The serendipitous invention of the wah-wah pedal

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The first wah-wah pedal is attributed to Brad Plunkett in 1966, who worked at Warwick Electronics Inc., which owned Thomas Organ Company. Warwick Electronics acquired the Vox name due to the brand name’s popularity and association with the Beatles. Their subsidiary, Thomas Organ Company, needed a modified design for the Vox amplifier, which had a midrange boost, so that it would be less expensive to manufacture.

In a 2005 interview (M. Vdovin, “Artist Interview: Brad Plunkett,” Universal Audio WebZine, vol. 3, October 2005) Brad Plunkett said, I “came up with a circuit that would allow me to move this midrange boost … As it turned out, it sounded absolutely marvelous while you were moving it. It was okay when it was standing still, but the real effect was when you were moving it and getting a continuous change in harmonic content. We turned that on in the lab and played the guitar through it… I turned the potentiometer and he played a couple licks on the guitar, and we went crazy.

A couple of years later… somebody said to me one time, ‘You know Brad, I think that thing you invented changed music.’”

Acoustic reverberators

Today, reverb is most often added to a recording using artificial reverberators, such as software plug-ins or digital reverb hardware units. But there are a lot of other approaches.

Many recording studios have used special rooms known as reverberation chambers to add reverb to a performance. Elevator shafts and stairwells (as in New York City’s Avatar Recording Studio) work well as highly reverberant rooms. The reverb can also be controlled by adding absorptive materials like curtains and rugs.

Spring reverbs are found in many guitar amplifiers and have been used in Hammond organs. The audio signal is coupled to one end of the spring by a transducer that creates waves traveling through the spring. At the far end of the spring, another transducer converts the motion of the string into an electrical signal, which is then added to the original sound. When a wave arrives at an end of the spring, part of the wave’s energy is reflected. However, these reflections have different delays and attenuations from what would be found in a natural acoustic environment, and there may be some interaction between the waves in a spring, thus this results in a slightly unusual (though not unpleasant) reverb sound.

Often several springs with different lengths and tensions are enclosed in a metal box, known as the reverb pan, and used together. This avoids uniform behavior and creates a more realistic, pseudorandom series of echoes. In most reverb units though, the spring lengths and tensions are fixed in the design process, and not left to the user to control.

The plate reverb is similar to a spring reverb, but instead of springs, the  transducers are attached at several locations on a metal plate. These transducers send vibrations through the plate, and reflections are produced whenever a wave reaches the plate’s edge. The location of the transducers and the damping of the plate can be adjusted to control the reverb. However, plate reverbs are expensive and bulky, and hence not widely used.

Water tank reverberators have also been used. Here, the audio signal is modulated with an ultrasonic signal and transmitted through a tank of water. The output is then demodulated, resulting in the reverberant output sound. Other reverberators include pipes with microphones placed at various points.

These acoustic and analogue reverberators can be interesting to create and use, but they lack the simplicity and ease of use of digital reverberators. Ultimately, the choice of implementation is a matter of taste.

High resolution audio- finally, rigorously put to the test. And the verdict is…

Yes, you can hear a difference! (but it is really hard to measure)

See http://www.aes.org/e-lib/browse.cfm?elib=18296 http://www.aes.org/journal/ for the June 2016 article in the Journal of the Audio Engineering Society  on “A meta-analysis of high resolution audio perceptual evaluation”

For years, I’ve been hearing people in the audio engineering community arguing over whether or not it makes any difference to record, mix and playback better than CD quality (44.1 kHz, 16 bit) or better than production quality (48 kHz, 16 bit) audio. Some people swear they can hear a difference, others have stories about someone they met who could always pick out the differences, others say they’re all just fooling themselves. A few people could mention a study or two that supported their side, but the arguments didn’t seem to ever get resolved.

Then, a bit more than a year ago I was at a dinner party where a guy sitting across from me was about to complete his PhD in meta-analysis. Meta-analysis? I’d never heard of it. But the concept, analysing and synthesising the results of many studies to get a more definitive answer and gain more insights and knowledge, really intrigued me. So it was about time that someone tried this on the question of perception of hi-res audio.

Unfortunately, no one I asked was willing to get involved. A couple of experts thought there couldn’t be enough data out there to do the meta-analysis. A couple more thought that the type of studies (not your typical clinical trial with experimental and control groups) couldn’t be analysed using the established statistical approaches in meta-analysis. So, I had to do it myself. This also meant I had to be extra careful, and seek out as much advice as possible, since no one was looking over my shoulder to tell me when I was wrong or stupid.

The process was fascinating. The more I looked, the more I uncovered studies of high resolution audio perception. And my main approach for finding them (start with a few main papers, then look at everyone they cited and everyone who cited them, and repeat with any further interesting papers found), was not mentioned in the guidance to meta-analysis that I read. Then getting the data was interesting. Some researchers had it all prepared in handy, well-labelled spreadsheets, one other found it in an old filing cabinet, one had never kept it at all! And for some data, I had to write little programs to reverse engineer the raw data from T values for trials with finite outcomes.

Formal meta-analysis techniques could be applied, and I gained a strong appreciation for both the maths behind them, and the general guidance that helps ensure rigour and helps avoid bias in the meta-study, But the results, in a few places, disagreed with what is typical. The potential biases in the studies seemed to occur more often with those that did not reject the null hypothesis, i.e., those that found no evidence for discriminating between high resolution and CD quality audio. Evidence of publication bias seemed to mostly go away if one put the studies into subgroups. And use of binomial probabilities allowed the statistical approaches in meta-analysis to be applied to studies where there was not a control group (‘no effect’ can be determined just from binomial probabilities).

The end result was that people could, sometimes, perceive the difference between hi-res and CD audio. But they needed to be trained and the test needed to be carefully designed. And it was nice to see that the experiments and analysis were generally a little better today than in the past, so research is advancing. Still, most tests had some biases towards false negatives. So perhaps, careful experiments, incorporating all the best approaches, may show this perception even more strongly.

Meta-analysis is truly fascinating, and audio engineering, psychoacoustics, music technology and related fields need more of it.

The Swoosh of the Sword

When we watch Game of Thrones or play the latest Assassin’s Creed the sound effect added to a sword being swung adds realism, drama and overall excitement to our viewing experience.

There are a number of methods for producing sword sound effects, from filtering white noise with a bandpass filter to solving the fundamental equations for fluid dynamics using finite volume methods. One method investigated by the Audio Engineering research team at QMUL was to find semi-empirical equations used in the Aeroacoustic community as an alternative to solving the full Navier Stokes equations. Running in real-time these provide computationally efficient methods of achieving accurate results – we can model any sword, swung at any speed and even adjust the model to replicate the sound of a baseball bat or golf club!

The starting point for these sound effect models is that of the Aeolian tone, (see previous blog entry – https://intelligentsoundengineering.wordpress.com/2016/05/19/real-time-synthesis-of-an-aeolian-tone/). The Aeolian tone is the sound generated as air flows around an object, in the case of our model, a cylinder. In the previous blog we describe the creation of a sound synthesis model for the Aeolian tone, including a link to a demo version of the model.

For a sword we take a number of the Aeolian tone models and place them on a virtual sword at different place settings. This is shown below:

coordSwordSource

Each Aeolian tone model is called a compact source. It can be seen that more are placed at the tip of the sword rather than the hilt. This is because the acoustic intensity is far higher for faster moving sources. There are 6 sources placed at the tip, positioned at a distance of 7 x the sword diameter. This distance is based on when the aerodynamic effects become de-correlated, although a simplification. One source is placed at the hilt and the final source equidistant between the last tip source and the hilt.

The complete model is presented in a GUI as shown below:

SwordDemoGUI

Referring to the both previous figures, it can be seen that the user is able to move the observer position within a 3D space. The thickness of the blade can be set at the tip and the hilt as well as the length of the blade. It is then linearly interpolated over the blade length so that each source diameter can be calculated.

The azimuth and elevation of the sword pre and post swing can be set. The strike position is fixed to an azimuth of 180 degrees and this is the point where the sword reaches its maximum speed. The user sets the top speed of the tip from the GUI. The Prime button makes sure all the variables are pushed through into the correct places in equations and the Go button triggers the swing.

It can be seen that there are 4 presets. Model 1 is a thin fencing type sword and Model 2 is a thicker sword. To test versatility of the model we decided to try and model a golf club. The preset PGA will set the model to implement this. The golf club model involves making the diameter of the source at the tip much larger, to represent the striking face of a golf club. It was found that those unfamiliar with golf did not identify the sound immediately so a simple golf ball strike sound is synthesised as the club reaches top speed.

To test versatility further, we created a model to replicate the sound of a baseball bat; preset MLB. This is exactly the same model as the sword with the dimensions just adjusted to the length of a bat plus the tip and hilt thickness. A video with all the preset sounds is given below. This includes two sounds created by a model with reduced physics, LoQ1 & LoQ2. These were created to investigate if there is any difference in perception.

The demo model was connected to the animation of a knight character in the Unity game engine. The speed of the sword is directly mapped from the animation to the sound effect model and the model observer position set to the camera position. A video of the result is given below:

Doppler, Leslie and Hammond

Donald Leslie (1913–2004) bought a Hammond organ in 1937, as a substitute for a pipe organ. But at home in a small room, it could not reproduce the grand sound of an organ. Since the pipe organ has different locations for each pipe, he designed a moving loudspeaker.

The Leslie speaker uses an electric motor to move an acoustic horn in a circle around a loudspeaker. Thus we have a moving sound source and a stationary listener, which is a well-known situation that produces the Doppler effect.

It exploits the Doppler effect to produce frequency modulation. The classic Leslie speaker has a crossover that divides the low and high frequencies. It consists of a fixed treble unit with spinning horns, a fixed woofer and spinning rotor. Both the horns (actually, one horn and a dummy used as a counterbalance) and a bass sound baffle rotate, thus creating vibrato due to the changing velocity in the direction of the listener, and tremolo due to the changing distance. The rotating elements can move at varied speeds, or stopped completely. Furthermore, the system is partially enclosed and it uses a rotating speaker port. So the listener hears multiple reflections at different Doppler shifts to produce a chorus-like effect.

The Leslie speaker has been widely used in popular music, especially when the Hammond B-3 organ was played out through a Leslie speaker. This combination can be heard on many classic and progressive rock songs, including hits by Boston, Santana, Steppenwolf, Deep Purple and The Doors. And the Leslie speaker has also found extensive use in modifying guitar and vocal sounds.

Ironically, Donald Leslie had originally tried to license his loudspeaker to the Hammond company, and even gave the Hammond company a special demonstration. But at the time, Laurens Hammond (founder of the Hammond organ company) did not like the concept at all.

Blogs, blogs, blogs

We’re collaborating on a really interesting project called ‘Cross-adaptive processing as musical intervention,’ led by Professor Øyvind Brandtsegg of the Norwegian University of Science and Technology. Essentially, this project involves cross-adaptive audio effects, where the processing applied to one audio signal is dependent on analysis of other signals. We’ve used this concept quite a lot to build intelligent music production systems. But in this project, Øyvind and his collaborators are exploring creative uses of cross-adaptive audio effects in live performance. The effects applied to one source may change depending on what and how another performer plays, so a performer may change what they play to overtly influence everyone else’s sound, thus taking the interplay in a jam session to a whole new level.

One of the neat things that they’ve done to get this project off the ground is created a blog, http://crossadaptive.hf.ntnu.no/ , which is a great way to get all the reports and reflections out there quickly and widely.

This got me thinking of a few other blogs that we should mention. First and foremost is Prof, Trevor Cox of the University of Salford’s wonderful blog, ‘The Sound Blog: Dispatches from Acoustic and Audio Engineering,’ is available at https://acousticengineering.wordpress.com/ . This blog was one of the principal inspirations for our own blog here.

Another leading researcher’s interesting blog is https://marianajlopez.wordpress.com/ – Mariana is looking into aspects of sound design that I feel really don’t get enough attention from the academic community… yet. Hopefully, that will change soon.

There’s plenty of blogs about music production. A couple of good ones are http://thestereobus.com/ and http://productionadvice.co.uk/blog/ . They are full of practical advice, insights and tutorials.

A lot of the researchers in the Audio Engineering team have their own personal blogs, which discuss their research, their projects and various other things related to their career or just cool technologies.

See,

http://brechtdeman.com/blog.html – Brecht De Man ‘s blog. He’s researching semantic and knowledge engineering approaches to music production systems (and a lot more).

https://auralcharacter.wordpress.com/ – Alessia Milo’s blog. She’s looking at (and listening to) soundscapes, and their importance in architecture

http://davemoffat.com/wp/ – Dave Moffat is investigating evaluation of sound synthesis techniques, and how machine learning can be applied to synthesize a wide variety of sound effects.

https://rodselfridge.wordpress.com/ – Rod Selfridge is looking at real-time physical modelling techniques for procedural audio and sound synthesis.

More to come on all of them, I’m sure.

Let us know of any other blogs that we should mention, and we’ll update this entry or add new entries.

Real-Time Synthesis of an Aeolian tone

Aeroacoustics are sounds generated by objects and the air and is a unique group of sounds. Examples of these sounds are a sword swooshing through the air, jet engines, propellers as well as the wind blowing through cracks, etc.  The Aeolian tone is one of the fundamental sounds; the cavity tone and edge tone being others. When designing these sound effects we want to model these fundamental sounds. It then should be possible to make a wide range of sound effects based on these. We want the sounds to be true to the physics generating them and operate in real-time. Completed effects will be suitable for use in video games, TV, film and virtual or augmented reality.

The Aeolian tone is the sound generated when air moves past a string, cylinder or similar object. It’s the whistling noise we may hear coming from a fence in the wind or the swoosh of a sword. An Aeolian Harp is a wind instrument that has been harnessing the Aeolian tone for hundreds of years. If fact, the word Aeolian comes from the Greek god of wind Aeolus.

The physics behind this sound….

When air moves past a cylinder spirals called vortices form behind it, moving away with the air flow. The vortices build up on both sides of the cylinder and detach in an alternating sequence. We call this vortex shedding and the downstream trail of vortices, a Von Karman Vortex Street. An illustration of this is given below:

strouh2

As a vortex sheds from each side there is a change in the lift force from one side to the other. It’s the frequency of this oscillating force that is the fundamental tone frequency. The sound radiates in a direction perpendicular to the flow. There is also a smaller drag force associated with each vortex shed. It is much smaller than the lift force, twice the frequency and radiates parallel to the flow. Both the lift and drag tones have harmonics present.

Can we replicate this…?

In 1878 Vincent Strouhal realized there was a relationship between the diameter of a string, the speed it was travelling thought the air and the frequency of tone produces. We find the Strouhal number varies with the turbulence around the cylinder. Luckily, we have a parameter that represents the turbulence called the Reynolds number. It’s calculated from the viscosity, density and velocity of air, and the diameter of the string. From this we can calculate the Strouhal number and get the fundamental tone frequency.

This is the heart of our model and was the launching point for our model. Acoustic sound sources can be often represented by compact sound sources. These are monopoles, dipoles and quadrupoles. For the Aeolian tone the compact sound source is a dipole.

We have an equation for the acoustic intensity. This is proportional to airspeed to the power of 6. It also includes the relationship between the sound source and listener. The bandwidth around the fundamental tone peak is proportional to the Reynolds number. We calculate this from published experimental results.

The vortex wake acoustic intensity is also calculated. This is much lower that the tone dipole at low airspeed but is proportional to airspeed to the power of 8. There is little wake sound below the fundamental tone frequency and it decreases proportional to the frequency squared.

We use the graphical programming language Pure Data to realise the equations and relationships. A white noise source and bandpass filters can generate the tone sounds and harmonics. The wake noise is a brown noise source shaped by high pass filtering. You can get the Pure Data patch of the model by clicking here.

Our sound effect operates in real-time and is interactive. A user or game engine can adjust:

  • Airspeed
  • Diameter and length of the cylinder
  • Distance between observer and source
  • Azimuth and elevation between observer and source
  • Panning and gain

We can now use the sound source to build up further models. For example, an airspeed model that replicates the wind can reproduce the sound of wind through a fence. The swoosh of a sword is sources lines up in a row with speed adjusted to radius of the arc.

Model complete…?

Not quite. We can calculate the bandwidth of the fundamental tone but have no data for the bandwidth of harmonics. In the current model we set them at the same value. The equation of the acoustic intensity of the wake is an approximation. The equation represents the physics but is not an exact value. We have to use best judgement when scaling it to the acoustic intensity of the fundamental tone.

A string or wire has a natural vibration frequency. There is an interaction between this and the vortex shedding frequency. This modifies the sound heard by a significant factor.