the zero crossing with a sine wave input. The smaller “box” above the counter in the photo is a two-channel zero crossing detector. I designed and built this detector to output a pulse whose leading edge coincides in time with the input zero crossing. The counter measures the time between the leading edges of the two pulses: the reference and the phase shifted signal. The detector circuit diagram (see Figure 3) and parts list (see Table 4) are included. I packaged the Detector circuit board with a simple ±5-V regulated power supply in a Wolgram MC-7A enclosure.

Figure 3: The ual zero-crossing detector circuit board

Table 4: The two-channel zero crossing detector's parts list

Looking at one of the detector’s channels in Figure 3, U1 is an input buffer. Resistors R5, R6, and D1 clip the negative-going half of the input sine wave. The comparator circuit (U2) outputs a very short pulse at the input zero crossing. This pulse is “stretched” by the monostable multivibrator in U3 to about 12 ms as set by the time-constant of C1 and R19. Two front panel toggle switches select either the positive-going or negative-going output pulses. The reference and shifted pulses—45° at 10 kHz—are shown in Photo 3.

Photo 3: The digital storage scope display of reference pulse (above) and phase shifted pulse (below) for 45 degrees of shift at 10 kHz. The pulse width is 12 us. The pulse amplitude is 5 V. Pulse baselines are shifted for clarity.


New phase meters are expensive but used models can sometimes be found on eBay or from used electronic test equipment dealers, just try a Google search. In addition to the Dranetz 305 (which I found on eBay), other useful models include:

  • Aerometrics model PM720 phase meter, 5 Hz to 500 kHz, analog meter display. Aerometrics  denies any association with this unit but it is often listed under this name.
  • Hewlett-Packard model 3575A gain-phase meter, 1 Hz to 13 MHz, four-digit display
  • Wavetek model 750 phase meter, 10 Hz to 2 MHz, four-digit display

In addition, you can find application notes and magazine articles that describe how to build your own phase meter. These are usually fairly simple designs. The following appear to be useful: Intersil Application Note AN9637 (This is identical to Design Idea #1890 that was published in the July 4, 1996 issue of EDN); Elliott Sound Products Project 135; and Salvati, M. J., “Phase Meter Profits From Improvements,” Design Idea, Electronic Design magazine, April 11, 1991.


I sent a copy of this article to Hansen for comments. He agreed that having the oscillator built-in is a good feature. He also commented as follows:

“A problem with my phase meter calibrator design is that the distortion increases with phase shift, and the amplitude drops as well. It might be possible that the zero-crossing detector might be fooled by the higher order distortion harmonics. I’d be interested in what you find out in this regard.”

So, I measured the amplitude drop and distortion at 150°, which should be worst case. I set the 20-Hz variable output to an arbitrary 2.00 V. Keeping the output level control unchanged, I measured what you see in Table 5. This amount of drop seems acceptable.

Table 5: I set the 20-Hz variable output to 2 V, and I kept the output level control unchanges as I measured these.

I used a Hewlett-Packard model 3581A wave analyzer to measure the harmonics. Refer to Table 6. These numbers look acceptable and the zero-crossing detector output at 20 kHz and 150 degrees measures 22 ms on an oscilloscope with a calculated 21.5 ms at the actual frequency of 19.61 kHz.

Table 6: I used a Hewlett-Packard 3581A wave analyzer to measure the harmonics. These numbers are acceptable and the zero-crossing detector output at 20 kHz and 150 degrees measures 22 ms on an oscilloscope with a calculated 21.5 ms at the actual frequency of 19.61 kHz.

I am very satisfied that the calibrator is suitable to troubleshoot and calibrate any phase meter you are likely to find, either new or used. Without overdoing the math, there is enough design information here to allow you to tailor the design to a specific frequency range, keeping in mind the 1000:1 practical frequency range of the Wein-bridge oscillator, without using range switching.

The circuit board designs listed in the parts lists are available in CIRCAD format and are posted on the TDL website. (CIRCAD is a circuit board design program available from Holophase. The boards in the file were designed with Version 4, a free download of which is available on the Holophase website.) The physical boards are not available.

Ron Tipton lives in Las Cruces, NM. Visit the TDL Technology website for more information about his audio designs and services.


[1] Jung, W. and Sams, H., Audio IC Op-Amp Applications, 2nd Edition, Sams Publishing, 1978.

Editor’s note: audioXpress, like, is an Elektor International Media publication.

Seven-Controller EtherCAT Orchestra

When I first saw the Intel Industrial Control in Concert demonstration at Design West 2012 in San Jose, CA, I immediately thought of Kurt Vonnegut ‘s 1952 novel Player Piano. The connection, of course, is that the player piano in the novel and Intel’s Atom-based robotic orchestra both play preprogrammed music without human involvement. But the similarities end there. Vonnegut used the self-playing autopiano as a metaphor for a mechanized society in which wealthy industrialists replaced human workers with automated machines. In contrast, Intel’s innovative system demonstrated engineering excellence and created a buzz in the in the already positive atmosphere at the conference.

In “EtherCAT Orchestra” (Circuit Cellar 264, July 2012), Richard Wotiz carefully details the awe-inspiring music machine that’s built around seven embedded systems, each of which is based on Intel’s Atom D525 dual-core microprocessor. He provides information about the system you can’t find on YouTube or hobby tech blogs. Here is the article in its entirety.

EtherCAT Orchestra

I have long been interested in automatically controlled musical instruments. When I was little, I remember being fascinated whenever I ran across a coin-operated electromechanical calliope or a carnival hurdy-gurdy. I could spend all day watching the many levers, wheels, shafts, and other moving parts as it played its tunes over and over. Unfortunately, the mechanical complexity and expertise needed to maintain these machines makes them increasingly rare. But, in our modern world of pocket-sized MP3 players, there’s still nothing like seeing music created in front of you.

I recently attended the Design West conference (formerly the Embedded Systems Conference) in San Jose, CA, and ran across an amazing contraption that reminded me of old carnival music machines. The system was created for Intel as a demonstration of its Atom processor family, and was quite successful at capturing the attention of anyone walking by Intel’s booth (see Photo 1).

Photo 1—This is Intel’s computer-controlled orchestra. It may not look like any musical instrument you’ve ever seen, but it’s quite a thing to watch. The inspiration came from Animusic’s “Pipe Dream,” which appears on the video screen at the top. (Source: R. Wotiz)

The concept is based on Animusic’s music video “Pipe Dream,” which is a captivating computer graphics representation of a futuristic orchestra. The instruments in the video play when virtual balls strike against them. Each ball is launched at a precise time so it will land on an instrument the moment each note is played.

The demonstration, officially known as Intel’s Industrial Control in Concert, uses high-speed pneumatic valves to fire practice paintballs at plastic targets of various shapes and sizes. The balls are made of 0.68”-diameter soft rubber. They put on quite a show bouncing around while a song played. Photo 2 shows one of the pneumatic firing arrays.

Photo 2—This is one of several sets of pneumatic valves. Air is supplied by the many tees below the valves and is sent to the ball-firing nozzles near the top of the photo. The corrugated hoses at the top supply balls to the nozzles. (Source: R. Wotiz)

The valves are the gray boxes lined up along the center. When each one opens, a burst of air is sent up one of the clear hoses to a nozzle to fire a ball. The corrugated black hoses at the top supply the balls to the nozzles. They’re fed by paintball hoppers that are refilled after each performance. Each nozzle fires at a particular target (see Photo 3).

Photo 3—These are the targets at which the nozzles from Photo 2 are aimed. If you look closely, you can see a ball just after it bounced off the illuminated target at the top right. (Source: R. Wotiz)

Each target has an array of LEDs that shows when it’s activated and a piezoelectric sensor that detects a ball’s impact. Unfortunately, slight variations in the pneumatics and the balls themselves mean that not every ball makes it to its intended target. To avoid sounding choppy and incomplete, the musical notes are triggered by a fixed timing sequence rather than the ball impact sensors. Think of it as a form of mechanical lip syncing. There’s a noticeable pop when a ball is fired, so the system sounds something like a cross between a pinball machine and a popcorn popper. You may expect that to detract from the music, but I felt it added to the novelty of the experience.

The control system consists of seven separate embedded systems, all based on Intel’s Atom D525 dual-core microprocessor, on an Ethernet network (see Figure 1).

Figure 1—Each block across the top is an embedded system providing some aspect of the user interface. The real-time interface is handled by the modules at the bottom. They’re controlled by the EtherCAT master at the center. (Source. R. Wotiz)

One of the systems is responsible for the real-time control of the mechanism. It communicates over an Ethernet control automation technology (EtherCAT) bus to several slave units, which provide the I/O interface to the sensors and actuators.


EtherCAT is a fieldbus providing high-speed, real-time control over a conventional 100 Mb/s Ethernet hardware infrastructure. It’s a relatively recent technology, originally developed by Beckhoff Automation GmbH, and currently managed by the EtherCAT Technology Group (ETG), which was formed in 2003. You need to be an ETG member to access most of their specification documents, but information is publicly available. According to information on the ETG website, membership is currently free to qualified companies. EtherCAT was also made a part of international standard IEC 61158 “Industrial Communication Networks—Fieldbus Specifications” in 2007.

EtherCAT uses standard Ethernet data frames, but instead of each device decoding and processing an individual frame, the devices are arranged in a daisy chain, where a single frame is circulated through all devices in sequence. Any device with an Ethernet port can function as the master, which initiates the frame transmission. The slaves need specialized EtherCAT ports. A two-port slave device receives and starts processing a frame while simultaneously sending it out to the next device (see Figure 2).

Figure 2—Each EtherCAT slave processes incoming data as it sends it out the downstream port. (Source: R. Wotiz))

The last slave in the chain detects that there isn’t a downstream device and sends its frame back to the previous device, where it eventually returns to the originating master. This forms a logical ring by taking advantage of both the outgoing and return paths in the full-duplex network. The last slave can also be directly connected to a second Ethernet port on the master, if one is available, creating a physical ring. This creates redundancy in case there is a break in the network. A slave with three or more ports can be used to form more complex topologies than a simple daisy chain. However, this wouldn’t speed up network operation, since a frame still has to travel through each slave, one at a time, in both directions.

The EtherCAT frame, known as a telegram, can be transmitted in one of two different ways depending on the network configuration. When all devices are on the same subnet, the data is sent as the entire payload of an Ethernet frame, using an EtherType value of 0x88A4 (see Figure 3a).

Figure 3a—An EtherCAT frame uses the standard Ethernet framing format with very little overhead. The payload size shown includes both the EtherCAT telegram and any padding bytes needed to bring the total frame size up to 64 bytes, the minimum size for an Ethernet frame. b—The payload can be encapsulated inside a UDP frame if it needs to pass through a router or switch. (Source: R. Wotiz)

If the telegrams must pass through a router or switch onto a different physical network, they may be encapsulated within a UDP datagram using a destination port number of 0x88A4 (see Figure 3b), though this will affect network performance. Slaves do not have their own Ethernet or IP addresses, so all telegrams will be processed by all slaves on a subnet regardless of which transmission method was used. Each telegram contains one or more EtherCAT datagrams (see Figure 4).

Each datagram includes a block of data and a command indicating what to do with the data. The commands fall into three categories. Write commands copy the data into a slave’s memory, while read commands copy slave data into the datagram as it passes through. Read/write commands do both operations in sequence, first copying data from memory into the outgoing datagram, then moving data that was originally in the datagram into memory. Depending on the addressing mode, the read and write operations of a read/write command can both access the same or different devices. This enables fast propagation of data between slaves.

Each datagram contains addressing information that specifies which slave device should be accessed and the memory address offset within the slave to be read or written. A 16-bit value for each enables up to 65,535 slaves to be addressed, with a 65,536-byte address space for each one. The command code specifies which of four different addressing modes to use. Position addressing specifies a slave by its physical location on the network. A slave is selected only if the address value is zero. It increments the address as it passes the datagram on to the next device. This enables the master to select a device by setting the address value to the negative of the number of devices in the network preceding the desired device. This addressing mode is useful during system startup before the slaves are configured with unique addresses. Node addressing specifies a slave by its configured address, which the master will set during the startup process. This mode enables direct access to a particular device’s memory or control registers. Logical addressing takes advantage of one or more fieldbus memory management units (FMMUs) on a slave device. Once configured, a FMMU will translate a logical address to any desired physical memory address. This may include the ability to specify individual bits in a data byte, which provides an efficient way to control specific I/O ports or register bits without having to send any more data than needed. Finally, broadcast addressing selects all slaves on the network. For broadcast reads, slaves send out the logical OR of their data with the data from the incoming datagram.

Each time a slave successfully reads or writes data contained in a datagram, it increments the working counter value (see Figure 4).

Figure 4—An EtherCAT telegram consists of a header and one or more datagrams. Each datagram can be addressed to one slave, a particular block of data within a slave, or multiple slaves. A slave can modify the datagram’s Address, C, IRQ, Process data, and WKC fields as it passes the data on to the next device. (Source: R. Wotiz)

This enables the master to confirm that all the slaves it was expecting to communicate with actually handled the data sent to them. If a slave is disconnected, or its configuration changes so it is no longer being addressed as expected, then it will no longer increment the counter. This alerts the master to rescan the network to confirm the presence of all devices and reconfigure them, if necessary. If a slave wants to alert the master of a high-priority event, it can set one or more bits in the IRQ field to request the master to take some predetermined action.


Frames are processed in each slave by a specialized EtherCAT slave controller (ESC), which extracts incoming data and inserts outgoing data into the frame as it passes through. The ESC operates at a high speed, resulting in a typical data delay from the incoming to the outgoing network port of less than 1 μs. The operating speed is often dominated by how fast the master can process the data, rather than the speed of the network itself. For a system that runs a process feedback loop, the master has to receive data from the previous cycle and process it before sending out data for the next cycle. The minimum cycle time TCYC is given by: TCYC = TMP + TFR + N × TDLY  + 2 × TCBL + TJ. TMP = master’s processing time, TFR = frame transmission time on the network (80 ns per data byte + 5 μs frame overhead), N = total number of slaves, TDLY  = sum of the forward and return delay times through each slave (typically 600 ns), TCBL = cable propagation delay (5 ns per meter for Category 5 Ethernet cable), and TJ = network jitter (determined by master).[1]

A slave’s internal processing time may overlap some or all of these time windows, depending on how its I/O is synchronized. The network may be slowed if the slave needs more time than the total cycle time computed above. A maximum-length telegram containing 1,486 bytes of process data can be communicated to a network of 1,000 slaves in less than 1 ms, not including processing time.

Synchronization is an important aspect of any fieldbus. EtherCAT uses a distributed clock (DC) with a resolution of 1 ns located in the ESC on each slave. The master can configure the slaves to take a snapshot of their individual DC values when a particular frame is sent. Each slave captures the value when the frame is received by the ESC in both the outbound and returning directions. The master then reads these values and computes the propagation delays between each device. It also computes the clock offsets between the slaves and its reference clock, then uses these values to update each slave’s DC to match the reference. The process can be repeated at regular intervals to compensate for clock drift. This results in an absolute clock error of less than 1 μs between devices.


The orchestra’s EtherCAT network is built around a set of modules from National Instruments. The virtual conductor is an application running under LabVIEW Real-Time on a CompactRIO controller, which functions as the master device. It communicates with four slaves containing a mix of digital and analog I/O and three slaves consisting of servo motor drives. Both the master and the I/O slaves contain a FPGA to implement any custom local processing that’s necessary to keep the data flowing. The system runs at a cycle time of 1 ms, which provides enough timing resolution to keep the balls properly flying.

I hope you’ve enjoyed learning about EtherCAT—as well as the fascinating musical device it’s used in—as much as I have.

Author’s note: I would like to thank Marc Christenson of SISU Devices, creator of this amazing device, for his help in providing information on the design.


[1] National Instruments Corp., “Benchmarks for the NI 9144 EtherCAT Slave Chassis,”


Animusic, LLC,

Beckhoff Automation GmbH, “ET1100 EtherCAT Slave Controller Hardware Data Sheet, Version 1.8”, 2010,

EtherCAT Technology Group, “The Ethernet Fieldbus”, 2009,

Intel, Atom microprocessor, www/us/en/processors/atom/atom-processor.html.


Atom D525 dual-core microprocessor

Intel Corp.

LabVIEW Real-Time modules, CompactRIO controller, and EtherCAT devices

National Instruments Corp.

Circuit Cellar 264 is now on newsstands, and it’s available at the CC-Webshop.

Simple Circuits: Turn a Tube Radio Into an MP3 Amp

Want to give your MP3 player vintage tube sound? You can with the proper circuits, an antique radio, and a little know-how. In addition to generating amazing sound, the design will be an eye catcher in your home or office.

Here I present excerpts from Bill Reeve’s article, “Repurposing Antique Radios as Tube Amplifiers,” in which he provides vintage radio resources, simple circuit diagrams, and essential part info. He also covers the topics of external audio mixing and audio switching. The article appeared in the May 2012 edition of audioXpress magazine.

Manufactured from the 1930s through the 1960s, vacuum tube radios often contain high-quality audio amplifiers at the end of their RF signal chain. You can repurpose these radios into vintage, low-power tube amplifiers—without marring them in any way or detracting from their original charm and functionality as working analog radios.

Wood-cased radios have especially good sound quality, and the battery compartments in antique “portable” radios (like the Philco 48-360 or the Zenith Transoceanics) provide perfect locations for additional circuitry. When restored properly, large furniture-style radios that were built for “high fidelity” (like the late 1930s and early 1940s Philco console radios) can fill a room with rich beautiful sound.

Simple Circuits

The simple circuits described in this article perform two functions. They mix an external line-level stereo signal (typically from an MP3 player or computer) and reference it to the radio’s circuit. They also use the radio’s on/off knob to switch this external signal to the radio’s audio amplifier.

There is not one circuit that will work for every antique radio. (Original schematics for antique tube radios are available on the web But the circuits described here can be adapted to any radio topology. All the parts can be ordered from an electronics supplier like Digi-Key, and the circuit can be soldered on a prototyping printed circuit board (such as RadioShack P/N 276-168B).

External audio mixing

Figure 1 and Figure 2 show some examples of circuit schematics that mix the line-level stereo audio signals together (almost all tube radios are monophonic), while providing galvanic isolation from high voltages within the radio. Figure 1 shows an inexpensive solution suitable for most table-top radios.

Figure 1: An inexpensive circuit for mixing an MP3 player’s stereo audio signals safely into an antique radio. None of the component values are critical. (Source: B. Reeve, AX 5/12)

These radios have relatively small speakers that are unable to reproduce deep bass, so an inexpensive audio transformer (available from on-line distributors) does the job. I picked up a bucket of Tamura TY-300PR transformers for $0.50 each at an electronics surplus store, and similar transformers are commercially available. Alternatively, the Hammond 560G shown in Figure 2 is an expensive, highquality audio transformer suitable to high-fidelity radios (like the furniture-sized Philco consoles). A less expensive (and fine-sounding) alternative is the Hammond 148A.

Figure 2: A high-fidelity circuit for mixing external stereo audio signals safely into an antique radio. (Source: B. Reeve, AX 5/12)

I use Belden 9154 twisted, shielded audio cable for wiring internal to the radio, but twisted, 24-gauge wire will work well. An 8′ long audio cable with a 3.5-mm stereo jack on each end can be cut in half to make input cables for two radios, or you can use the cord from trashed ear-buds. You can route the audio cable out the back of the chassis. Photo 1 is a photograph of a 1948 Philco portable tube radio restored and used as an MP3 player amplifier.

Photo 1: A 1948 Philco portable tube radio restored and repurposed as an MP3 amplifier. (Source: B. Reeve, AX 5/12)

Audio switching using the radio’s on/off knob

After creating the mixed, radio-referenced signal, the next step is to build a circuit that switches the voltage driving the radio’s audio amplifier between its own internal broadcast and the external audio signal.

Figure 3 illustrates this audio routing control using the radio’s existing front panel power knob. Turn the radio on, and it behaves like the old analog radio it was designed to be (after the tubes warm up). However, if you turn the radio off, then on again within a few of seconds, the external audio signal is routed to the radio’s tube amplifier and speaker.

The circuit shown in Figure 3 uses a transformer to create the low voltage used by the switching circuit. There are many alternative power transformers available, and many methods of creating a transformerless power supply. Use your favorite….

The next photos (see Photo 2a and Photo 2b) show our additional circuit mounted in the lower (battery) compartment of a Zenith Transoceanic AM/shortwave receiver. Note the new high-voltage (B+) capacitors (part of the radio’s restoration) attached to a transformer housing with blue tie wraps.

Photo 2a: The inside view of a Zenith Transoceanic AM/shortwave radio restored and augmented as an MP3 audio amplifier. b: This is an outside view of the repurposed Zenith Transoceanic AM/shortwave radio. (Source: B. Reeve, AX 5/12)

The added circuit board that performs the audio re-routing is mounting to a 0.125″ maple plywood base, using screws countersunk from underneath. The plywood is securely screwed to the inside base of the radio housing. Rubber grommets are added wherever cables pass through the radio’s steel frame.—Bill Reeve

Click here to view the entire article. The article is password protected. To access it, “ax” and the author’s last name (no spaces). and audioXpress are Elektor International Media publications.   

A Workspace for Radio & Metrology Projects

Ralph Berres, a television technician in Germany, created an exemplary design space in his house for working on projects relating to his two main technical interests: amateur radio and metrology (the science of measurement). He even builds his own measurement equipment for his bench.

Ralph Berres built this workspace for his radio and metrology projects

“I am a licensed radio amateur with the call sign DF6WU… My hobby is high-frequency and low-frequency metrology,” Berres wrote in his submission.

Amateur radio is popular among Circuit Cellar readers. Countless electrical engineers and technical DIYers I’ve met or worked with during the past few years are amateur radio operators. Some got involved in radio during childhood. Others obtained radio licenses more recently. For instance, Rebecca Yang of chronicled the process in late 2011. Check it out: and

Do you want to share images of your workspace, hackspace, or “circuit cellar” with the world? Click here to email us your images and workspace info.


Project Spotlight: Electronics + Wood Fab Speakers

MIT graduate student David Mellis is interested in how designers are combining high-tech parts like microcontrollers with low-tech materials in clever ways. Yesterday, I pointed everyone to Mellis’s inspiring 3-D Printed Mouse project. Now let’s look at another creative design—Fab Speakers.

Whether you’re a microcontroller fanatic, professional engineer, audiophile, musician, or all of the aforementioned, this open-source Fab Speakers project will surely inspire you to customize your own. I’d love to see how others tackle a similar DIY project!

Fab Speakers (Source: D. Mellis)

Mellis writes:

These portable speakers are made from laser-cut wood, fabric, veneer, and electronics. They are powered by three AAA batteries and compatible with any standard audio jack (e.g. on an iPhone, iPod, or laptop).

The speakers are an experiment in open-source hardware applied to consumer electronics. By making their original design files freely available online, in a way that’s easy for others to modify, I hope to encourage people to make and modify them. In particular, I’d love to see changes or additions that I didn’t think about and to have those changes shared publicly for others to use or continue to modify. The speakers have been designed to be relatively simple and cheap in the hopes of facilitating their production by others …

Use 6mm (1/4″) plywood. For the veneer, 1 9/16″ edging backed with an iron-on adhesive is ideal (like this one from Rockler), but anything should work if you cut it to that width. Pick whatever fabric you like. For the electronic components, see the bill-of-materials above. You’ll also need two-conductor speaker wire, available at Radio Shack… There’s also a wall-mounted, oval-shaped variation on the design. It uses the same circuit board, but combines both speakers into a single unit that can hang on a nail or screw in the wall. You’ll want to replace the batteries with a 5V power supply (included in the bill of materials); just cut off the connector and solder the wires directly into the + and – holes for the battery holder. You’ll also want to omit the power switch and just solder together the holes where it would have gone.

The design's battery holder (Source: D. Mellis)

Mellis gave me permission to write about the projects and post some of the photos from his website.

Click here to check out all the files for this project.

Hollow-State Amps & Frequency Response

“Glass audio” has been growing in popularity among average audio enthusiasts for the past decade. Music-loving consumers worldwide enjoy the look and sound (i.e., the “warmth”) of tube amps, and innovative companies are creating demand by selling systems featuring tubes, iPod/MP3 hookups, and futuristic-looking enclosures. I suspect hybrid modern/retro designs will continue to gain popularity.

Many serious audiophiles enjoy incorporating glass tubes in their custom audio designs to create the sounds and audio system aesthetics to match their tastes. If you’re a DIYer of this sort, you’ll benefit from knowing how amps work and understanding topics such as frequency responses. In the April 2012 issue of audioXpress, columnist Richard Honeycutt details just that in his article titled “The Frequency Response of Hollow-State Amplifiers.”

Below is an excerpt from Honeycutt’s article. Click the link at the bottom of this post to read the entire article.

Early electronic devices were intended mainly for speech amplification and reproduction. By the 1930s, however, musical program material gained importance, and an extended frequency response became a commercial necessity. This emphasis grew until, in the 1950s and 1960s, the Harmon Kardon Citation audio amplifier claimed frequency response from 1 to 100,000 Hz flat within a decibel or better. Although today, other performance metrics have surpassed frequency response in advertising emphasis—in part because wide, flat frequency response is now easier to obtain with modern circuitry—frequency response remains a very important parameter …

Just which factors determine the low- and high-frequency limitations of vacuum tube amplifiers? In order to examine these factors, we need to discuss a bit of electric circuit theory. If a voltage source—AC or DC, it doesn’t matter—is connected to a resistance, the resulting current is given by Ohm’s Law: I = V/R. If the voltage source is of the AC variety, and the resistor is replaced by a capacitor or inductor, the current is given by: I = V/X where X is the reactance of the capacitor or inductor. Reactance limits current flow by means of temporary energy storage: capacitive reactance XC does so via the electric field, and inductive reactance XL stores energy in the magnetic field.

Figure 1 - The values of reactance provided by a 0.1-μF capacitor and a 254-mH inductor, for a frequency range of 10 to 30,000 Hz (Source: R. Honeycutt, AX April 2012)

Figure 1 shows the values of reactance provided by a 0.1 μF capacitor and a 254 mH inductor, for a frequency range of 10 to 30,000 Hz. Notice that capacitive reactance decreases with frequency; whereas, inductive reactance increases as frequency increases.

Click here to read the entire article.

audioXpress is an Elektor group publication.



Weekly Elektor Wrap Up: Preamplifier 2012, Pico C, & a Webshop Hunt

It’s time for our Friday Elektor wrap up. Our Elektor colleagues were hard at work during this first week of April. Here’s quick review.

Elektor Preamplifier 2012: The Sound of Silence

Elektor has a 40-year history of high-end audio (tube and solid state) coverage: projects, books, circuit boards, and even DVDs. The latest project is the Preamplifier 2012, which was designed by renowned audio specialist Douglas Self, with Elektor audio staffer Ton Giesberts doing the board designs and testing on Elektor’s $50,000 audio precision analyzer! It achieves incredibly low noise figures using low impedance design techniques throughout, but still based on an affordable and easy-to-find opamp: the NE5532. The Preamplifier 2012’s most notable characteristics are its ultra low noise MC/MD section (get out your vinyl records) and the remarkably low-value pots in the Baxandall tone control (like 1-kΩ).Douglas Self and Elektor Audio Labs already stunned the audio community with their NE5532 Op-amplifier a while ago with 32 NE5532 op-amps basically paralleled on a board producing 10 W of extremely high-quality sound. Simply put: they know what they’re doing!You can read about the seven-board design in the April 2012 edition. In fact, why not follow the series?

Part 1:

Part 2:

Part 3: currently in editing for June 2012 edition.

NE5532 Opamplifier:

Pico C Webinar Announcement

Elektor announced this week that it will run a new webinar via element14 on the Elektor Pico C meter, which was featured in the April 2011 editions. The Pico C meter can measure small capacitances. In February 2012 the device was upgraded with new firmware.

According to an Elektor news item, UK-based author/designer Jon Drury will run the webinar slated for Thursday, April 19, 2012. He’ll cover a unique way of giving the original instrument a much wider range while also extending its functionality, all with new software and practically no changes to the existing Pico C hardware. Microcontroller fans, including AVR enthusiasts, can also learn how to adapt the software for different calibration capacitors. Elektor staffers are reporting that Jon may also give a sneak preview of his PicoLO oscilloscope and Pico DDS generator.  You can register at element14.

“E” Hunt!

In other news, Elektor is challenging you to find hidden Easter eggs in its webshop. Find eggs, get a discount. Click here to get started.



FPGA-Based VisualSonic Design Project

The VisualSonic Studio project on display at Design West last week was as innovative as it was fun to watch in operation. The design—which included an Altera DE2-115 FPGA development kit and a Terasic 5-megapixel CMOS Sensor (D5M)—used interactive tokens to control computer-generated music.

at Design West 2012 in San Jose, CA (Photo: Circuit Cellar)

I spoke with Allen Houng, Strategic Marketing Manager for Terasic, about the project developed by students from National Taiwan University. He described the overall design, and let me see the Altera kit and Terasic sensor installation.

A view of the kit and sensor (Photo: Circuit Cellar)

Houng also he also showed me the design in action. To operate the sound system, you simply move the tokens to create the sound effects of your choosing. Below is a video of the project in operation (Source: Terasic’s YouTube channel).