Aural diversity

We are part of a research network that has just been funded, focused around Aural diversity.

Aural Diversity arises from the observation that everybody hears differently. The assumption that we all possess a standard, undifferentiated pair of ears underpins most listening scenarios. Its the basis of many audio technologies, and has been a basis for much of our understanding of hearing and hearing perception. But the assumption is demonstrably incorrect, and taking it too far means that we miss out on many opportunities for advances in auditory science and audio engineering. We may well ask: whose ears are standard? whose ear has primacy? The network investigates the consequences of hearing differences in areas such as: music and performance, soundscape and sound studies, hearing sciences and acoustics, hearing care and hearing technologies, audio engineering and design, creative computing and AI, and indeed any field that has hearing or listening as a major component.

The term ‘auraldiversity’ echoes ‘neurodiversity’ as a way of distinguishing between ‘normal’ hearing, defined by BS ISO 226:2003 as that of a healthy 18-25 year-old, and atypical hearing (Drever 2018, ‘Primacy of the Ear’). This affects everybody to some degree. Each individual’s ears are uniquely shaped. We have all experienced temporary changes in hearing, such as when having a cold. And everybody goes through presbyacusis (age-related hearing loss) at varying rates after the teenage years.

More specific aural divergences are the result of an array of hearing differences or impairments which affect roughly 1.1 billion people worldwide (Lancet, 2013). These include noise-related, genetic, ototoxic, traumatic, and disorder-based hearing loss, some of which may cause full or partial deafness. However, “loss” is not the only form of impairment: auditory perceptual disorders such as tinnitus, hyperacusis and misophonia involve an increased sensitivity to sound.

And its been an issue in our research too. We’ve spent years developing automatic mixing systems that produce audio content like a sound engineer would (De Man et al 2017, ‘Ten Years of Automatic Mixing’). But to do that, we usually assume that there is a ‘right way’ to mix, and of course, it really depends on the listener, the listener’s environment, and many other factors. Our recent research has focused on developing simulators that allow anyone to hear the world as it really sounds to someone with hearing loss.

AHRC is funding the network for two years, beginning July 2021. The network is led by  Andrew Hugill of the University of Leicester. The core partners are the Universities of Leicester, Salford, Nottingham, Leeds, Goldsmiths, Queen Mary University of London (the team behind this blog), and the Attenborough Arts Centre. The wider network includes many more universities and a host of organisations concerned with hearing and listening.

The network will stage five workshops, each with a different focus:

  • Hearing care and technologies. How the use of hearing technologies may affect music and everyday auditory experiences.
  • Scientific and clinical aspects. How an arts and humanities approach might complement, challenge, and enhance scientific investigation.
  • Acoustics of listening differently. How acoustic design of the built and digital environments can be improved.
  • Aural diversity in the soundscape. Includes a concert featuring new works by aurally diverse artists for an aurally diverse audience.
  • Music and performance. Use of new technologies in composition and performance.

See http://auraldiversity.org for more details.

Research highlights for the AES Show Fall 2020

AES_FallShow2020_logo_x

#AESShow

We try to write a preview of the technical track for almost all recent Audio Engineering Society (AES) Conventions, see our entries on the 142nd, 143rd, 144th, 145th147th and 148th Conventions. Like the 148th Convention, the 149th convention, or just the AES Show, is an online event. But one challenge with these sorts of online events is that anything not on the main live stream can get overlooked. The technical papers are available on demand. So though many people can access them, perhaps more than would attend the presentation in person if possible. But they don’t have the feel of an event.

Hopefully, I can give you some idea of the exciting nature of these technical papers. And they really do present a lot of cutting edge and adventurous research. They unveil, for the first time some breakthrough technologies, and both surprising and significant advances in our understanding of audio engineering and related fields.

This time, since all the research papers are available throughout the Convention and beyond, starting Oct. 28th, I haven’t organised them by date. Instead, I’ve divided them into the regular technical papers (usually longer, with more reviewing), and the Engineering Briefs, or E-briefs. The E-briefs are typically smaller, often presenting work-in-progress, late-breaking or just unusual research. Though this time, the unusual appears in the regular papers too.

But first… listening tests. Sooner or later, almost every researcher has to do them. And a good software package will help the whole process run easier. There are two packages presented at the convention. Dale Johnson will present the next generation of a high quality one in the E-Brief ‘HULTI-GEN Version 2 – A Max-based universal listening test framework’. And Stefan Gorzynski will present the paper ‘A flexible software tool for perceptual evaluation of audio material and VR environments’.

E-Briefs

A must for audio educators is Brett Leonard’s ‘A Survey of Current Music Technology & Recording Arts Curriculum Order’. These sorts of programs are often ‘made up’ based on the experience and knowledge of the people involved. Brecht surveyed 35 institutions and analysed the results to establish a holistic framework for the structure of these degree programmes.

The idea of time-stretching as a live phenomenon might seem counterintuitive. For instance, how can you speed up a signal if its only just arriving? And if you slow it down, then surely after a while it lags far enough behind that it is no longer ‘live’. A novel solution is explored in Colin Malloy’s ‘An approach for implementing time-stretching as a live realtime audio effect

The wonderfully titled ‘A Terribly Good Speaker: Understanding the Yamaha NS-10 Phenomenon,’ is all about how and why a low quality loudspeaker with bad reviews became seen as a ‘must have’ amongst many audio professionals. It looks like this presentation will have lessons for those who study marketing, business trends and consumer psychology in almost any sector, not just audio.

Just how good are musicians at tuning their instruments? Not very good, it seems. Or at least, that was what was found out in ‘Evaluating the accuracy of musicians and sound engineers in performing a common drum tuning exercise’, presented by Rob Toulson. But before you start with your favourite drummer joke, note that the participants were all experienced musicians or sound engineers, but not exclusively drummers. So it might be that everyone is bad at drum tuning, whether they’re used to carrying drumsticks around or not.

Matt Cheshire’s ‘Snare Drum Data Set (SDDS): More snare drums than you can shake a stick at’ is worth mentioning just for the title.

Champ Darabundit will present some interesting work on ‘Generalized Digital Second Order Systems Beyond Nyquist Frequency’, showing that the basic filter designs can be tuned to do a lot more than just what is covered in the textbooks. Its interesting and good work, but I have a minor issue with it. The paper only has one reference that isn’t a general overview or tutorial. But there’s lots of good, relevant related work, out there.

I’m involved in only one paper at this convention (shame!). But its well worth checking out. Angeliki Mourgela is presenting ‘Investigation of a Real-Time Hearing Loss Simulation for Audio Production’. It builds on an initial hearing loss simulator she presented at the 147th Convention, but now its higher quality, real-time and available as a VST plugin. This means that audio producers can easily preview what their content would sound like to most listeners with hearing loss.

Masking is an important and very interesting auditory phenomenon. With the emergence of immersive sound, there’s more and more research about spatial masking. But questions come up, like whether artificially panning a source to a location will result in masking the same way as actually placing a source at that location. ‘Spatial auditory masking caused by phantom sound images’, presented by Masayuki Nishiguchi, will show how spatial auditory masking works when sources are placed at virtual locations using rendering techniques.

Technical papers

There’s a double bill presented by Hsein Pew, ‘Sonification of Spectroscopic analysis of food data using FM Synthesis’ and ‘A Sonification Algorithm for Subjective Classification of Food Samples.’ They are unusual papers, but not reallly about classifying food samples. The focus is on the sonification method, which turns data into sounds, allowing listeners to easily discriminate between data collections.

Wow. When I first saw Moorer in the list of presenting authors, I thought ‘what a great coincidence that a presenter has the same last name as one of the great legends in audio engineering. But no, it really is James Moorer. We talked about him before in our blog about the greatest JAES papers of all time. And the abstract for his talk, ‘Audio in the New Millenium – Redux‘, is better than anything I could have written about the paper. He wrote, “In the author’s Heyser lecture in 2000, technological advances from the point of view of digital audio from 1980 to 2000 were summarized then projected 20 years into the future. This paper assesses those projections and comes to the somewhat startling conclusion that entertainment (digital video, digital audio, computer games) has become the driver of technology, displacing military and business forces.”

The paper with the most authors is presented by Lutz Ehrig. And he’ll be presenting a breakthrough, the first ‘Balanced Electrostatic All-Silicon MEMS Speakers’. If you don’t know what that is, you’re not alone. But its worth finding out, because this may be tomorrow’s widespread commercial technology.

If you recorded today, but only using equipment from 1955, would it really sound like a 65 year old recording? Clive Mead will present ‘Composing, Recording and Producing with Historical Equipment and Instrument Models’ which explores just that sort of question. He and his co-authors created and used models to simulate the recording technology and instruments, available at different points in recorded music history.

Degradation effects of water immersion on earbud audio quality,’ presented by Scott Beveridge, sounds at first like it might be very minor work, dipping earbuds in water and then listening to distorted sound from them. But I know a bit about the co-authors. They’re the type to apply rigorous, hardcore science to a problem. And it has practical applications too, since its leading towards methods by which consumers can measure the quality of their earbuds.

Forensic audio is a fascinating field, though most people have only come across it in film and TV shows like CSI, where detectives identify incriminating evidence buried in a very noisy recording. In ‘Forensic Interpretation and Processing of User Generated Audio Recordings’, audio forensics expert Rob Maher looks at how user generated recordings, like when many smartphones record a shooting, can be combined, synchronised and used as evidence.

Mark Waldrep presents a somewhat controversial paper, ‘Native High-Resolution versus Red Book Standard Audio: A Perceptual Discrimination Survey’. He sent out high resolution and CD quality recordings to over 450 participants, asking them to judge which was high resolution. The overall results were little better than guessing. But there were a very large number of questionable decisions in his methodology and interpretation of results. I expect this paper will get the online audiophile community talking for quite some time.

Neural networks are all the rage in machine learning. And for good reason- for many tasks, they outperform all the other methods. There are three neural network papers presented, Tejas Manjunath’s ‘Automatic Classification of Live and Studio Audio Recordings using Convolutional Neural Networks‘, J. T. Colonel’s (who is now part of the team behind this blog) ‘Low Latency Timbre Interpolation and Warping using Autoencoding Neural Networks’ and William Mitchell’s ‘Exploring Quality and Generalizability in Parameterized Neural Audio Effects‘.

The research team here did some unpublished work that seemed to suggest that the mix had only a minimal effect on how people respond to music for untrained listeners, but this became more significant with trained sound engineers and musicians. Kelsey Taylor’s research suggests there’s a lot more to uncover here. In ‘I’m All Ears: What Do Untrained Listeners Perceive in a Raw Mix versus a Refined Mix?’, she performed structured interviews and found that untrained listeners perceive a lot of mixing aspects, but use different terms to describe it.

No loudness measure is perfect. Even the well established ones, like ITU 1770 for broadcast content, or the Glasberg Moore auditory model of loudness perception, see http://www.aes.org/e-lib/browse.cfm?elib=16608 here and http://www.aes.org/e-lib/browse.cfm?elib=17098, have been noted before. In ‘Using ITU-R BS.1770 to Measure the Loudness of Music versus Dialog-based Content’, Scott Norcross shows another issue with the ITU loudness measure, the difficulty in matching levels for speech and music.

Staying on the subject of loudness, Kazuma Watanabe presents ‘The Reality of The Loudness War in Japan -A Case Study on Japanese Popular Music’. This loudness war, the overuse of dynamic range compression, has resulted in lower quality recordings (and annoyingly loud TV and radio ads). It also led to measures like the ITU standard. Watanabe and co-authors measured the increased loudness over the last 30 years, and make a strong

Remember to check the AES E-Library which has all the full papers for all the presentations mentioned here, including listing all authors not just presenters. And feel free to get in touch with us. Josh Reiss (author of this blog entry), J. T. Colonel, and Angeliki Mourgela from the Audio Engineering research team within the Centre for Digital Music, will all be (virtually) there.

Congratulations, Dr. Marco Martinez Ramirez

Today one of our PhD student researchers, Marco Martinez Ramirez, successfully defended his PhD. The form of these exams, or vivas, varies from country to country, and even institution to institution, which we discussed previously. Here, its pretty gruelling; behind closed doors, with two expert examiners probing every aspect of the PhD. And it was made even more challenging since it was all online due to the virus situation.
Marco’s PhD was on ‘Deep learning for audio effects modeling.’

Audio effects modeling is the process of emulating an audio effect unit and seeks to recreate the sound, behaviour and main perceptual features of an analog reference device. Both digital and analog audio effect units  transform characteristics of the sound source. These transformations can be linear or nonlinear, time-invariant or time-varying and with short-term and long-term memory. Most typical audio effect transformations are based on dynamics, such as compression; tone such as distortion; frequency such as equalization; and time such as artificial reverberation or modulation based audio effects.

Simulation of audio processors is normally done by designing mathematical models of these systems. Its very difficult because it seeks to accurately model all components within the effect unit, which usually contains mechanical elements together with nonlinear and time-varying analog electronics. Most audio effects models are either simplified or optimized for a specific circuit or  effect and cannot be efficiently translated to other effects.

Marco’s thesis explored deep learning architectures for audio processing in the context of audio effects modelling. He investigated deep neural networks as black-box modelling strategies to solve this task, i.e. by using only input-output measurements. He proposed several different DSP-informed deep learning models to emulate each type of audio effect transformations.

Marco then explored the performance of these models when modeling various analog audio effects, and analyzed how the given tasks are accomplished and what the models are actually learning. He investigated virtual analog models of nonlinear effects, such as a tube preamplifier; nonlinear effects with memory, such as a transistor-based limiter; and electromechanical nonlinear time-varying effects, such as a Leslie speaker cabinet and plate and spring reverberators.

Marco showed that the proposed deep learning architectures represent an improvement of the state-of-the-art in black-box modeling of audio effects and the respective directions of future work are given.

His research also led to a new start-up company, TONZ, which build on his machine learning techniques to provide new audio processing interactions for the next generation of musicians and music makers.

Here’s a list of some of Marco’s papers that relate to his PhD research while a member of the Intelligent Sound Engineering team.

Congratulations again, Marco!

Venturous Views on Virtual Vienna – a preview of AES 148

#VirtualVienna

We try to write a preview of the technical track for almost all recent Audio Engineering Society (AES) Conventions, see our entries on the 142nd, 143rd, 144th, 145th and 147th Conventions. But this 148th Convention is very different.

It is, of course, an online event. The Convention planning committee have put huge effort into putting it all online and making it a really engaging and exciting experience (and in massively reducing costs). There will be a mix of live-streams, break out sessions, interactive chat rooms and so on. But the technical papers will mostly be on-demand viewing, with Q&A and online dialog with the authors. This is great in the sense that you can view it and interact with authors any time, but it means that its easy to overlook really interesting work.

So we’ve gathered together some information about a lot of the presented research that caught our eye as being unusual, exceptionally high quality, or just worth mentioning. And every paper mentioned here will appear soon in the AES E-Library, by the way. Currently though, you can browse all the abstracts by searching the full papers and engineering briefs on the Convention website.

Deep learning and neural networks are all the rage in machine learning nowadays. A few contributions to the field will be presented by Eugenio Donati with ‘Prediction of hearing loss through application of Deep Neural Network’, Simon Plain with ‘Pruning of an Audio Enhancing Deep Generative Neural Network’, Giovanni Pepe’s presentation of ‘Generative Adversarial Networks for Audio Equalization: an evaluation study’, Yiwen Wang presenting ‘Direction of arrival estimation based on transfer function learning using autoencoder network’, and the author of this post, Josh Reiss will present work done mainly by sound designer/researcher Guillermo Peters, ‘A deep learning approach to sound classification for film audio post-production’. Related to this, check out the Workshop on ‘Deep Learning for Audio Applications – Engineering Best Practices for Data’, run by Gabriele Bunkheila of MathWorks (Matlab), which will be live-streamed  on Friday.

There’s enough work being presented on spatial audio that there could be a whole conference on the subject within the convention. A lot of that is in Keynotes, Workshops, Tutorials, and the Heyser Memorial Lecture by Francis Rumsey. But a few papers in the area really stood out for me. Toru Kamekawa’s investigated a big question with ‘Are full-range loudspeakers necessary for the top layer of 3D audio?’ Marcel Nophut’s ‘Multichannel Acoustic Echo Cancellation for Ambisonics-based Immersive Distributed Performances’ has me intrigued because I know a bit about echo cancellation and a bit about ambisonics, but have no idea how to do the former for the latter.

And I’m intrigued by ‘Creating virtual height loudspeakers using VHAP’, presented by Kacper Borzym. I’ve never heard of VHAP, but the original VBAP paper is the most highly cited paper in the Journal of the AES (1367 citations at the time of writing this).

How good are you at understanding speech from native speakers? How about when there’s a lot of noise in the background? Do you think you’re as good as a computer? Gain some insight into related research when viewing the presentation by Eugenio Donati on ‘Comparing speech identification under degraded acoustic conditions between native and non-native English speakers’.

There’s a few papers exploring creative works, all of which look interesting and have great titles. David Poirier-Quinot will present ‘Emily’s World: behind the scenes of a binaural synthesis production’. Music technology has a fascinating history. Michael J. Murphy will explore the beginning of a revolution with ‘Reimagining Robb: The Sound of the World’s First Sample-based Electronic Musical Instrument circa 1927’. And if you’re into Scandinavian instrumental rock music (and who isn’t?), Zachary Bresler’s presentation of ‘Music and Space: A case of live immersive music performance with the Norwegian post-rock band Spurv’ is a must.

robb

Frank Morse Robb, inventor of the first sample-based electronic musical instrument.

But sound creation comes first, and new technologies are emerging to do it. Damian T. Dziwis will present ‘Body-controlled sound field manipulation as a performance practice’. And particularly relevant given the worldwide isolation going on is ‘Quality of Musicians’ Experience in Network Music Performance: A Subjective Evaluation,’ presented by Konstantinos Tsioutas.

Portraiture looks at how to represent or capture the essence and rich details of a person. Maree Sheehan explores how this is achieved sonically, focusing on Maori women, in an intriguing presentation on ‘Audio portraiture sound design- the development and creation of audio portraiture within immersive and binaural audio environments.’

We talked about exciting research on metamaterials for headphones and loudspeakers when giving previews of previous AES Conventions, and there’s another development in this area presented by Sebastien Degraeve in ‘Metamaterial Absorber for Loudspeaker Enclosures’

Paul Ferguson and colleagues look set to break some speed records, but any such feats require careful testing first, as in ‘Trans-Europe Express Audio: testing 1000 mile low-latency uncompressed audio between Edinburgh and Berlin using GPS-derived word clock’

Our own research has focused a lot on intelligent music production, and especially automatic mixing. A novel contribution to the field, and a fresh perspective, is given in Nyssim Lefford’s presentation of ‘Mixing with Intelligent Mixing Systems: Evolving Practices and Lessons from Computer Assisted Design’.

Subjective evaluation, usually in the form of listening tests, is the primary form of testing audio engineering theory and technology. As Feynman said, ‘if it disagrees with experiment, its wrong!’

And thus, there are quite a few top-notch research presentations focused on experiments with listeners. Minh Voong looks at an interesting aspect of bone conduction with ‘Influence of individual HRTF preference on localization accuracy – a comparison between regular and bone conducting headphones. Realistic reverb in games is incredibly challenging because characters are always moving, so Zoran Cvetkovic tackles this with ‘Perceptual Evaluation of Artificial Reverberation Methods for Computer Games.’ The abstract for Lawrence Pardoe’s ‘Investigating user interface preferences for controlling background-foreground balance on connected TVs’ suggests that there’s more than one answer to that preference question. That highlights the need for looking deep into any data, and not just considering the mean and standard deviation, which often leads to Simpson’s Paradox. And finally, Peter Critchell will present ‘A new approach to predicting listener’s preference based on acoustical parameters,’ which addresses the need to accurately simulate and understand listening test results.

There are some talks about really rigorous signal processing approaches. Jens Ahren will present ‘Tutorial on Scaling of the Discrete Fourier Transform and the Implied Physical Units of the Spectra of Time-Discrete Signals.’ I’m excited about this because it may shed some light on a possible explanation for why we hear a difference between CD quality and very high sample rate audio formats.

The Constant-Q Transform represents a signal in frequency domain, but with logarithmically spaced bins. So potentially very useful for audio. The last decade has seen a couple of breakthroughs that may make it far more practical.  I was sitting next to Gino Velasco when he won the “best student paper” award for Velasco et al.’s “Constructing an invertible constant-Q transform with nonstationary Gabor frames.” Schörkhuber and Klapuri also made excellent contributions, mainly around implementing a fast version of the transform, culminating in a JAES paper. and the teams collaborated together on a popular Matlab toolbox. Now there’s another advance with Felix Holzmüller presenting ‘Computational efficient real-time capable constant-Q spectrum analyzer’.

The abstract for Dan Turner’s ‘Content matching for sound generating objects within a visual scene using a computer vision approach’ suggests that it has implications for selection of sound effect samples in immersive sound design. But I’m a big fan of procedural audio, and think this could have even higher potential for sound synthesis and generative audio systems.

And finally, there’s some really interesting talks about innovative ways to conduct audio research based on practical challenges. Nils Meyer-Kahlen presents ‘DIY Modifications for Acoustically Transparent Headphones’. The abstract for Valerian Drack’s ‘A personal, 3D printable compact spherical loudspeaker array’, also mentions its use in a DIY approach. Joan La Roda’s own experience of festival shows led to his presentation of ‘Barrier Effect at Open-air Concerts, Part 1’. Another presentation with deep insights derived from personal experience is Fabio Kaiser’s ‘Working with room acoustics as a sound engineer using active acoustics.’ And the lecturers amongst us will be very interested in Sebastian Duran’s ‘Impact of room acoustics on perceived vocal fatigue of staff-members in Higher-education environments: a pilot study.’

Remember to check the AES E-Library which will soon have all the full papers for all the presentations mentioned here, including listing all authors not just presenters. And feel free to get in touch with us. Josh Reiss (author of this blog entry), J. T. Colonel, and Angeliki Mourgela from the Audio Engineering research team within the Centre for Digital Music, will all be (virtually) there.

Awesome student projects in sound design and audio effects

I teach classes in Sound Design and Digital Audio Effects. In both classes, the final assignment involves creating an original work that involves audio programming and using concepts taught in class. But the students also have a lot of free reign to experiment and explore their own ideas. The results are always great. Lots of really cool ideas, many of which could lead to a publication, or would be great to listen to regardless of the fact that it was an assignment.

The last couple of years, I posted about it here and here.  Here’s a few of the projects this year.

From the Sound Design class;

  • A procedural audio model of a waterfall. The code was small, involving some filtered noise sources with random gain changes, but the result was great.waterfall2
  • An interactive animation of a girl writing at a desk during a storm. There were some really neat tricks to get a realistic thunder sound.
  • A procedurally generated sound scene for a walk through the countryside. The student found lots of clever ways to generate the sounds of birds, bees, a river and the whoosh of a passing car.
  • New sound design replacing the audio track in a film scene. Check it out.

And from the Digital Audio Effects class;

  • I don’t need to mention anything about the next one. Just read the student’s tweet.

 

  • Rainmaker, a VST plugin that takes an incoming signal and transforms it into a ‘rain’ like sound, starting above the listener and then floating down below.

  • A plugin implementation of the Karplus-Strong algorithm, except an audio sample is used to excite the string instead of a noise burst. It gives really interesting timbral qualities.

  • Stormify, an audio plugin that enables users to add varying levels of rain and wind to the background of their audio, making it appear that the recording took place in inclement weather.
  • An all-in-one plugin for synthesising and sculpting drum-like sounds.
  • The Binaural Phase Vocoder, a VST/AU plugin whereby users can position a virtual sound source in a 3D space and process the sound through an overlap-add phase vocoder.
  • A multiband multi-effect consisting of three frequency bands and three effects on each band: delay, distortion, and tremolo. Despite the seeming complexity, the interface was straightforward and easy to use.

multi-interface

There were many other interesting assignments, including several sonifications of images. But this selection really shows both the talent of the students and the possibilities to create new and interesting sounds.

Intelligent Music Production book is published

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Ryan Stables is an occasional collaborator and all around brilliant person. He started the annual Workshop on Intelligent Music Production (WIMP) in 2015. Its been going strong ever since, with the 5th WIMP co-located with DAFx, this past September. The workshop series focuses on the application of intelligent systems (including expert systems, machine learning, AI) to music recording, mixing, mastering and related aspects of audio production or sound engineering.

Ryan had the idea for a book about the subject, and myself (Josh Reiss) and Brecht De Man (another all around brilliant person) were recruited as co-authors. What resulted was a massive amount of writing, editing, refining, re-editing and so on. We all contributed big chunks of content, but Brecht pulled it all together and turned it into something really high quality giving a comprehensive overview of the field, suitable for a wide range of audiences.

And the book is finally published today, October 31st! Its part of the AES Presents series by Focal Press, a division of Routledge. You can get it from the publisher, from Amazon or any of the other usual places.

And here’s the official blurb

Intelligent Music Production presents the state of the art in approaches, methodologies and systems from the emerging field of automation in music mixing and mastering. This book collects the relevant works in the domain of innovation in music production, and orders them in a way that outlines the way forward: first, covering our knowledge of the music production processes; then by reviewing the methodologies in classification, data collection and perceptual evaluation; and finally by presenting recent advances on introducing intelligence in audio effects, sound engineering processes and music production interfaces.

Intelligent Music Production is a comprehensive guide, providing an introductory read for beginners, as well as a crucial reference point for experienced researchers, producers, engineers and developers.

 

Radical and rigorous research at the upcoming Audio Engineering Society Convention

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We previewed the 142nd, 143rd, 144th  and 145th Audio Engineering Society (AES) Conventions, which we also followed with wrap-up discussions. Then we took a break, but now we’re back to preview the 147th AES  convention, October 16 to 19 in New York. As before, the Audio Engineering research team here aim to be quite active at the convention.

We’ve gathered together some information about a lot of the research-oriented events that caught our eye as being unusual, exceptionally high quality, involved in, attending, or just worth mentioning. And this Convention will certainly live up to the hype.

Wednesday October 16th

When I first read the title of the paper ‘Evaluation of Multichannel Audio in Automobiles versus Mobile Phones‘, presented at 10:30, I thought it was a comparison of multichannel automotive audio versus the tinny, quiet mono or barely stereo from a phone. But its actually comparing results of a listening test for stereo vs multichannel in a car, with results of a listening test for stereo vs multichannel for the same audio, but from a phone and rendered over headphones. And the results look quite interesting.

Deep neural networks are all the rage. We’ve been using DNNs to profile a wide variety of audio effects. Scott Hawley will be presenting some impressive related work at 9:30, ‘Profiling Audio Compressors with Deep Neural Networks.’

We previously presented work on digital filters that closely match their analog equivalents. We pointed out that such filters can have cut-off frequencies beyond Nyquist, but did not explore that aspect. ‘Digital Parametric Filters Beyond Nyquist Frequency‘, at 10 am, investigates this idea in depth.

I like a bit of high quality mathematical theory, and that’s what you get in Tamara Smyth’s 11:30 paper ‘On the Similarity between Feedback/Loopback Amplitude and Frequency Modulation‘, which shows a rather surprising (at least at first glance) equivalence between two types of feedback modulation.

There’s an interesting paper at 2pm, ‘What’s Old Is New Again: Using a Physical Scale Model Echo Chamber as a Real-Time Reverberator‘, where reverb is simulated not with impulse response recordings, or classic algorithms, but using scaled models of echo chambers.

At 4 o’clock, ‘A Comparison of Test Methodologies to Personalize Headphone Sound Quality‘ promises to offer great insights not just for headphones, but into subjective evaluation of audio in general.

There’s so many deep learning papers, but the 3-4:30 poster ‘Modal Representations for Audio Deep Learning‘ stands out from the pack. Deep learning for audio most often works with raw spectrogram data. But this work proposes learning modal filterbank coefficients directly, and they find it gives strong results for classification and generative tasks. Also in that session, ‘Analysis of the Sound Emitted by Honey Bees in a Beehive‘ promises to be an interesting and unusual piece of work. We talked about their preliminary results in a previous entry, but now they’ve used some rigorous audio analysis to make deep and meaningful conclusions about bee behaviour.

Immerse yourself in the world of virtual and augmented reality audio technology today, with some amazing workshops, like Music Production in VR and AR, Interactive AR Audio Using Spark, Music Production in Immersive Formats, ISSP: Immersive Sound System Panning, and Real-time Mixing and Monitoring Best Practices for Virtual, Mixed, and Augmented Reality. See the Calendar for full details.

Thursday, October 17th

An Automated Approach to the Application of Reverberation‘, at 9:30, is the first of several papers from our team, and essentially does something to algorithmic reverb similar to what “Parameter Automation in a Dynamic Range Compressor” did for a dynamic range compressor.

Why do public address (PA) systems sound for large venues sound so terrible? They actually have regulations for speech intelligibility. But this is only measured in empty stadiums. At 11 am, ‘The Effects of Spectators on the Speech Intelligibility Performance of Sound Systems in Stadia and Other Large Venues‘ looks at the real world challenges when the venue is occupied.

Two highlights of the 9-10:30 poster session, ‘Analyzing Loudness Aspects of 4.2 Million Musical Albums in Search of an Optimal Loudness Target for Music Streaming‘ is interesting, not just for the results, applications and research questions, but also for the fact that involved 4.2 million albums. Wow! And there’s a lot more to audio engineering research than what one might think. How about using acoustic sensors to enhance autonomous driving systems, which is a core application of ‘Audio Data Augmentation for Road Objects Classification‘.

Audio forensics is a fascinating world, where audio engineering is often applied to unusually but crucially. One such situation is explored at 2:15 in ‘Forensic Comparison of Simultaneous Recordings of Gunshots at a Crime Scene‘, which involves looking at several high profile, real world examples.

Friday, October 18th

There are two papers looking at new interfaces for virtual reality and immersive audio mixing, ‘Physical Controllers vs. Hand-and-Gesture Tracking: Control Scheme Evaluation for VR Audio Mixing‘ at 10:30, and ‘Exploratory Research into the Suitability of Various 3D Input Devices for an Immersive Mixing Task‘ at 3:15.

At 9:15, J. T. Colonel from our group looks into the features that relate, or don’t relate, to preference for multitrack mixes in ‘Exploring Preference for Multitrack Mixes Using Statistical Analysis of MIR and Textual Features‘, with some interesting results that invalidate some previous research. But don’t let negative results discourage ambitious approaches to intelligent mixing systems, like Dave Moffat’s (also from here) ‘Machine Learning Multitrack Gain Mixing of Drums‘, which follows at 9:30.

Continuing this theme of mixing analysis and automation is the poster ‘A Case Study of Cultural Influences on Mixing Preference—Targeting Japanese Acoustic Major Students‘, shown from 3:30-5, which does a bit of meta-analysis by merging their data with that of other studies.

Just below, I mention the need for multitrack audio data sets. Closely related, and also much needed, is this work on ‘A Dataset of High-Quality Object-Based Productions‘, also in the 3:30-5 poster session.

Saturday, October 19th

We’re approaching a world where almost every surface can be a visual display. Imagine if every surface could be a loudspeaker too. Such is the potential of metamaterials, discussed in ‘Acoustic Metamaterial in Loudspeaker Systems Design‘ at 10:45.

Another session, 9 to 11:30 has lots of interesting presentations about music production best practices. At 9, Amandine Pras presents ‘Production Processes of Pop Music Arrangers in Bamako, Mali‘. I doubt there will be many people at the convention who’ve thought about how production is done there, but I’m sure there will be lots of fascinating insights. This is followed at 9:30 by ‘Towards a Pedagogy of Multitrack Audio Resources for Sound Recording Education‘. We’ve published a few papers on multitrack audio collections, sorely needed for researchers and educators, so its good to see more advances.

I always appreciate filling the gaps in my knowledge. And though I know a lot about sound enhancement, I’ve never dived into how its done and how effective it is in soundbars, now widely used in home entertainment. So I’m looking forward to the poster ‘A Qualitative Investigation of Soundbar Theory‘, shown 10:30-12. From the title and abstract though, this feels like it might work better as an oral presentation. Also in that session, the poster ‘Sound Design and Reproduction Techniques for Co-Located Narrative VR Experiences‘ deserves special mention, since it won the Convention’s Best Peer-Reviewed Paper Award, and promises to be an important contribution to the growing field of immersive audio.

Its wonderful to see research make it into ‘product’, and ‘Casualty Accessible and Enhanced (A&E) Audio: Trialling Object-Based Accessible TV Audio‘, presented at 3:45, is a great example. Here, new technology to enhance broadcast audio for those with hearing loss iwas trialed for a popular BBC drama, Casualty. This is of extra interest to me since one of the researchers here, Angeliki Mourgela, does related research, also in collaboration with BBC. And one of my neighbours is an actress who appears on that TV show.

I encourage the project students working with me to aim for publishable research. Jorge Zuniga’s ‘Realistic Procedural Sound Synthesis of Bird Song Using Particle Swarm Optimization‘, presented at 2:30, is a stellar example. He created a machine learning system that uses bird sound recordings to find settings for a procedural audio model. Its a great improvement over other methods, and opens up a whole field of machine learning applied to sound synthesis.

At 3 o’clock in the same session is another paper from our team, Angeliki Mourgela presenting ‘Perceptually Motivated Hearing Loss Simulation for Audio Mixing Reference‘. Roughly 1 in 6 people suffer from some form of hearing loss, yet amazingly, sound engineers don’t know what the content will sound like to them. Wouldn’t it be great if the engineer could quickly audition any content as it would sound to hearing impaired listeners? That’s the aim of this research.

About three years ago, I published a meta-analysis on perception of high resolution audio, which received considerable attention. But almost all prior studies dealt with music content, and there are good reasons to consider more controlled stimuli too (noise, tones, etc). The poster ‘Discrimination of High-Resolution Audio without Music‘ does just that. Similarly, perceptual aspects of dynamic range compression is an oft debated topic, for which we have performed listening tests, and this is rigorously investigated in ‘Just Noticeable Difference for Dynamic Range Compression via “Limiting” of a Stereophonic Mix‘. Both posters are in the 3-4:30 session.

The full program can be explored on the Convention Calendar or the Convention website. Come say hi to us if you’re there! Josh Reiss (author of this blog entry), J. T. Colonel, Angeliki Mourgela and Dave Moffat from the Audio Engineering research team within the Centre for Digital Music, will all be there.

Cool sound design and audio effects projects

Every year, I teach two classes (modules), Sound Design and Digital Audio Effects. In both classes, the final assignment involves creating an original work that involves audio programming and using concepts taught in class. But the students also have a lot of free reign to experiment and explore their own ideas. Last year, I had a well received blog entry about the projects.

The results are always great. Lots of really cool ideas, many of which could lead to a publication, or would be great to listen to regardless of the fact that it was an assignment. Here’s a few of the projects this year.

From the Sound Design class;

  • A truly novel abstract sound synthesis (amplitude and frequency modulation) where parameters are controlled by pitch recognition and face recognition machine learning models, using the microphone and the webcam. Users could use their voice and move their face around to affect the sound.
  • An impressive one had six sound models: rain, bouncing ball, sea waves, fire, wind and explosions. It also had a website where each synthesised sound could be compared against real recordings. We couldn’t always tell which was real and which was synthesised!

SoundSelect.png

  • An auditory model of a London Underground train, from the perspective of a passenger on a train, or waiting at a platform. It had a great animation.

train

  • Two projects involved creating interactive soundscapes auralising an image. One involved a famous photo taken by the photographer, Gregory Crewdson. encapsulating  a dark side of suburban America through surreal, cinematic imagery. The other was an estate area, where there are no bodies visible , giving the impression of an eerie atmosphere where background noises and small sounds are given prominence.

And from the Digital Audio Effects class;

  • A create-your-own distortion effect, where the user can interactively modify the wave shaping curve.
  • Input dependant modulation signal based on the physical mass/ spring system
  • A Swedish death metal guitar effect combining lots of effects for a very distinctive sound
  • A very creative all-in-one audio toy, ‘Ring delay’. This  augmented ping-pong delay effect gives controls over the panning of the delays, the equalization of the audio input and delays, and the output gain. Delays can be played backwards, and the output can be set out-of-phase. Finally, a ring modulator can modulate the audio input to create new sounds to be delayed.
  • Chordify, which transforms an incoming signal, ideally individual notes, into a chord of three different pitches.

chordify

  • An audio effects chain inspired by interning at a local radio station. The student helped the owner produce tracks using effects chain presets. But this producers understanding of compressors, EQ, distortion effects… was fairly limited. So the student recreated one of the effects chains into a plugin that only has two adjustable parameters which control multiple parameters inside. 
  • Old Styler, a plug-in that applies sort of a ‘vintage’ effect so that it sounds like from an old radio or an old, black and white movie. Here’s how it sounds.

  • There were some advanced reverbs, including a VST implementation of a state-of-the-art reverberation algorithm known as a Scattering Delay Network (SDN), and a Church reverb incorporating some additional effects to get that ‘church sound’ just right.
  • A pretty amazing cave simulator, with both reverb and random water droplet sounds as part of the VST plug-in.

CaveSimulator

  • A bit crusher, which also had noise, downsampling and filtering to allow lots of ways to degrade the signal.
  • A VST implementation of the Euclidian Algorithm for world rhythms as described by Goddfried Toussaint in his paper The Euclidean Algorithm Generates Traditional Musical Rhythms.
  • A mid/side processor, with excellent analysis to verify that the student got the implementation just right.
  • Multi-functional distortion pedal. Guitarists often compose music in their bedroom and would benefit from having an effect to facilitate filling the song with a range of sounds, traditionally belonging to other instruments. That’s what this plug-in did, using a lot of clever tricks to widen the soundstage of the guitar.
  • Related to the multi-functional distortion, two students created multiband distortion effects.
  • A Python project that separates a track into harmonic, percussive, and residual components which can be adjusted individually.
  • An effect that attempts to resynthesise any audio input with sine wave oscillators that take their frequencies from the well-tempered scale. This goes far beyond auto-tune, yet can be quite subtle.
  • A source separator plug-in based on Dan Barry’s ADRESS algorithm, described here and here. Along with Mikel Gainza, Dan Barry cofounded the company Sonic Ladder, which released the successful software Riffstation, based on their research.

There were many other interesting assignments, including several variations on tape emulation. But this selection really shows both the talent of the students and the possibilities to create new and interesting sounds.

Audiology and audio production PhD studentship available for UK residents

BBC R&D and Queen Mary University of London’s School of Electronic Engineering and Computer Science have an ICASE PhD studentship available for a talented researcher. It will involve researching the idea of intelligent mixing of broadcast audio content for hearing impaired audiences.

Perceptual Aspects of Broadcast Audio Mixing for Hearing Impaired Audiences

Project Description

This project will explore new approaches to audio production to address hearing loss, a growing concern with an aging population. The overall goal is to investigate, implement and validate original strategies for mixing broadcast content such that it can be delivered with improved perceptual quality for hearing impaired people.

Soundtracks for television and radio content typically have dialogue, sound effects and music mixed together with normal-hearing listeners in mind. But a hearing impairment may result in this final mix sounding muddy and cluttered. First, hearing aid strategies will be investigated, to establish their limitations and opportunities for improving upon them with object- based audio content. Then different mixing strategies will be implemented to counteract the hearing impairment. These strategies will be compared against each other in extensive listening tests, to establish preferred approaches to mixing broadcast audio content.

Requirements and details

This is a fully funded, 4 year studentship which includes tuition fees, travel and consumables allowance and a stipend covering living expenses.

Skills in signal processing, audio production and auditory models are preferred, though we encourage any interested and talented researchers to apply. A successful candidate will have an academic background in engineering, science or maths.

The student will be based in London. Time will be spent  between QMUL’s Audio Engineering team (the people behind this blog) in the Centre for Digital Music and BBC R&D South Lab, with a minimum of six months at each.

The preferred start date is January 2nd, 2019.
All potential candidates must meet UK residency requirements, e.g. normally EU citizen with long-term residence in the UK. Please check the regulations if you’re unsure.

If interested, please contact Prof. Josh Reiss at joshua.reiss@qmul.ac.uk .

Sneak preview of the research to be unveiled at the 145th Audio Engineering Society

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We’ve made it a tradition on this blog to preview the technical program at the Audio Engineering Society Conventions, as we did with the 142nd, 143rd, and 144th AES Conventions. The 145th AES  convention is just around the corner, October 17 to 20 in New York. As before, the Audio Engineering research team behind this blog will be quite active at the convention.

These conventions have thousands of attendees, but aren’t so large that you get lost or overwhelmed. Away from the main exhibition hall is the Technical Program, which includes plenty of tutorials and presentations on cutting edge research.

So we’ve gathered together some information about a lot of the events that caught our eye as being unusual, exceptionally high quality involved in, attending, or just worth mentioning. And this Convention will certainly live up to the hype. Plus, its a special one, the 70th anniversary of the founding of the AES.

By the way, I don’t think I mention a single loudspeaker paper below, but the Technical Program is full of them this time. You could have a full conference just on loudspeakers from them. If you want to become an expert on loudspeaker research, this is the place to be.

Anyway, lets dive right in.

Wednesday, October 17th

We know different cultures listen to music differently, but do they listen to audio coding artifacts differently? Find out at 9:30 when Sascha Disch and co-authors present On the Influence of Cultural Differences on the Perception of Audio Coding Artifacts in Music.

ABX, AB, MUSHRA… so many choices for subjective evaluation and listening tests, so little time. Which one to use, which one gives the strongest results? Lets put them all to the test while looking at the same question. This is what was done in Investigation into the Effects of Subjective Test Interface Choice on the Validity of Results, presented at 11:30. The results are strong, and surprising. Authors include former members of the team behind this blog, Nick Jillings and Brecht de Man, myself and frequent collaborator Ryan Stables.

From 10-11:30, Steve Fenton will be presenting the poster Automatic Mixing of Multitrack Material Using Modified Loudness Models. Automatic mixing is a really hot research area, one where we’ve made quite a few contributions. And a lot of it has involved loudness models for level balancing or fader settings. Someone really should do a review of all the papers focused on that, or better yet, a meta-analysis. Dr. Fenton and co-authors also have another poster in the same session, about a Real-Time System for the Measurement of Perceived Punch. Fenton’s PhD was about perception and modelling of punchiness in audio, and I suggested to him that the thesis should have just been titled ‘Punch!’

The researchers from Harman continue their analysis of headphone preference and quality with A Survey and Analysis of Consumer and Professional Headphones Based on Their Objective and Subjective Performances at 3:30. Harman obviously have a strong interest in this, but its rigorous, high quality research, not promotion.

In the 3:00 to 4:30 poster session, Daniel Johnston presents a wonderful spatial audio application, SoundFields: A Mixed Reality Spatial Audio Game for Children with Autism Spectrum Disorder. I’m pretty sure this isn’t the quirky lo-fi singer/songwriter Daniel Johnston.

Thursday, October 18th

There’s something bizarre about the EBU R128 / ITU-R BS.1770 specification for loudness measurements. It doesn’t give the filter coefficients as a function of sample rate. So, for this and other reasons, even though the actual specification is just a few lines of code, you have to reverse engineer it if you’re doing it yourself, as was done here. At 10 am, Brecht de Man presents Evaluation of Implementations of the EBU R128 Loudness Measurement, which looks carefully at different implementations and provides full implementations in several programming languages.

Roughly one in six people in developed countries suffer some hearing impairment. If you think that seems too high, think how many wear glasses or contact lenses or had eye surgery. And given the sound exposure, I’d expect the average to be higher with music producers. But we need good data on this. Thus, Laura Sinnott’s 3 pm presentation on Risk of Sound-Induced Hearing Disorders for Audio Post Production Engineers: A Preliminary Study is particularly relevant.

Some interesting posters in the 2:45 to 4:15 session. Maree Sheehan’s Audio Portraiture –The Sound of Identity, an Indigenous Artistic Enquiry uses 3D immersive and binaural sound to create audio portraits of Maori women. Its a wonderful use of state of the art audio technologies for cultural and artistic study. Researchers from the University of Alcala in Madrid present an improved method to detect anger in speech in Precision Maximization in Anger Detection in Interactive Voice Response Systems.

Friday, October 19th

There’s plenty of interesting papers this day, but only one I’m highlighting. By coincidence, its my own presentation of work with He Peng, on Why Can You Hear a Difference between Pouring Hot and Cold Water? An Investigation of Temperature Dependence in Psychoacoustics. This was inspired by the curious phenomenon and initial investigations described in a previous blog entry.

Saturday, October 20th

Get there early on Saturday to find out about audio branding from a designer’s perspective in the 9 am Creative Approach to Audio in Corporate Brand Experiences.

Object-based audio allows broadcasters to deliver separate channels for sound effects, music and dialog, which can then be remixed on the client-side. This has high potential for delivering better sound for the hearing-impaired, as described in Lauren Ward’s Accessible Object-Based Audio Using Hierarchical Narrative Importance Metadata at 9:45. I’ve heard this demonstrated by the way, and it sounds amazing.

A big challenge with spatial audio systems is the rendering of sounds that are close to the listener. Descriptions of such systems almost always begin with ‘assume the sound source is in the far field.’ In the 10:30 to 12:00 poster session, researchers from the Chinese Academy of Science present a real advance in this subject with Near-Field Compensated Higher-Order Ambisonics Using a Virtual Source Panning Method.

Rob Maher is one of the world’s leading audio forensics experts. At 1:30 in Audio Forensic Gunshot Analysis and Multilateration, he looks at how to answer the question ‘Who shot first?’ from audio recordings. As is often the case in audio forensics, I suspect this paper was motivated by real court cases.

When visual cues disagree with auditory cues, which ones do you believe? Or conversely, does low quality audio seem more realistic if strengthened by visual cues? These sorts of questions are investigated at 2 pm in the large international collaboration Influence of Visual Content on the Perceived Audio Quality in Virtual Reality. Audio Engineering Society Conventions are full of original research, but survey and review papers are certainly welcomed, especially ones like the thorough and insightful HRTF Individualization: A Survey, presented at 2:30.

Standard devices for measuring auditory brainstem response are typically designed to work only with clicks or tone bursts. A team of researchers from Gdansk developed A Device for Measuring Auditory Brainstem Responses to Audio, presented in the 2:30 to 4 pm poster session.

 

Hopefully, I can also give a wrap-up after the Convention, as we did here and here.