Q&A: Hacker, Roboticist, and Website Host

Dean “Dino” Segovis is a self-taught hardware hacker and maker from Pinehurst, NC. In 2011, he developed the Hack A Week website, where he challenges himself to create and post weekly DIY projects. Dino and I recently talked about some of his favorite projects and products. —Nan Price, Associate Editor

 

NAN: You have been posting a weekly project on your website, Hack A Week, for almost three years. Why did you decide to create the website?

Dean "Dino" Segovis at his workbench

Dean “Dino” Segovis at his workbench

DINO: One day on the Hack A Day website I saw a post that caught my attention. It was seeking a person to fill a potential position as a weekly project builder and video blogger. It was offering a salary of $35,000 a year, which was pretty slim considering you had to live in Santa Monica, CA. I thought, “I could do that, but not for $35,000 a year.”

That day I decided I was going to challenge myself to come up with a project and video each week and see if I could do it for at least one year. I came up with a simple domain name, www.hackaweek.com, bought it, and put up a website within 24 h.

My first project was a 555 timer-based project that I posted on April 1, 2011, on my YouTube channel, “Hack A Week TV.” I made it through the first year and just kept going. I currently have more than 3.2 million video views and more than 19,000 subscribers from all over the world.

NAN: Hack A Week features quite a few robotics projects. How are the robots built? Do you have a favorite?

rumblebot head

Dino’s very first toy robot hack was the Rumble robot. The robot featured an Arduino that sent PWM to the on-board H-bridge in the toy to control the motors for tank steering. A single PING))) sensor helped with navigation.

Rumble robot

The Rumble robot

DINO: I usually use an Arduino as the robot’s controller and Roomba gear motors for locomotion. I have built a few others based on existing wheeled motorized toys and I’ve made a few with the Parallax Propeller chip.

My “go-to” sensor is usually the Parallax PING))) ultrasonic sensor. It’s easy to connect and work with and the code is straightforward. I also use bump sensors, which are just simple contact switches, because they mimic the way some insects navigate.

Nature is a great designer and much can be learned from observing it. I like to keep my engineering simple because it’s robust and easy to repair. The more you complicate a design, the more it can do. But it also becomes more likely that something will fail. Failure is not a bad thing if it leads to a better design that overcomes the failure. Good design is a balance of these things. This is why I leave my failures and mistakes in my videos to show how I arrive at the end result through some trial and error.

My favorite robot would be “Photon: The Video and Photo Robot” that I built for the 2013 North Carolina Maker Faire. It’s my masterpiece robot…so far.

NAN: Tell us a little more about Photon. Did you encounter any challenges while developing the robot?

Photon awaits with cameras rolling, ready to go forth and record images.

Photon awaits with cameras rolling, ready to go forth and record images.

DINO: The idea for Photon first came to me in February 2013. I had been playing with the Emic 2 text-to-speech module from Parallax and I thought it would be fun to use it to give a robot speech capability. From there the idea grew to include cameras that would record and stream to the Internet what the robot saw and then give the robot the ability to navigate through the crowd at Maker Faire.

I got a late start on the project and ended up burning the midnight oil to get it finished in time. One of the bigger challenges was in designing a motorized base that would reliably move Photon across a cement floor.

The problem was in dealing with elevation changes on the floor covering. What if Photon encountered a rug or an extension cord?

I wanted to drive it with two gear motors salvaged from a Roomba 4000 vacuum robot to enable tank-style steering. A large round base with a caster at the front and rear worked well, but it would only enable a small change in surface elevation. I ended up using that design and made sure that it stayed away from anything that might get it in trouble.

The next challenge was giving Photon some sensors so it could navigate and stay away from obstacles. I used one PING))) sensor mounted on its head and turned the entire torso into a four-zone bump sensor, as was a ring around the base. The ring pushed on a series of 42 momentary contact switches connected together in four zones. All these sensors were connected to an Arduino running some simple code that turned Photon away from obstacles it encountered. Power was supplied by a motorcycle battery mounted on the base inside the torso.

The head held two video cameras, two smartphones in camera mode, and one GoPro camera. One video camera and the GoPro were recording in HD; the other video camera was recording in time-lapse mode. The two smartphones streamed live video, one via 4G to a Ustream channel and the other via Wi-Fi. The Ustream worked great, but the Wi-Fi failed due to interference.

Photon’s voice came from the Emic 2 connected to another Arduino sending it lines of text to speak. The audio was amplified by a small 0.5-W LM386 amplifier driving a 4” speaker. An array of blue LEDs mounted on the head illuminated with the brightness modulated by the audio signal when Photon spoke. The speech was just a lot of lines of text running in a timed loop.

Photon’s brain includes two Arduinos and an LM386 0.5-W audio amplifier with a sound-to-voltage circuit added to drive the mouth LED array. Photon’s voice comes from a Parallax Emic 2 text-to-speech module.

Photon’s brain includes two Arduinos and an LM386 0.5-W audio amplifier with a sound-to-voltage circuit added to drive the mouth LED array. Photon’s voice comes from a Parallax Emic 2 text-to-speech module.

Connecting all of these things together was very challenging. Each component needed a regulated power supply, which I built using LM317T voltage regulators. The entire current draw with motors running was about 1.5 A. The battery lasted about 1.5 h before needing a recharge. I had an extra battery so I could just swap them out during the quick charge cycle and keep downtime to a minimum.

I finished the robot around 11:00 PM the night before the event. It was a hit! The videos Photon recorded are fascinating to watch. The look of wonder on people’s faces, the kids jumping up to see themselves in the monitors, the smiles, and the interaction are all very interesting.

NAN: Many of your Hack A Week projects include Parallax products. Why Parallax?

DINO: Parallax is a great electronics company that caters to the DIY hobbyist. It has a large knowledge base on its website as well as a great forum with lots of people willing to help and share their projects.

About a year ago Parallax approached me with an offer to supply me with a product in exchange for featuring it in my video projects on Hack A Week. Since I already used and liked the product, it was a perfect offer. I’ll be posting more Parallax-based projects throughout the year and showcasing a few of them on the ELEV-8 quadcopter as a test platform.

NAN: Let’s change topics. You built an Electronic Fuel Injector Tester, which is featured on HomemadeTools.net. Can you explain how the 555 timer chips are used in the tester?

DINO: 555 timers are great! They can be used in so many projects in so many ways. They’re easy to understand and use and require only a minimum of external components to operate and configure.

The 555 can run in two basic modes: monostable and astable.

Dino keeps this fuel injector tester in his tool box at work. He’s a European auto technician by day.

Dino keeps this fuel injector tester in his tool box at work. He’s a European auto technician by day.

An astable circuit produces a square wave. This is a digital waveform with sharp transitions between low (0 V) and high (+ V). The durations of the low and high states may be different. The circuit is called astable because it is not stable in any state: the output is continually changing between “low” and “high.”

A monostable circuit produces a single output pulse when triggered. It is called a monostable because it is stable in just one state: “output low.” The “output high” state is temporary.

The injector tester, which is a monostable circuit, is triggered by pressing the momentary contact switch. The single-output pulse turns on an astable circuit that outputs a square-wave pulse train that is routed to an N-channel MOSFET. The MOSFET turns on and off and outputs 12 V to the injector. A flyback diode protects the MOSFET from the electrical pulse that comes from the injector coil when the power is turned off and the field collapses. It’s a simple circuit that can drive any injector up to 5 A.

This is a homebrew PCB for Dino's fuel injector tester. Two 555s drive a MOSFET that switches the injector.

This is a homebrew PCB for Dino’s fuel injector tester. Two 555s drive a MOSFET that switches the injector.

NAN: You’ve been “DIYing” for quite some time. How and when did your interest begin?

DINO: It all started in 1973 when I was 13 years old. I used to watch a TV show on PBS called ZOOM, which was produced by WGBH in Boston. Each week they had a DIY project they called a “Zoom-Do,” and one week the project was a crystal radio. I ordered the Zoom-Do instruction card and set out to build one. I got everything put together but it didn’t work! I checked and rechecked everything, but it just wouldn’t work.

I later realized why. The instructions said to use a “cat’s whisker,” which I later found out was a thin piece of wire. I used a real cat’s whisker clipped from my cat! Anyway, that project sparked something inside me (pun intended). I was hooked! I started going house to house asking people if they had any broken or unwanted radios and or TVs I could have so I could learn about electronics and I got tons of free stuff to mess with.

My mom and dad were pretty cool about letting me experiment with it all. I was taking apart TV sets, radios, and tape recorders in my room and actually fixing a few of them. I was in love with electronics. I had an intuition for understanding it. I eventually found some ham radio guys who were great mentors and I learned a lot of good basic electronics from them.

NAN: Is there a particular electronics engineer, programmer, or designer who has inspired the work you do today?

DINO: Forrest Mims was a great inspiration in my early 20s. I got a big boost from his “Engineer’s Notebooks.” The simple way he explained things and his use of graph paper to draw circuit designs really made learning about electronics easy and fun. I still use graph paper to draw my schematics during the design phase and for planning when building a prototype on perf board. I’m not interested in any of the software schematic programs because most of my projects are simple and easy to draw. I like my pencil-and-paper approach.

NAN: What was the last electronics-design related product you purchased and what type of project did you use it with?

DINO: An Arduino Uno. I used two of these in the Photon robot.

NAN: What new technologies excite you and why?

DINO: Organic light-emitting diodes (OLEDs). They’ll totally change the way we manufacture and use digital displays.

I envision a day when you can go buy your big-screen TV that you’ll bring home in a cardboard tube, unroll it, and place it on the wall. The processor and power supply will reside on the floor, out of the way, and a single cable will go to the panel. The power consumption will be a fraction of today’s LCD or plasma displays and they’ll be featherweight by comparison. They’ll be used to display advertising on curved surfaces anywhere you like. Cell phone displays will be curved and flexible.

How about a panoramic set of virtual reality goggles or a curved display in a flight simulator? Once the technology gets out of the “early adopter” phase, prices will come down and you’ll own that huge TV for a fraction of what you pay now. One day we might even go to a movie and view it on a super-huge OLED panorama screen.

NAN: Final question. If you had a full year and a good budget to work on any design project you wanted, what would you build?

DINO: There’s a project I’ve wanted to build for some time now: A flight simulator based on the one used in Google Earth. I would use a PC to run the simulator and build a full-on seat-inside enclosure with all the controls you would have in a jet airplane. There are a lot of keyboard shortcuts for a Google flight simulator that could be triggered by switches connected to various controls (e.g., rudder pedals, flaps, landing gear, trim tabs, throttle, etc.). I would use the Arduino Leonardo as the controller for the peripheral switches because it can emulate a USB keyboard. Just program it, plug it into a USB port along with a joystick, build a multi-panel display (or use that OLED display I dream of), and go fly!

Google Earth’s flight simulator also lets you fly over the surface of Mars! Not only would this be fun to build and fly, it would also be a great educational tool. It’s definitely on the Hack A Week project list!

Editor’s Note: This article also appears in the Circuit Cellar’s upcoming March issue, which focuses on robotics. The March issue will soon be available for membership download or single-issue purchase.

 

Multi-Zone Home Audio System

Dave Erickson built his first multi-zone audio system in the early 1990s using C microprocessor code he developed on Freescale MC68HC11 microprocessors. The system has been an important part of his home.

“I used this system for more than 15 years and was satisfied with its ability to send different sounds to the different rooms in my house as well as the basement and the deck,” he says. “But the system needed an upgrade.”

In Circuit Cellar’s January and February issues, Erickson describes how he upgraded the eight-zone system, which uses microprocessor-controlled analog circuitry. In the end, his project not only improved his home audio experience, it also won second place in a 2011 STMicroelectronics design contest.

Several system components needed updating, including the IR remote, graphic LCD, and microprocessor. “IR remotes went obsolete, so the IR codes needed to change,” Erickson says. “The system was 90% hand-wired and pretty messy. The LCD and several other parts became obsolete and the C development tools had expired. Processors had evolved to include flash memory and development tools evolved beyond the old burn-and-pray method.”

“My goal was to build a modern, smaller, cleaner, and more efficient system,” he says. “I decided to upgrade it with a recent processor and LCD and to use real PC boards.”

Photo 1: Clockwise from the upper left, the whole-house system includes the crosspoint board, two quad preamplifiers, two two-zone stereo amplifiers, an AC transformer, power supplies, and the CPU board with the STMicroelectronics STM32VLDISCOVERY board.

Photo 1: Clockwise from the upper left, the whole-house system includes the crosspoint board, two quad preamplifiers, two two-zone stereo amplifiers, an AC transformer, power supplies, and the CPU board with the STMicroelectronics STM32VLDISCOVERY board.

Erickson chose the STMicroelectronics STM32F100 microprocessor and the work incentive of a design contest deadline (see Photo 1).

“STMicroelectronics’s excellent libraries and examples helped me get the complex ARM Cortex-M3 peripherals working quickly,” he says. “Choosing the STM32F100 processor was a bit of overkill, but I hoped to later use it to add future capabilities (e.g., a web page and Ethernet control) and possibly even a simple music server and audio streaming.”

In Part 1 of the series, Erickson explains the design’s audio sections, including the crosspoint board, quad preamplifiers, modular audio amplifiers, and packaging. He also addresses challenges along the way.

Erickson’s Part 1 provides the following overview of the system, including its “analog heart”—the crosspoint board:

Figure 1 shows the system design including the power supplies, front-panel controls, and the audio and CPU boards. The system is modular, so there is flexibility in the front-panel controls and the number of channels and amplifiers. My goal was to fit it all into one 19”, 2U (3.5”) high rack enclosure.

The CPU board is based on a STM32F100 module containing a Cortex-M3-based processor and a USB programming interface. The CPU receives commands from a front-panel keypad, an IR remote control, an encoder knob, RS-232, and external keypads for each zone. It displays its status on a graphic LCD and controls the audio circuitry on the crosspoint and two quad preamplifier boards.

The system block diagram shows the boards, controls, amplifiers, and power supplies.

The system block diagram shows the boards, controls, amplifiers, and power supplies.


Photo 2 shows the crosspoint board, which is the analog heart of the system. It receives line-level audio signals from up to eight stereo sources via RCA jacks and routes audio to the eight preamplifier channels located on two quad preamplifier boards. It also distributes digital control and power to the preamplifiers. The preamplifier boards can either send line-level outputs or drive stereo amplifiers, either internal or external to the system.

My current system uses four line-level outputs to drive PCs or powered speakers in four of the zones. It also contains internal 40-W stereo amplifiers to directly drive speakers in the four other zones. Up to six stereo amplifiers can reside in the enclosure.

Photo 2: The crosspoint board shows the RCA input jacks (top), ribbon cable connections to the quad preamplifiers (right), and control and power cable from the CPU (bottom). Rev0 has a few black wires (lower center).

Photo 2: The crosspoint board shows the RCA input jacks (top), ribbon cable connections to the quad preamplifiers (right), and control and power cable from the CPU (bottom). Rev0 has a few black wires (lower center).

DIYers dealing with signal leakage issues in their projects may learn something from Erickson’s approach to achieving low channel-to-channel crosstalk and no audible digital crosstalk. “The low crosstalk requirement is to prevent loud music in one zone from disturbing quiet passages in another,” he says.

In Part 1, Erickson explains the crosspoint and his “grounding/guarding” approach to transmitting high-quality audio, power, and logic control signals on the same cable:

The crosspoint receives digital control from the CPU board, receives external audio signals, and distributes audio signals to the preamplifier boards and then on to the amplifiers. It was convenient to use this board to distribute the control signals and the power supply voltages to the preamplifier channels. I used 0.1” dual-row ribbon cables to simplify the wiring. These are low-cost and easy to build.

To transmit high-quality audio along with power and logic control signals on the same cable, it is important to use a lot of grounds. Two 34-pin cables each connect to a quad preamplifier board. In each of these cables, four channels of stereo audio are sent with alternating signals and grounds. The alternating grounds act as electric field “guards” to reduce crosstalk. There are just two active logic signals: I2C clock and data. Power supply voltages (±12 and 5 V) are also sent to the preamplifiers with multiple grounds to carry the return currents.

I used a similar grounding/guarding approach throughout the design to minimize crosstalk, both from channel to channel and from digital to analog. On the two-layer boards, I used ground planes on the bottom layer. Grounded guard traces or ground planes are used on the top layer. These measures minimize the capacitance between analog traces and thus minimize crosstalk. The digital and I2C signals are physically separated from analog signals. Where they need to be run nearby, they are separated by ground planes or guard traces.

To find out more about how Erickson upgraded his audio system, download the January issue (now available online) and the upcoming February issue. In Part 2, Erickson focuses on his improved system’s digital CPU, the controls, and future plans.

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.)

CLASS A
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%…

CLASS B
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.

CLASS AB
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…

CLASS C
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
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.

CLASS E and F
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
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.

2.4-GHz RF High-Power Amplifier

Microchip

The SST12CP12 high-power amplifier

The SST12CP12 is a 2.4-GHz RF high-power amplifier that adds support for 256-QAM ultra-high data rate modulation. With its high linear output power, this amplifier significantly extends the range of IEEE 802.11b/g/n WLAN systems while providing excellent power at the maximum 256-QAM data rate. The amplifier is also spectrum-mask compliant up to 28.5 dBm for 802.11b/g communication and utilizes orthogonal frequency-division multiplexing (OFDM) to correct severe channel conditions without using complex equalization filters.

The SST12CP12 power amplifier has a 380mA at 23 dBm low operating current, which enables more transmission channels and a higher data rate for each system. The amplifier also features easy to use 50-Ω on-chip input match and simple output match. In addition, the integrated linear power detector provides temperature stability and immunity to voltage standing wave ratio (VSWR) radio-wave reflection to provide accurate output power control.

The SST12CP12 costs $0.97 each, in 10,000-unit quantities. It ships in a 3-mm × 3-mm × 0.55-mm, 16-pin QFN package.

Microchip Technology, Inc.
www.microchip.com

A Look at Low-Noise Amplifiers

Maurizio Di Paolo Emilio, who has a PhD in Physics, is an Italian telecommunications engineer who works mainly as a software developer with a focus on data acquisition systems. Emilio has authored articles about electronic designs, data acquisition systems, power supplies, and photovoltaic systems. In this article, he provides an overview of what is generally available in low-noise amplifiers (LNAs) and some of the applications.

By Maurizio Di Paolo Emilio
An LNA, or preamplifier, is an electronic amplifier used to amplify sometimes very weak signals. To minimize signal power loss, it is usually located close to the signal source (antenna or sensor). An LNA is ideal for many applications including low-temperature measurements, optical detection, and audio engineering. This article presents LNA systems and ICs.

Signal amplifiers are electronic devices that can amplify a relatively small signal from a sensor (e.g., temperature sensors and magnetic-field sensors). The parameters that describe an amplifier’s quality are:

  • Gain: The ratio between output and input power or amplitude, usually measured in decibels
  • Bandwidth: The range of frequencies in which the amplifier works correctly
  • Noise: The noise level introduced in the amplification process
  • Slew rate: The maximum rate of voltage change per unit of time
  • Overshoot: The tendency of the output to swing beyond its final value before settling down

Feedback amplifiers combine the output and input so a negative feedback opposes the original signal (see Figure 1). Feedback in amplifiers provides better performance. In particular, it increases amplification stability, reduces distortion, and increases the amplifier’s bandwidth.

 Figure 1: A feedback amplifier model is shown here.


Figure 1: A feedback amplifier model is shown.

A preamplifier amplifies an analog signal, generally in the stage that precedes a higher-power amplifier.

IC LOW-NOISE PREAMPLIFIERS
Op-amps are widely used as AC amplifiers. Linear Technology’s LT1028 or LT1128 and Analog Devices’s ADA4898 or AD8597 are especially suitable ultra-low-noise amplifiers. The LT1128 is an ultra-low-noise, high-speed op-amp. Its main characteristics are:

  • Noise voltage: 0.85 nV/√Hz at 1 kHz
  • Bandwidth: 13 MHz
  • Slew rate: 5 V/µs
  • Offset voltage: 40 µV

Both the Linear Technology and Analog Devices amplifiers have voltage noise density at 1 kHz at around 1 nV/√Hz  and also offer excellent DC precision. Texas Instruments (TI)  offers some very low-noise amplifiers. They include the OPA211, which has 1.1 nV/√Hz  noise density at a  3.6 mA from 5 V supply current and the LME49990, which has very low distortion. Maxim Integrated offers the MAX9632 with noise below 1nV/√Hz.

The op-amp can be realized with a bipolar junction transistor (BJT), as in the case of the LT1128, or a MOSFET, which works at higher frequencies and with a higher input impedance and a lower energy consumption. The differential structure is used in applications where it is necessary to eliminate the undesired common components to the two inputs. Because of this, low-frequency and DC common-mode signals (e.g., thermal drift) are eliminated at the output. A differential gain can be defined as (Ad = A2 – A1) and a common-mode gain can be defined as (Ac = A1 + A2 = 2).

An important parameter is the common-mode rejection ratio (CMRR), which is the ratio of common-mode gain to the differential-mode gain. This parameter is used to measure the  differential amplifier’s performance.

Figure 2: The design of a simple preamplifier is shown. Its main components are the Linear Technology LT112 and the Interfet IF3602 junction field-effect transistor (JFET).

Figure 2: The design of a simple preamplifier is shown. Its main components are the Linear Technology LT1128 and the Interfet IF3602 junction field-effect transistor (JFET).

Figure 2 shows a simple preamplifier’s design with 0.8 nV/√Hz at 1 kHz background noise. Its main components are the LT1128 and the Interfet IF3602 junction field-effect transistor (JFET).  The IF3602 is a dual N-channel JFET used as stage for the op-amp’s input. Figure 3 shows the gain and Figure 4 shows the noise response.

Figure 3: The gain of a low-noise preamplifier.

Figure 3: The is a low-noise preamplifier’s gain.

 

Figure 4: The noise response of a low-noise preamplifier

Figure 4: A low-noise preamplifier’s noise response is shown.

LOW NOISE PREAMPLIFIER SYSTEMS
The Stanford Research Systems SR560 low-noise voltage preamplifier has a differential front end with 4nV/√Hz input noise and a 100-MΩ input impedance (see Photo 1a). Input offset nulling is accomplished by a front-panel potentiometer, which is accessible with a small screwdriver. In addition to the signal inputs, a rear-panel TTL blanking input enables you to quickly turn the instrument’s gain on and off (see Photo 1b).

Photo 1a:The Stanford Research Systems SR560 low-noise voltage preamplifier

Photo 1a: The Stanford Research Systems SR560 low-noise voltage preamplifier. (Photo courtesy of Stanford Research Systems)

Photo 1 b: A rear-panel TTL blanking input enables you to quickly turn the Stanford Research Systems SR560 gain on and off.

Photo 1b: A rear-panel TTL blanking input enables you to quickly turn the Stanford Research Systems SR560 gain on and off. (Photo courtesy of Stanford Research Systems)

The Picotest J2180A low-noise preamplifier provides a fixed 20-dB gain while converting a 1-MΩ input impedance to a 50-Ω output impedance and 0.1-Hz to 100-MHz bandwidth (see Photo 2). The preamplifier is used to improve the sensitivity of oscilloscopes, network analyzers, and spectrum analyzers while reducing the effective noise floor and spurious response.

Photo 2: The Picotest J2180A low-noise preamplifier is shown.

Photo 2: The Picotest J2180A low-noise preamplifier is shown. (Photo courtesy of picotest.com)

Signal Recovery’s Model 5113 is among the best low-noise preamplifier systems. Its principal characteristics are:

  • Single-ended or differential input modes
  • DC to 1-MHz frequency response
  • Optional low-pass, band-pass, or high-pass signal channel filtering
  • Sleep mode to eliminate digital noise
  • Optically isolated RS-232 control interface
  • Battery or line power

The 5113 (see Photo 3 and Figure 5) is used in applications as diverse as radio astronomy, audiometry, test and measurement, process control, and general-purpose signal amplification. It’s also ideally suited to work with a range of lock-in amplifiers.

Photo 3: This is the Signal Recovery Model 5113 low-noise pre-amplifier.

Photo 3: This is the Signal Recovery Model 5113 low-noise preamplifier. (Photo courtesy of Signal Recovery)

Figure 5: Noise contour figures are shown for the Signal Recovery Model 5113.

Figure 5: Noise contour figures are shown for the Signal Recovery Model 5113.

WRAPPING UP
This article briefly introduced low-noise amplifiers, in particular IC system designs utilized in simple or more complex systems such as the Signal Recovery Model 5113, which is a classic amplifier able to obtain different frequency bands with relative gain. A similar device is the SR560, which is a high-performance, low-noise preamplifier that is ideal for a wide variety of applications including low-temperature measurements, optical detection, and audio engineering.

Moreover, the Krohn-Hite custom Models 7000 and 7008 low-noise differential preamplifiers provide a high gain amplification to 1 MHz with an AC output derived from a very-low-noise FET instrumentation amplifier.

One common LNA amplifier is a satellite communications system. The ground station receiving antenna will connect to an LNA, which is needed because the received signal is weak. The received signal is usually a little above background noise. Satellites have limited power, so they use low-power transmitters.

Telecommunications engineer Maurizio Di Paolo Emilio was born in Pescara, Italy. Working mainly as a software developer with a focus on data acquisition systems, he helped design the thermal compensation system (TCS) for the optical system used in the Virgo Experiment (an experiment for detecting gravitational waves). Maurizio currently collaborates with researchers at the University of L’Aquila on X-ray technology. He also develops data acquisition hardware and software for industrial applications and manages technical training courses. To learn more about Maurizio and his expertise, read his essay on “The Future of Data Acquisition Technology.”

CC281: Overcome Fear of Ethernet on an FPGA

As its name suggests, the appeal of an FPGA is that it is fully programmable. Instead of writing software, you design hardware blocks to quickly do what’s required of a digital design. This also enables you to reprogram an FPGA product in the field to fix problems “on the fly.”

But what if “you” are an individual electronics DIYer rather than an industrial designer? DIYers can find FPGAs daunting.

Issue281The December issue of Circuit Cellar issue should offer reassurance, at least on the topic of “UDP Streaming on an FPGA.” That’s the focus of Steffen Mauch’s article for our Programmable Logic issue (p. 20).

Ethernet on an FPGA has several applications. For example, it can be used to stream measured signals to a computer for analysis or to connect a camera (via Camera Link) to an FPGA to transmit images to a computer.

Nonetheless, Mauch says, “most novices who start to develop FPGA solutions are afraid to use Ethernet or DDR-SDRAM on their boards because they fear the resulting complexity.” Also, DIYers don’t have the necessary IP core licenses, which are costly and often carry restrictions.

Mauch’s UDP monitor project avoids such costs and restrictions by using a free implementation of an Ethernet-streaming device based on a Xilinx Spartan-6 LX FPGA. His article explains how to use OpenCores’s open-source tri-mode MAC implementation and stream UDP packets with VHDL over Ethernet.

Mauch is not the only writer offering insights into FPGAs. For more advanced FPGA enthusiasts, columnist Colin O’Flynn discusses hardware co-simulation (HCS), which enables the software simulation of a design to be offloaded to an FPGA. This approach significantly shortens the time needed for adequate simulation of a new product and ensures that a design is actually working in hardware (p. 52).

This Circuit Cellar issue offers a number of interesting topics in addition to programmable logic. For example, you’ll find a comprehensive overview of the latest in memory technologies, advice on choosing a flash file system for your embedded Linux system, a comparison of amplifier classes, and much more.

Mary Wilson
editor@circuitcellar.com

SRPP Headphone Amp (EE Tip #106)

Mention tube amplifiers and many designers go depressive instantly over the thought of a suitable output transformer. The part will be in the history books forever as esoteric, bulky and expensive because, it says, it is designed and manufactured for a specific tube constellation and output power. There exist thick books on tube output transformers, as well as gurus lecturing on them and winding them by hand. However, with some concessions to distortion (but keeping a lot of money in your pocket) a circuit configuration known as series regulated push-pull (SRPP) allows a low-power tube amplifier to be built that does not require the infamous output transformer. SRPP is normally used for pre-amplifier stages only, employing two triodes in what looks like a cascade arrangement.

Here we propose the use of two EL84 (6BQ5) power pentodes in triode SRPP configuration. The reasons for using the EL84 (6CA5) are mainly that it’s cheap, widely available ,and forgiving of the odd overload condition. Here, two of these tubes are SRPPed into an amplifier that’s sure to reproduce that ‘warm thermionic sound’ so much in demand these days.

Martin Louw Kristoffersen, Elektor, 081151-I, 7-8/2009

Martin Louw Kristoffersen, Elektor, 081151-I, 7-8/2009

Before describing the circuit operation, it must be mentioned that construction of this circuit must not be attempted unless you have experience in working with tubes at high voltages, or can rely on the advice and assistance of an “old hand.” As a safety measure, two anti-series connected Zener diodes are fitted at the amplifier output. These devices protect the output (i.e., your headphones and ears) against possibly dangerous voltages at switch-on, or when output capacitor C3 breaks down.

The power supply is sized for two channels (i.e., a stereo version of the amplifier). The values in brackets are for Elektor readers on 120-V AC networks. Note the doubled values of fuses F1 and F3 in the AC primary circuits. The PSU is a conventional design, possibly with the exception of the 6.3-V heater voltage being raised to a level of about +80 V through voltage divider R7-R8. This is done to prevent exceeding the maximum cathode heater voltage specified for the EL84 (6CA5). R6 is a bleeder resistor emptying the reservoir capacitors C8 and C9 in a quick but controlled manner when the amplifier is switched off. Rectifier diodes D3–D6 each have an anti-rattle capacitor across them.

In the amplifier, assuming the tubes used have roughly the same emission, the half-voltage level of about +145 V exists at the junction of the anode of V1 and the control grid of V2. The SRPP is no exception to the rule that high quality, (preferably) new capacitors are essential not just for reproduction and sound fidelity, but also for safety.

—Martin Louw Kristoffersen, Elektor, 081151-I, 7-8/2009

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.

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 www.justradios.com). 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).

CircuitCellar.com and audioXpress are Elektor International Media publications.   

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.