Twin-T Oscillator Configuration

Since retiring in 2013, electrical engineer Larry Cicchinelli has provided technical support at an educational radio station. For audio circuit debugging and testing, he uses a DIY battery-powered oscillator/volume unit (VU) meter. Details follow.

Originally, I was only going to build the audio source. When I thought about how I would use the unit, it occurred to me that the device should have a display. I decided to design and build an easy-to-use unit that would combine a calibrated audio source with a level display. Then, I would have a single, battery-powered instrument to do some significant audio circuit testing and debugging.

The front panel of the oscillator/volume unit (VU) meter contains all the necessary controls. (Source: L. Cicchinelli)

The front panel of the oscillator/volume unit (VU) meter contains all the necessary controls. (Source: L. Cicchinelli)

Cicchinelli describes the Twin-T Oscillator:

The oscillator uses the well-known Twin-T configuration with a minor modification to ensure a constant level over a range of power supply voltages. The circuit I implemented maintains its output level over a range of at least 6 to 15 V. Below 6 V, the output begins to distort if you have full output voltage (0 dBu). The modification consists of two antiparallel diodes in the feedback loop. The idea came from a project on DiscoverCircuits.com. The project designer also indicates that the diodes reduce distortion.

Figure 1 shows the oscillator’s schematic. Header H1 and diode D1 enable you to have two power sources. I installed a 9-V battery and snap connector in the enclosure as well as a connector for external power. The diode enables the external source to power the unit if its voltage is greater than the battery. Otherwise the battery will power the unit. The oscillator draws about 4 mA so it does not create a large battery drain.

The standard professional line level is 4 dBu, which is 1.228 VRMS or 3.473 VPP into a 600-Ω load. The circuit values enable you to use R18 to calibrate it, so the maximum output can be set to the 4-dBu level. A 7.7 (3.473/0.45) gain is required to provide 4 dBu at the transformer. Using the resistors shown in Figure 1, R18 varies the gain of U1.2 from about 4.3 to 13.

The Twin-T oscillator’s circuitry

Figure 1: The Twin-T oscillator’s circuitry

You may need to use different resistor values for R18, R19, and R20 to achieve a different maximum level. If you prefer to use 0 dBm (0.775 VRMS into 600 Ω) instead of 4 dBu, you should change R20 to about 5 kΩ to give R18 a range more closely centered on a 4.87 (2.19/0.45) gain. The R20’s value shown in Figure 1 will probably work, but the required gain is too close to the minimum necessary for comfort. Most schematics for a Twin-T oscillator will show the combination of R3 and R4 as a single resistor of value Rx/2. They will also show the combination of C1 and C2 as a single capacitor of value Cx × 2. These values lead to the following formula:

CicchinelliEQ1

As you can see in the nearby photo, the Twin-T Oscillator and VU meter contain separate circuit boards.

The Twin-T oscillator and dual VU meter have separate circuit boards

The Twin-T oscillator and dual VU meter have separate circuit boards

This article first appeared in audioXpress January 2014. audioXpress is one of Circuit Cellar‘s sister publications.

 

Arduino-Based Tube Stereo Preamp Project

If you happen to be electrical engineer as well as an audiophile, you’re in luck. With an Arduino, some typical components, and a little knowhow, you can build DIY tube stereo preamplifier design.

Shannon Parks—owner of Mahomet, IL-based Parks Audio—designed his “Budgie” preamp after reading an article about Arduino while he was thinking about refurbishing a classic Dynaco PAS-3.

Budgie preamp (Source: S. Parks)

Budgie preamp (Source: S. Parks)

In a recent audioXpress article about the project, Parks noted:

Over the last 10 years, I have built many tube power amplifiers but I had never built a tube preamplifier. The source switching seemed particularly daunting. A friend recommended that I refurbish a classic Dynaco PAS-3 which has been a popular choice with many upgrade kit suppliers. Unfortunately, the main part of these older designs is a clumsy rotary selector switch, not to mention the noisy potentiometers and slide switches. In the 1980s, commercial stereo preamplifiers started using IC microcontrollers that permitted cleaner designs with push-button control, relays for signal switching, and a wireless remote. While reading an article about the Arduino last year, I realized these modern features could easily be incorporated into a DIY preamplifier design.

All the circuits are on one custom PCB along with the power supply and microcontroller (Source: S. Parks)

All the circuits are on one custom PCB along with the power supply and microcontroller (Source: S. Parks)

Parks said the Arduino made sense for a few key reasons:

I found these features were incredibly useful:

  • A bank of relays could switch between the four stereo inputs as well as control mute, standby, gain, and bass boost settings.
  • A red power LED could use PWM to indicate if the preamplifier is muted or in standby.
  • An IR receiver with a remote could control a motor-driven volume potentiometer, change the source input selection, and turn the unit on/off. Any IR remote could be used with a code learning mode.
  • A backlit display could easily show all the settings at a glance.
  • Momentary push buttons could select the input device, bass boost, gain, and mute settings.
  • Instead of using several Arduino shields wired to an Arduino board, all the circuits could fit on one custom PCB along with the power supply and the microcontroller.

Parks used an Arduino Nano, which 0.73” × 1.70”. “The tiny Nano can be embedded using a 32-pin dual in-line package (DIP) socket, which cleans up the design. It can be programmed in-circuit and be removed and easily replaced,” he noted.

Parks used an Arduino Nano for the preamp project (Source: S. Parks)

Parks used an Arduino Nano for the preamp project (Source: S. Parks)

Parks described the shift register circuit:

The Budgie preamplifier uses a serial-in, parallel-out (SIPO) shift register to drive a bank of relays ….

A SIPO shift register is used to drive a bank of relays (Source: S. Parks)

A SIPO shift register is used to drive a bank of relays (Source: S. Parks)

Only four Arduino digital outputs—enable, clock, latch, and data—are needed to control eight DPDT relays. These correspond to the four outputs labeled D3, D4, D5, and D7 s …. The Texas Instruments TPIC6C595 shift register used in this project has heavy-duty field-effect transistor (FET) outputs that can handle voltages higher than logic levels. This is necessary for operating the 24-V relays. It also acts as a protective buffer between the Arduino and the relays.

Here you see the how to set up the Arduino Nano, LCD, power supply, push button , IR and motor control circuits (Source: S. Parks)

Here you see the how to set up the Arduino Nano, LCD, power supply, push button , IR and motor control circuits (Source: S. Parks)

As for the audio circuit, Parks explained:

The 12B4 triode was originally designed to be used in televisions as a vertical deflection amplifier. New-old-stock (NOS) 12B4s still exist. They can be purchased from most US tube resellers. However, a European equivalent doesn’t exist. The 12B4 works well in preamplifiers as a one-tube solution, having both high input impedance and low output impedance, without need for an output transformer. An audio circuit can then be distilled down to a simple circuit with few parts consisting of a volume potentiometer and a grounded cathode gain stage.
The 12B4 has about 23-dB gain, which is more than is needed. This extra gain is used as feedback to the grid, in what is often referred to as an anode follower circuit. The noise, distortion, and output impedance are reduced (see Figure 3). Using relays controlled by the Arduino enables switching between two feedback amounts for adjustable gain. For this preamplifier, I chose 0- and 6-dB overall gain. A second relay enables a bass boost with a series capacitor.
You only need a lightweight 15-to-20-V plate voltage to operate the 12B4s at 5 mA. Linearity is very good due to the small signal levels involved, as rarely will the output be greater than 2 VPP. A constant current source (CCS) active load is used with the 12B4s instead of a traditional plate resistor. This maximizes the possible output voltage swing before clipping. For example, a 12B4 biased at 5-mA plate current with a 20-kΩ plate resistor would drop 100 V and would then require a 120-V supply voltage or higher. Conversely, the CCS will only drop about 2 V. Its naturally high impedance also improves the tube’s gain and linearity while providing high levels of power supply noise rejection.

This article first appeared in Circuit Cellar’s sister publication, audioXpress (July 2014).

 

 

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]