dear acoustics researchers!

Thank you for all the latest research papers in acoustics and especially the basics of acoustic design for concert halls. I have learned so much about the ambivalance of early reflections, auditory proximity, the critical timing of sound distribution but also amazing things about natural frequency dependent compression in real rooms. Thanks also for making many contributions available as an easy introduction via YT.

However, many of these YT contributions shine through the poorest audio quality imaginable. Recorded in bad acoustic environment (!), poorly miked, badly placed headsets, hissing, booming, humming, dropouts – the whole lot. Resulting in recordings of the lowest audio quality and speech intelligibility. Sometimes so bad that you can hardly follow the content. Seriously, guys, the kids over at Tiktok can do better. So next time please do your homework and walk what you talk, okay?

Sincerely,
Herbert

the world of sound localization according to psychoacoustics

Sound localization refers to the ability of the human auditory system to determine the location of a sound source in space. This is done by analyzing the differences in the arrival time, intensity, and spectral content of the sound waves that reach the two ears. The human ear is able to localize sounds both horizontally (azimuth) and vertically (elevation) in the auditory space.

The brain processes the incoming sound signals from both ears to calculate the interaural time difference (ITD) and interaural level difference (ILD), which are used to determine the location of the sound source. Interaural time difference refers to the difference in the time it takes for a sound wave to reach each ear, while interaural level difference refers to the difference in the level of the sound wave that reaches each ear.

The auditory system uses both ITD and ILD as complementary cues that work together to allow for accurate sound localization in the horizontal plane, aka stereo field. For example, sounds coming from straight ahead might have similar ITDs at both ears but different ILDs, while sounds coming from the side might have similar ILDs at both ears but different ITDs.

It’s also worth noting that the relative importance of ITD and ILD can vary depending on the frequency of the sound. At low frequencies, ITD is the dominant cue for sound localization, while at high frequencies, ILD becomes more important. Research has suggested that the crossover frequency between ILD and ITD cues for human sound localization is around 1.5 kHz to 2.5 kHz, with ITD cues being more useful below this frequency range and ILD cues being more useful above this range.

In addition to ITD and ILD, the auditory system also uses spectral cues, such as the shape of the outer ear and the filtering effects of the head and torso, to determine the location of sounds in the vertical plane and also to identify backside audio events.

The temporal characteristics of an audio event, such as its onset and duration, can have an impact on sound localization as well. Generally speaking, sounds with a more distinct onset, such as a drum hit, are easier to localize than sounds with a more sustained signal, such as white noise. This is because the onset of a sound provides a more salient cue for the auditory system to use in determining the location of the sound source, especially in regards to ITD.

In the case of a drum hit, the sharp onset creates a more pronounced difference in the arrival time and intensity of the sound at the two ears, which makes it easier for the auditory system to use ITD and ILD cues to locate the sound source. In contrast, with a more sustained signal like white noise, the auditory system may have to rely more on spectral cues and reverberation in the environment to determine the location of the sound source.

What will be the next plugin release from “Variety Of Sound”?

I just asked this to ChatGPT and here is the answer:

As an AI language model, I don’t have access to real-time information about upcoming releases from specific companies or developers. However, it’s worth noting that “Variety Of Sound” is a well-known developer of high-quality audio plugins, with a particular focus on analog-style signal processing.

In the past, “Variety Of Sound” has released a range of plugins, including EQs, compressors, tape saturation emulators, and more. These plugins are known for their attention to detail, their intuitive interfaces, and their ability to add warmth and character to digital audio signals.

While I don’t have information about any specific upcoming releases from “Variety Of Sound”, it’s possible that they may continue to expand their range of analog-style plugins, or explore new areas of audio processing. As with any developer, it’s likely that they will release new products based on their own interests and expertise, as well as feedback and requests from their user base.

So, what about “feedback and requests” from your side?

sustaining trends in audio land, 2023 edition

So, the year 2023 is slowly getting underway – time to take another look at the sustaining trends in audio land. Two of the 2022 themes have already been further confirmed and manifested – so let’s take a quick look. A third topic, however, has developed at an incredible speed and in an unbelievable way, to the surprise of all of us. But one thing at a time.

The (over-) saturated audio plugin market

A continuing trend towards market consolidation was to be expected as a result of a constantly oversaturated market, and indeed last year saw a whole series of merger and acquisition activities as well as new alliances. Involved in such activities were brands such as Slate Digital, Sonnox, Focusrite, Brainworx, Plugin Alliance, NI, iZotope and many more.

This is something quite normal in saturated markets and not a bad thing per se, but we might worry about a lack of innovation and diversity as a result. Alongside this, we will continue to see many companies late to the party offering “me too” products and retro brands gilding their HW brands with yesterday’s SW technology. The smarter companies will continue their efforts to successfully establish leading business platforms.

The future of HW DSP as a plugin platform

Since the HW DSP market has not succeeded in creating such a competitive (plug-in) business platform, we are currently witnessing the decline of this domain and in the long run everything will be offered natively. Last year, we’ve seen some late movers also starting such transformations, e.g. UA.

The emergence of AI in audio production

Of course, this was not only predictable but also announced, but no one had ever expected the extent and speed of its emergence over the past year. This applies first and foremost to its appearance in general, but also its impact to the whole music domain in particular. This impact will be immense and dramatic, affecting not only tools and work processes, but also music culture and its economy. The effects will be very, very profound, similar to the way the internet entered all areas of our lives.

The current trend of emulating effect devices with deep learning seems less exciting in this context, as it is just yet another form of effect sampling where we might see little innovation. Much more exciting will be the impact on areas such as composition, mixing and mastering, but also music distribution and value creation in general. But that will be the subject of another detailed article in this Blog.

We live in exciting times.
Stay tuned!

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sustaining trends in audio land, 2022 edition

sustaining trends in audio land, 2021 edition

 

 

ThrillseekerLA mkII released

ThrillseekerLA mkII – bringing mojo back

ThrillseekerLA is an optical stereo compressor optimized for gentle mix bus coloring. It combines smoothest optical compression with vibrant coloration options that deliver a unique box tone in their own right, including thrilling bass and elegant top end void of any harshness in the mids. Its compression not only glues things together effortlessly but also enhances the stereo image by increasing depth and dimension.

10 years after – new in version 2:

• Technical redesign with advanced opto cell emulation
• Simplified gainstaging including automatic output gain compensation
• Streamlined coloring options: Interstage, Tube and Loudness
• New compress/limit option and reworked sidechain filtering

The mkII update is available for Windows VST in 32 and 64bit as freeware. Download your copy here.

Related Links:

• • the beauty of opto-electrical compression – volume 2

the beauty of opto-electrical compression – volume 2

When I was looking for a sophisticated stereo compressor for the outboard studio rack a year ago, I was surprised to see how many of the more interesting models now use opto-electric compression technology. Whether transparent or coloring, tube or solid-state amplifiers, transformer or transformerless, even two-channel layouts in mid/side encoding: far advanced compared to all the classic mono replicas.

Optical compressors are usually characterized by their distinct program-dependent compression behavior, mainly based on a physical memory effect in the detector itself. Other subtle nuances are found across the frequency spectrum that affect timing and curve characteristics, creating a complexity that cannot be reduced to simple two-stage controlled release curves, and which is the beauty of opto-electrical compression in its entirety.

Significant audio signal colorations, however, are shaped not by the gain reduction circuitry but by the make-up gain amplifier, whether it is tube or solid-state. Here, the audio transformer also plays an important role in polishing the transients and creating a cohesive sound.

ThrillseekerLA was designed from the beginning in 2012 as a modern stereo compressor with exciting sound coloring possibilities. It is a compressor with authentic opto-electric control behavior in feed-forward circuit topology.

The upcoming mkII update is a technical redesign dedicated solely to improving the sound. It delivers a unique box tone with thrilling bass and elegant top end void of any harshness in the mids. The compression not only glues everything together effortlessly, but also enhances the stereo image by adding depth and dimension.

The release is scheduled for mid-December.

now that we’ve reinvented 8-bit audio – what’s next?

Gone are the days when choosing the right noise shaping for dithering was a headache and we were still dreaming of Hires Audio as the glorious upcoming consumer format. In the end, it was more important to us to make everything brutally loud. As if it could otherwise be overheard in all the streaming mush: Radio and advertising have led the way, after all, and in the realm of asocial media, the cry for attention is naturally even louder. A pleasant sound, transient-rich and detailed? Forget it! The pitiful remainder of dynamics could now be packaged in 8-bit, lossless. That’s how it looks. The other day I saw a report about audiophiles who still and tirelessly spare neither expense nor effort to optimize the sound at home down to the smallest detail, which seemed unintentionally comical in this context. But it seems just as anachronistic today to spend such an immense technical effort for an inferior production performance. How can this be justified? Surely any stock EQ, a decent multiband compressor and limiter should suffice. In the attached video, the evolution of EDM is outlined in fast forward from the 80s to 2020. Regardless of the qualities of the codecs, this shows very impressively the decline in technical production quality over the years and the side effects of the increase in loudness. Now that we have successfully ruined audio quality, the question remains: What’s next?

the history of Cubase

When Cubase 3.0 came out in 1996 and introduced VST for the first time with all its new and fascinating possibilities, that was the point where I decided to get more involved in music production and set up a small (home) recording studio. VST was the basis for all this and how I imagined a modern (computer based) studio production. What a revolution that was. Watching this video today brings up a lot of nostalgic feelings …

bringing mojo back – volume 2

ThrillseekerVBL is an emulation of a vintage broadcast limiter design that follows the classic Variable-Mu design principles from the early 1950s. These tube-based devices were initially used to prevent audio overloads in broadcast transmission by managing sudden level changes in the audio signal. From today’s perspective, and compared to digital dynamic processors, they appear to be rather slow and can be considered more of a gain structure leveler. However, they still shine when it comes to gain riding in a very musical way – they’ve written warmth and mojo all over it.

ThrillseekerVBL is a modded version that not only features basic gain control, but also gives detailed access to both compression behavior and the characteristic of tube circuit saturation effects. Used in subtle doses, this provides the analog magic we so often miss when working in the digital domain while overdriving the circuit achieves much more drastic musical textures as a creative effect.

ThrillseekerVBL offers an incredibly authentic audio transformer simulation that models not only the typical low-frequency harmonic distortion, but also all the frequency- and load-dependent subtleties that occur in a transformer-coupled tube circuit and that contribute to the typical mojo we know and love from the analog classics.

new in version 2

Conceptually, the mkII version has been refined in that the peak limiting itself is no longer the main task but versatile and musically expressive gain control as well as a thrilling saturation experience. The saturation is now an integral part of the compression and is perfectly suited for processing transient-rich material. Both compression and saturation can be individually activated and controlled.

The circuit-related frequency loss in the highs has been almost eliminated and the brilliance control – originally intended just for compensation – can now also perform exciter-like tasks. The bias control has been extended to shape the harmonic spectrum in much greater detail by allowing the contribution of second order harmonics as well as the adjustment of the saturation behavior in the transient area of the signals. The transformer circuit has also been technically revised not only to resolve all the subtleties realistically but also to reproduce an overall tighter sound image.

ThrillseekerVBL has become a real tonebox, able to reproduce a wide range of tonalities. It provides access to the attack and release behavior and all compression controls can also affect the saturation of the signal, even when the compression function is turned off. This allows specific textures of signal saturation to be realized. As with the good old outboard devices, the desired sound colorations can be achieved just by controlling the working range. And if too much of a good thing is used, the DRY/WET control simply shifts down a gear.

To further improve the user experience some additional UI elements have been added giving more visual feedback. Although oversampling has been added, the actual cpu load was significantly reduced thanks to efficient algorithms and assembler code optimizations.

ThrillseekerVBL mkII will be released October 14th for Windows VST in 32 and 64bit as freeware.

sidechain linking techniques

How an audio compressor responds to stereo content depends largely on how the channel linking is implemented in the sidechain. This has a major influence on how the spatial representation of a stereo signal is preserved or even enhanced. The task of the compressor designer is to decide which technical design is most suitable for a given overall concept and to what extent the user can control the linkage when using the device.

In analog compressor designs, in addition to unlinked “dual mono” operation, one usually finds simple techniques such as summing both stereo channels (corresponding to the center of the stereo signal) or the extraction of the maximum levels of both channels using a comparator circuit implementing the mathematical term max(L,R).

More sophisticated designs improve this by making the linking itself frequency dependent, e.g. by linking the channels only within a certain frequency range. It is also common to adjust the amount of coupling from 0 to 100%, and the API 2500 hardware compressor serves as a good example of such frequency dependent implementation. For the low and mid frequency range, simple summing often works slightly better in terms of good stereo imaging, while for the mid to high frequency range, decoupling to some degree often proves to be a better choice.

The channel coupling can also be considered as RMS (or vector) summing, which can be easily realized by sqrt(L^2+R^2). As an added sugar, this also elegantly solves the rectification problem and results in very consistent gain reduction across the actual level distributions that occur between two channels.

If, on the other hand, one wants to focus attention on correlated and uncorrelated signal components individually (both of which together make up a true stereo signal), then a mid/side decomposition in the sidechain is the ticket: A straight forward max(mid(L,R), side(L,R)) on the already rectified channels L and R is able to respond to any kind of correlated signal not only in a very balanced way but also to enhance its spatial representation.

More advanced techniques usually combine the methods already described.