Cross-adaptive audio effects: automatic mixing, live performance and everything in between

Our paper on Applications of cross-adaptive audio effects: automatic mixing, live performance and everything in between has just been published in Frontiers in Digital Humanities. It is a systematic review of cross-adaptive audio effects and their applications.

Cross-adaptive effects extend the boundaries of traditional audio effects by having many inputs and outputs, and deriving their behavior based on analysis of the signals and their interaction. This allows the audio effects to adapt to different material, seemingly being aware of what they do and listening to the signals. Here’s a block diagram showing how a cross-adaptive audio effect modifies a signal.

cross-adaptive architecture

Last year, we published a paper reviewing the history of automatic mixing, almost exactly ten years to the day from when automatic mixing was first extended beyond simple gain changes for speech applications. These automatic mixing applications rely on cross-adaptive effects, but the effects can do so much more.

Here’s an example automatic mixing system from our youtube channel, IntelligentSoundEng.

When a musician uses the signals of other performers directly to inform the timbral character of her own instrument, it enables a radical expansion of interaction during music making. Exploring this was the goal of the Cross-adaptive processing for musical intervention project, led by Oeyvind Brandtsegg, which we discussed in an earlier blog entry. Using cross-adaptive audio effects, musicians can exert control over each the instruments and performance of other musicians, both leading to new competitive aspects and new synergies.

Here’s a short video demonstrating this.

Despite various projects, research and applications involving cross-adaptive audio effects, there is still a fair amount of confusion surrounding the topic. There are multiple definitions, sometimes even by the same authors. So this paper gives a brief history of applications as well as a classification of effects types and clarifies issues that have come up in earlier literature. It further defines the field, lays a formal framework, explores technical aspects and applications, and considers the future from artistic, perceptual, scientific and engineering perspectives.

Check it out!


Analogue matched digital EQ: How far can you go linearly?

(Background post for the paper “Improving the frequency response magnitude and phase of
analogue-matched digital filters” by John Flynn & Josh Reiss for AES Milan 2018)

Professional audio mastering is a field that is still dominated by analogue hardware. Many mastering engineers still favour their go-to outboard compressors and equalisers over digital emulations. As a practising mastering engineer myself, I empathise. Quality analogue gear has a proven track record in terms of sonic quality spanning about a century. Even though digital approximations of analogue tools have gotten better, particularly over the past decade, I too have tended to reach for analogue hardware. However, through my research at Queen Mary with Professor Josh Reiss, that is changing.

When modelling an analogue EQ, a lot of focus has been in modelling distortions and other non-linearities, we chose to look at the linear component. Have we reached a ceiling in terms of modelling an analogue prototype filter in the digital domain? Can we do better? We found that yes there was room for improvement and yes we can do better.

The milestone of research in this area is Orfanidis’ 1997 paper “Digital parametric equalizer design with prescribed Nyquist-frequency gain“, the first major improvement over the bilinear transform which has a reknowned ‘cramped’ sound in the high frequencies. Basically, the bilinear transform is what all first generation digital equalisers is based on. It’s high frequencies towards 20kHz drops sharply, giving a ‘closed/cramped’ sound. Orfanidis and later improvements by Massberg [9] & Gunness/Chauhan [10] give a much better approximation of an analogue prototype.


However [9],[10] improve magnitude, they don’t capture analogue phase. Bizarrely, the bilinear transform performs reasonably well on phase. So we knew it was possible.

So the problem is: how do you get a more accurate magnitude match to analogue than [9],[10]? While also getting a good match to phase? Many attempts, including complicated iterative Parks/McClellen filter design approaches, fell flat. It turned out that Occam was right, in this case a simple answer was the better answer.

By combining a matched-z transform, frequency sampling filter design and a little bit of clever coefficient manipulation, we achieved excellent results. A match to the analogue prototype to an arbitrary degree. At low filter lengths you get a filter that performs as well as [9],[10] in magnitude but also matches analogue phase. By using longer filter lengths the match to analogue is extremely precise, in both magnitude and phase (lower error is more accurate)



Since submitting the post I have released the algorithm in a plugin with my mastering company and been getting informal feedback from other mastering engineers about how this sounds in use.


Overall the word back has been overwhelmingly positive, with one engineer claiming it to be the “the best sounding plugin EQ on the market to date”. It’s nice know that those long hours staring at decibel error charts have not been in vain.

Are you heading to AES Milan next month? Come up and say hello!


Creative projects in sound design and audio effects

This past semester I taught 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.

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 examples.

From the Sound Design class;

  • Synthesizing THX’s audio trademark, Deep Note. This is a complex sound, ‘a distinctive synthesized crescendo that glissandos from a low rumble to a high pitch’. It was created by the legendary James Moorer, who is responsible for some of the greatest papers ever published in the Journal of the Audio Engineering Society.
  • Recreating the sound of a Space Shuttle launch, with separate components for ‘Air Burning/Lapping’ and ‘Flame Eruption/Flame Exposing’ by generating the sounds of the Combustion chain and the Exhaust chain.
  • A student created a soundscape inspired by the 1968 Romanian play ‘Jonah (A four scenes tragedy)’,  written by Marin Sorescu. Published in 1968, when Romania was ruled by the communist regime. By carefully modulating the volume of filtered noise, she was able to achieve some great synthesis of waves crashing on a shore.
  • One student made a great drum and bass track, manipulating samples and mixing in some of his own recorded sounds. These included a nice ‘thud’ by filtering the sound of a tightened towel, percussive sounds by shaking rice in a plastic container. and the sizzling sound of frying bacon for tape hiss.
  • Synthesizing the sound of a motorbike, including engine startup, gears and driving sound, gear lever click and indicator.
  • A short audio piece to accompany a ghost story, using synthesised and recorded sounds. What I really like is that the student storyboarded it.


  • A train on a stormy day, which had the neat trick of converting a footstep synthesis model into the chugging of a train.
  • The sounds of the London Underground, doors sliding and beeping, bumps and breaks… all fully synthesized.

And from the Digital Audio Effects class;

  • An autotune specifically for bass guitar. We discussed auto-tune and its unusual history previously.
  • Sound wave propagation causes temperature variation, but speed of sound is a function of temperature. Notably, the positive half cycle of a wave (compression) causes an increase in temperature and velocity, while the negative half (rarefaction) causes a decrease in temperature and velocity, turning a sine wave into something like a sawtooth. This effect is only significant in high pressure sound waves. Its also frequency dependent; high frequency components travel faster than low frequency components.
    Mark Daunt created a MIDI instrument as a VST Plug-in that generates sounds based on this shock-wave formation formula. Sliders allow the user to adjust parameters in the formula and use a MIDI keyboard to play tones that express characteristics of the calculated waveforms.

  • Synthesizing applause, a subject which we have discussed here before. The student has been working in this area for another project, but made significant improvements for the assignment, including adding presets for various conditions.
  • A student devised a distortion effect based on waveshaping in the form of a weighted sum of Legendre polynomials. These are interesting functions and her resulting sounds are surprising and pleasing. Its the type of work that could be taken a lot further.
  • One student had a bug in an implementation of a filter. Noticing that it created some interesting sounds, he managed to turn it into a cool original distortion effect.
  • There’s an Octagon-shaped room with strange acoustics here on campus. Using a database of impulse response measurements from the room, one student created a VST plug-in that allows the user to hear how audio sounds for any source and microphone positions. In earlier blog entries, we discussed related topics, acoustic reverberators and anechoic chambers.

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  • Another excellent sounding audio effect was a spectral delay using the phase vocoder, with delays applied differently depending on frequency bin. This created a sound like ‘stars falling from the sky’. Here’s a sine sweep before and after the effect is applied.

There were many other interesting assignments (plucked string effect for piano synthesizer, enhanced chorus effects, inharmonic resonator, an all-in-one plug-in to recreate 80s rock/pop guitar effects…). But this selection really shows both the talent of the students and the possibilities to create new and interesting sounds.

Ten Years of Automatic Mixing


Automatic microphone mixers have been around since 1975. These are devices that lower the levels of microphones that are not in use, thus reducing background noise and preventing acoustic feedback. They’re great for things like conference settings, where there may be many microphones but only a few speakers should be heard at any time.

Over the next three decades, various designs appeared, but it didn’t really grow much from Dan Dugan’s original Dan Dugan’s original concept.

Enter Enrique Perez Gonzalez, a PhD student researcher and experienced sound engineer. On September 11th, 2007, exactly ten years ago from the publication of this blog post, he presented a paper “Automatic Mixing: Live Downmixing Stereo Panner.” With this work, he showed that it may be possible to automate not just fader levels in speech applications, but other tasks and for other applications. Over the course of his PhD research, he proposed methods for autonomous operation of many aspects of the music mixing process; stereo positioning, equalisation, time alignment, polarity correction, feedback prevention, selective masking minimization, etc. He also laid out a framework for further automatic mixing systems.

Enrique established a new field of research, and its been growing ever since. People have used machine learning techniques for automatic mixing, applied auditory neuroscience to the problem, and explored where the boundaries lie between the creative and technical aspects of mixing. Commercial products have arisen based on the concept. And yet all this is still only scratching the surface.

I had the privilege to supervise Enrique and have many anecdotes from that time. I remember Enrique and I going to a talk that Dan Dugan gave at an AES convention panel session and one of us asked Dan about automating other aspects of the mix besides mic levels. He had a puzzled look and basically said that he’d never considered it. It was also interesting to see the hostile reactions from some (but certainly not all) practitioners, which brings up lots of interesting questions about disruptive innovations and the threat of automation.


Next week, Salford University will host the 3rd Workshop on Intelligent Music Production, which also builds on this early research. There, Brecht De Man will present the paper ‘Ten Years of Automatic Mixing’, describing the evolution of the field, the approaches taken, the gaps in our knowledge and what appears to be the most exciting new research directions. Enrique, who is now CTO of Solid State Logic, will also be a panellist at the Workshop.

Here’s a video of one of the early Automatic Mixing demonstrators.

And here’s a list of all the early Automatic Mixing papers.

  • E. Perez Gonzalez and J. D. Reiss, A real-time semi-autonomous audio panning system for music mixing, EURASIP Journal on Advances in Signal Processing, v2010, Article ID 436895, p. 1-10, 2010.
  • Perez-Gonzalez, E. and Reiss, J. D. (2011) Automatic Mixing, in DAFX: Digital Audio Effects, Second Edition (ed U. Zölzer), John Wiley & Sons, Ltd, Chichester, UK. doi: 10.1002/9781119991298. ch13, p. 523-550.
  • E. Perez Gonzalez and J. D. Reiss, “Automatic equalization of multi-channel audio using cross-adaptive methods”, Proceedings of the 127th AES Convention, New York, October 2009
  • E. Perez Gonzalez, J. D. Reiss “Automatic Gain and Fader Control For Live Mixing”, IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA), New Paltz, New York, October 18-21, 2009
  • E. Perez Gonzalez, J. D. Reiss “Determination and correction of individual channel time offsets for signals involved in an audio mixture”, 125th AES Convention, San Francisco, USA, October 2008
  • E. Perez Gonzalez, J. D. Reiss “An automatic maximum gain normalization technique with applications to audio mixing.”, 124th AES Convention, Amsterdam, Netherlands, May 2008
  • E. Perez Gonzalez, J. D. Reiss, “Improved control for selective minimization of masking using interchannel dependency effects”, 11th International Conference on Digital Audio Effects (DAFx), September 2008
  • E. Perez Gonzalez, J. D. Reiss, “Automatic Mixing: Live Downmixing Stereo Panner”, 10th International Conference on Digital Audio Effects (DAFx-07), Bordeaux, France, September 10-15, 2007

What the f*** are DFA faders?

I’ve been meaning to write this blog entry for a while, and I’ve finally gotten around to it. At the 142nd AES Convention, there were two papers that really stood out which weren’t discussed in our convention preview or convention wrap-up. One was about Acoustic Energy Harvesting, which we discussed a few weeks ago, and the other was titled ‘The DFA Fader: Exploring the Power of Suggestion in Loudness The DFA Fader: Exploring the Power of Suggestion in Loudness Judgments.’ When I mentioned this paper to others, their response was always the same, “What’s a DFA Fader?” . Well, the answer is hinted at in the title of this blog entry.

The basic idea is that musicians often give instructions to the sound engineer that he or she can’t or doesn’t want to follow. For instance, a vocalist might say “Turn me up” in a soundcheck, but the sound engineer knows that the vocals are at a nice level already and any more amplification might cause feedback. Sometimes, this sort of thing can be communicated back to the musician in a nice way. But there’s also the fallback option; a fader on the mixing console that “Does F*** All”, aka DFA. The engineer can slide the fader or twiddle an unconnected dial, smile back and say ‘Ok, does this sound a bit better?’.

A couple of companies have had fun with this idea. Funk Logic’s Palindrometer, shown below, is nothing more than a filler for empty rack space. Its an interface that looks like it might do something, but at best, it just flashes some LEDs when one toggles switches and turns the knobs.


RANE have the PI 14 Pseudoacoustic Infector . Its worth checking out the full description, complete with product review and data sheets. I especially like the schematic, copied below.


And in 2014, our own Brecht De Man  released The Wire, a freely available VST and AudioUnit plug-in that emulates a gold-plated, balanced, 100% lossless audio connector.


Anyway, the authors of this paper had the bright idea of doing legitimate subjective evaluation of DFA faders. They didn’t make jokes in the paper, not even to explain the DFA acronym. They took 22 participants and divided them into an 11 person control group and an 11 person test group. In the control group, each subject participated in twenty trials where two identical musical excerpts were presented and the subject had to rate the difference in loudness of vocals between the two excerpts. Only ten excerpts were used, so each pair was used in two trials. In the test group, a sound engineer was present and he made scripted suggestions that he was adjusting the levels in each trial. He could be seen, but participants couldn’t see his hands moving on the console.

Not surprisingly, most trials showed a statistically significant difference between test and control groups, confirming the effectiveness of verbal suggestions associated with the DFA fader. And the authors picked up on an interesting point; results were far more significant for stimuli where vocals were masked by other instruments. This links the work to psychoacoustic studies. Not only is our perception of loudness and timbre influenced by the presence of a masker, but we have a more difficult time judging loudness and hence are more likely to accept the suggestion from an expert.

The authors did an excellent job of critiquing their results. But unfortunately, the full data was not made available with the paper. So we are left with a lot of questions. What were these scripted suggestions? It could make a big difference if the engineer said “I’m going to turn the vocals way up” versus “Let me try something. Does it sound any different now?” And were some participants immune to the suggestions? And because participants couldn’t see a fader being adjusted (interviews with sound engineers had stressed the importance of verbal suggestions), we don’t know how that could influence results.

There is something else that’s very interesting about this. It’s a ‘false experiment’. The whole listening test is a trick since for all participants and in all trials, there was never any loudness differences between the two presented stimuli. So indirectly, it looks at an ‘auditory placebo effect’ that is more fundamental than DFA faders. What were the ratings for loudness differences that participants gave? For the control group especially, did they judge these differences to be small because they trusted their ears, or large because they knew that loudness judging is the nature of the test? Perhaps there is a natural uncertainty in loudness perception regardless of bias. How much weaker does a listener’s judgment become when repeatedly asked to make very subtle choices in a listening test? There’s been some prior work tackling some of these questions, but I think this DFA Faders paper opened up a lot of avenues of interesting research.

sónar innovation challenge 2017: the enhanced dj assistant

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The Audio Engineering team (C4DMwas present in this year’s edition of Sónar+D in Barcelona. Sónar+D is an international conference integrated to Sónar festival that focus on the interdisciplinary approach between creativity and technology.

The Sónar Innovation Challenge (SIC), co-organized by the MTG, <<is an online and on site platform for the creative minds that want to be one step ahead and experiment with the future of technology. It brings together innovative tech companies and creators, collaborating to solve challenges that will lead to disruptive prototypes showcased in Sónar+D.>>

In this year’s challenge, Marco Martínez was part of the enhanced dj assistant by the Music Technology Group at Universitat Pompeu Fabra, which challenged participants to create a user-friendly, visually appealing and musically motivated system that DJs can use to remix music collections in exciting new ways.

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Thus, after nearly one month of online meetings, the challengers and mentors finally met at Sónar, and during 4 days of intensive brain-storming-programming-prototyping at more than 30°C the team came with ATOMIX:

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Visualize, explore and manipulate atoms of sound from
multitrack recordings, enhancing the creative
possibilities for live artists and DJs.

From multitrack recording (stems) and using advanced algorithms and cutting edge technologies in feature extraction, clustering, synthesis and visualisation. It segments a collection of stems into atoms of sound and groups them by timbre similarity. Thus, through concatenative synthesis, ATOMIX allows you to manipulate and exchange atoms of sound in real-time with professional DAW controls, achieving a one-of-a-kind live music exploration.

The project is still in a prototype stage and we hope to hear news of development very soon.

The beginning of stereo

5a9cc9_6da9661bf6bc4c6bbc8d49e310139509 Alan and Doreen Blumlein wedding photo

The sound reproduction systems for the early ‘talkie’ movies  often had only a single loudspeaker. Because of this, the actors all sounded like they were in the same place, regardless of their position on screen.

In 1931, the electronics and sound engineer Alan Blumlein and his wife Doreen went to see a movie where this monaural sound reproduction occured. According to Doreen, as they were leaving the cinema, Alan said to her ‘Do you realise the sound only comes from one person?’  And she replied, ‘Oh does it?’  ‘Yes.’ he said, ‘And I’ve got a way to make it follow the person’.

The genesis of these ideas is uncertain (though it might have been while watching the movie), but he described them to Isaac Shoenberg, managing director at EMI and Alan’s mentor, in the late summer of 1931. Blumlein detailed his stereo technology in the British patent “Improvements in and relating to Sound-transmission, Sound-recording and Sound-reproducing systems,” which was accepted June 14, 1933.