Q&A with Arduino-Based Skube Codesigner

The Arduino-based Skube

The Arduino-based Skube

Andrew Spitz is a Copenhagen, Denmark-based sound designer, interaction designer, and programmer. 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.

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

Skube – A Last.fm & Spotify Radio from Andrew Nip on Vimeo.

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.

New DSP “Lab-in-a-Box” for ARM-Based Audio Systems

Cambridge, UK-based, ARM and its partners will start shipping a DSP “Lab-in-a-Box” (LiB) to universities worldwide to help boost practical skills development and the creation of new ARM-based audio systems. This will include products such as high-definition home media and voice-controlled home automation systems. The LiB kits contain ARM Cortex-M4-based microcontroller boards by STMicroelectronics and audio cards from Wolfson Microelectronics and Farnell element14.ARMDSPLiBWeb

As the centerpiece of the ARM University Program, LiB packages offer ARM-based technology and high-quality teaching and training materials that support electronics and computer engineering courses. DSP courses have traditionally used software simulation packages, or hands-on labs using relatively expensive development kits costing around $300 per student. By comparison, this new DSP LiB will cost around $50 and will allow students to practice theory with advanced hardware sourced from widely-available products.

“Our Lab-in-a-Box offerings are proving hugely popular in universities because of the low-cost access to state-of-the-art technology,” said Khaled Benkrid, manager of the Worldwide University Program, ARM. “The DSP kits, powered by ARM Cortex-M4-based processors, enable high performance yet energy-efficient digital signal processing at a very affordable price. We expect to see them being used by students to create commercially-viable audio applications and it’s another great example of our partnership supporting engineers in training and beyond.”

The DSP LiB will begin shipping to universities in July 2014. It is the latest in a series of initiatives led by ARM which span multiple academic topics including embedded systems design, programming and SoC design. The DSP kits will also be offered to developers outside academia at a later date.

[via audioXpress.com]

High Dynamic-Range Audio Processor

STMicroelectronics recently introduced a new digital audio processor with greater than 100-dB SNR and Dynamic Range. The device can process most digital input formats including 6.1/7.1 channel and 192-kHz, 24-bit DVD audio and DSD/SACD. When configured in a 5.1 application, its additional two channels can be used to supply audio line-out or headphone drive.

Source: STMicroelectronics

Source: STMicroelectronics

The STA311B is a single chip solution for digital audio processing and control in multichannel applications, providing FFXTM (Full Flexible Amplification) compatible outputs. Together with a FFXTM power amplifier it can provide high-quality, high-efficiency, all-digital amplification.

The chip accepts digitized audio input information in either I2S (left or right justified), LSB or MSB first, with word lengths of 16, 18, 20 and 24 bits. Its pop-noise removal feature does not discriminate against the music genre but instead prevents any audible transients or pops finding their way through to the power amp where they may damage the speakers. Device control is via an I2C interface. The STA311B embeds eight audio-processing channels with up to 10 independent user-selectable bi-quadratic filters per channel to allow easy implementation of tone and music genre equalization templates. It is capable of input and output mixing with multi-band dynamic range compression. The chip also has input sampling frequency auto-detection, input/output RMS metering and employs pulse-width modulated output channels.

The STA311B is supplied in an 8.0 × 8.0 × 0.9 mm VFQFPN package.

[via Elektor]

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.