that unique plate reverb sound

Unlike digital reverberation, the plate reverb is one of the true analog attempts in recreating convincing reverberation build right into a studio device. It is basically an electro-mechanical device containing a plate of steel, transducers and a contact microphone to pickup the induced vibrations from that plate.

The sound is basically determined by the physical properties of the plate and its mechanical damping. Its not about reflecting waves from the plates surface but about the propagation of waves within the plate. While the plate itself has a fixed, regular shaped size and can be seen as a flat (two dimensional) room itself it actually does not produce early reflection patterns as we are used to from real rooms with solid walls. In fact there are no such reflections distinguishable by human hearing. On the other hand there appears to be a rather instant onset and the reverb build-up has a very high modal density already.

Also reverb diffusion appears to be quite unique within the plate. The wave propagation through metal performs different compared to air (e.g. speed/frequency wise) and also the plate itself – being a rather regular shape with a uniform surface and material – defines the sound. This typically results in a very uniform reverb tail although the higher frequencies tend to resonate a little bit more. Also due to the physics and the damping of the plate, we usually do not see hear very long decay times.

All in all, the fast and consistent reverb build up combined with its distinct tonality defines that specific plate reverb sound and explains why it is still so much beloved even after decades. The lack of early reflections can be easily compensated for just by adding some upfront delay lines to improve stereo localization if a mix demands it. The other way around, the plate reverb makes a perfect companion for all kinds of delay effects.

interview series (12) – Daniel Weiss

First of all, congrats on your Technical Grammy Award this year! Daniel, you’ve once started DSP developments during the early days of digital audio. What was the challenge to that time?

Thank you very much, Herbert.

Yes, I started doing digital audio back in 1979 when I joined Studer-Revox. In that year Studer started their digital audio lab with a group of newly employed engineers. At that time there were no DSPs or CPUs with enough power to do audio signal processing. We used multiplier and adder chips from the 74 chip series and/or those large multiplier chips they used in military applications. The “distributed arithmetic” technique we applied. Very efficient, but compared to today’s processors very inflexible.

The main challenges regarding audio applications were:

  • A/D and D/A converters had to be designed with audio in mind.
  • Digital audio storage had to rely on video tape recorders with their problems.
  • Signal processing was hardware coded, i.e. very inflexible.
  • DAWs as we know them today have not been feasible due to the lack of speedy processors and the lack of large harddisks. (The size of the first harddisks started at about 10 MByte…).
  • Lack of any standards. Sampling frequencies, wordlengths and interfaces have not been standardized back then.

Later the TMS32010 DSP from TI became available – a very compromised DSP, hardly useable for pro audio.

And a bit later I was able to use the DSP32 from AT&T, a floating point DSP which changed a lot for digital audio processing.

What makes such a converter design special in regards to audio and was the DSP math as we know it today already in place or was that also something rather emerging to that time?

The A/D and D/A converters back then had the problem that they either were not fast enough to do audio sampling frequencies (like 44.1 kHz) and/or their resolution was not high enough, i.e. not 14 Bits or higher.

There were some A/D and D/A modules available which were able to do digital audio conversion, but those were very expensive. One of the first (I think) audio specific D/A converters was the Philips TDA1540 which is a 14 bit converter but which has a linearity better than 14 bit. So we were able to enhance the TDA1540 by adding an 8 bit converter chip to generate two more bits for a total of about 16bits conversion quality.

The DSP math was the same as it is today – mathematics is still the same, right? And digital signal processing is applied mathematics using the binary numbering system. The implementation of adders and multipliers to some extent differed to today’s approaches, though. The “distributed arithmetic” I mentioned for instance worked with storage registers, shift registers, a lookup table in ROM and an adder / storage register to implement a complete FIR filter. The multiplication was done via the ROM content with the audio data being the addresses of the ROM and the output of the ROM being the result after the multiplication.

An explanation is given here: http://www.ee.iitm.ac.in/vlsi/_media/iep2010/da.pdf

Other variants to do DSP used standard multiplier and adder chips which have been cascaded for higher word-lengths. But the speed of those chips was rather compromised when comparing to today’s processors.

Was there still a need to workaround such word-length and sample rate issues when you designed and manufactured the very first digital audio equipment under your own brand? The DS1 compressor already introduced 96kHz internal processing right from the start, as far as I remember. What were the main reasons for 96kHz processing?

When I started at Studer the sampling frequencies have been all over the place. No standards yet. So we did a universal Sampling Frequency Converter (Studer SFC16) which also had custom built interfaces as those haven’t been standardized either. No AES/EBU for instance.

Later when I started Weiss Engineering the 44.1 and 48 kHz standards had already been established. We then also added 88.2 / 96kHz capabilities to the modular bw102 system, which was what we had before the EQ1, DS1 units. It somehow became fashionable to do high sampling frequencies. There are some advantages to that, such as a higher tolerance to non-linearly treated signals or less severe analog filtering in converters.

The mentioned devices were critically acclaimed not only by mastering engineers over the years. What makes them so special? Is it the transparency or some other distinct design principle? And how to achieve that?

There seems to be a special sound with our devices. I don’t know what exactly the reason is for that. Generally we try to do the units technically as good as possible. I.e. low noise, low distortion, etc.
It seems that this approach helps when it comes to sound quality….
And maybe our algorithms are a bit special. People sometimes think that digital audio is a no brainer – there is that cookbook algorithm I implement and that is it. But in fact digital offers as many variants as analog does. Digital is just a different representation of the signal.

Since distortion is such a delicate matter within the design of a dyncamic processor: Can you share some insights about managing distortion in such a (digital) device?

The dynamic processor is a level controller where the level is set by a signal which is generated out of the audio signal. So it is an amplitude modulator which means that sidebands are generated. The frequency and amplitude of the sidebands depend on the controlling signal and the audio signal. Thus in a worst case it can happen that a sideband frequency lies above half the sampling frequency (the Nyquist frequency) and thus gets mirrored at the Nyquist frequency. This is a bad form of distortion as it is not harmonically related to the audio signal.
This problem can be solved to some extent by rising the sampling frequency (e.g. doubling it) before the dynamic processing is applied, such that the Nyquist frequency is also doubled.

Another problem in dynamics processors is the peak detection. In high frequency peaks the actual peak can be positioned between two consecutive samples and thus get undetected because the processor only sees the actual samples. This problem can be solved to some extent by upsampling the sidechain (where the peak detection takes place) to e.g. 2 or 4 times the audio sampling frequency. This then allows to have kind of a “true peak” measurement.

Your recent move from DSP hardware right into the software plugin domain should not have been that much of a thing. Or was it?

Porting a digital unit to a plug-in version is somewhat simpler compared to the emulation of an analog unit.
But the porting of our EQ1 and DS1 units was still fairly demanding, though. The software of five DSPs and a host processor had to be ported to the computer platform. The Softube company did that for us.

Of course we tried to achieve a 1:1 porting, such that the hardware and the plugin would null perfectly. This is almost the case. There are differences in the floating point format between DSPs and computer, so it is not possible to get absolutely the same – unless one would use fixed point arithmetic; which we do not like to use for the applications at hand.
The plugin versions in addition have more features because the processing power of a computer CPU is much higher than the five (old) DSPs the hardware uses. E.g. the sampling frequency can go up to 192kHz (hardware: 96kHz) and the dynamics EQ can be dynamic in all seven bands (hardware: 4 bands maximum).

Looking into the future of dynamic processing: Do you see anything new on the horizon or just the continuation of recent trends?

We at Weiss Engineering haven’t looked into the dynamics processing world recently. Probably one could do some more intelligent approaches than the current dynamics processors use. Like e.g. look at a whole track and decide on that overview what to do with the levels over time. Also machine learning could help – I guess some people are working in that direction regarding dynamics processing.

From your point of view: Will the loudness race ever come to an end and can we expect a return of more fidelity back into the consumer audio formats?

The streaming platforms help in getting the loudness race to a more bearable level. Playlists across a whole streaming platform should have tracks in them with a similar loudness level for similar genres. If one track sticks out it does not help. Some platforms luckily take measures in that direction.

Daniel, do you use any analog audio equipment at all?

We may have a reputation in digital audio, but we do analog as well. A/D and D/A converters are mostly analog and our A1 preamp has an analog signal path. Plus more analog projects are in the pipeline…

Related Links

interview series (11) – Andreas Eschenwecker

Andy, your Vertigo VSC compressor has already become a modern classic. What has been driven you to create such a device?

I really like VCA compressors. VCA technology gives you a lot of freedom in design and development and the user gets a very flexible tool at the end. I was very unhappy with all VCA compressors on the market around 2000. Those were not very flexible for different applications. These units were working good in one certain setting only. Changing threshold or other parameters was fiddley and so on. But the main point starting the VSC project was the new IC VCA based compressors sounded one dimensional and boxy.

Does this mean your design goal was to have a more transparent sounding device or does the VSC also adds a certain sound but just in a different/better way?

Transparency without sounding clean and artificial. The discrete Vertigo VCAs deliver up to 0,6% THD. Distortion can deliver depth without sounding muddy.

Does this design favour certain harmonics or – the other way around – supresses some unwanted distortions?

The VSC adds a different distortion spectrum depending when increasing input level or adding make-up. The most interesting fact is that most of the distortion and artifacts are created in the release phase of the compressor. The distortion is not created on signal peaks. It’s becoming obvious when the compressor sets back from gainreduction to zero gainreduction. Similar to a reverb swoosh… after the peak that was leveled.

Where does your inspiration comes from for such technical designs?

With my former company I repaired and did measurements on many common classic and sometimes ultra-rare compressors. Some sounded pretty good but were unreliable – some were very intuitive in a studio situation, some not…
At this time I slowly developed an idea what kind of compressor I would like to use in daily use.

From your point of view: To which extend did the compressor design principles changed over the years?

The designs changed a lot. Less discrete parts, less opto compressors (because a lot of essential parts are no longer produced), tube compressors suffer from poor new tube manufacturing and some designers nowadays go more for RMS detection and feed forward topology. With modern components there was no need for a feedback SC arrangement anymore. I think RMS is very common now because of its easy use at the first glance. For most applications I prefer Peak detection.

Having also a VSC software version available: Was it difficult to transfer all that analog experience into the digital domain? What was the challenge?

In my opinion the challenge is to sort out where to focus on. What influence has the input transformer or the output stage? Yes some of course. Indeed most of the work was going into emulating the detection circuit.

Which advantages did you experienced with the digital implementation or do you consider analog to be superior in general?

I am more an analog guy. So I still prefer the hardware. What I like about the digital emulations is that some functions are easy to implement in digital and would cost a fortune in production of the analog unit.

Any plans for the future you might want to share?

At the moment I struggle with component delays. 2021/22 is not the right time for new analog developments. I guess some new digital products come first.

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interview series (10) – Vladislav Goncharov

Vlad, what was your very first DSP plugin development, how did it once started and what was your motivation behind?

My first plugin was simple a audio clipper. But I decided to not release it. So my first public released plugin was Molot compressor. I was a professional software engineer but with zero DSP knowledge (my education was about databases, computer networks and stuff like that). I played a guitar as a hobby, recorded demos at home and one day I found that such thing as audio plugins exist. I was amazed by their amount and also by the fact that there are free plugins too. And I realised that one day I can build something like this myself. I just had to open a DSP book, read a chapter or two and it was enough to start. So my main motivation was curiosity, actually.

Was that Molot compressor concept inspired by some existing devices or a rather plain DSP text book approach?

That days there was a rumour that it’s impossible to make good sounding digital compressor because of aliasing and stuff. I tried to make digital implementation as fluid as possible, without hard yes/no logic believing this is how perfect digital compressor should sound. And the way I implemented the algorithm made the compressor to sound unlike anything I heard before. I didn’t had any existing devices in my head to match and I didn’t watch textbook implementations too. The sound was just how I made it. I did 8 versions of the algorithm trying to make it as usable as possible from user point perspective (for example “harder” knee should sound “harder”, I removed dual-band implementation because it was hard to operate) and the last version of the project was named “comp8”.

Did you maintained that specific sound within Molot when you relaunched it under the TDR joint venture later on? And while we are at it: When and how did that cooperation with Fabien started?

TDR Molot development was started with the same core sound implementation as original Molot had. But next I tried to rework every aspect of the DSP to make it sound better but keep the original feel at the same time. It was very hard but I think I succeeded. I’m very proud of how I integrated feedback mode into TDR Molot for example. About Fabien: He wrote me to discuss faults in my implementation he thought I had (I’m not sure it was Molot or Limiter 6), we also discussed TDR Feedback Compressor he released that days, we argued against each other but what’s strange the next day we both changed our minds and agreed with our opposite opinions. It was like “You were right yesterday. No, I think you were right”. Next there was “KVR Developer Challenge” and Fabien suggested to collaborate and create a product for this competition. That was 2012.

And the Feedback Compressor was the basis for Kotelnikov later on, right?

No, Kotelnikov is 100% different from Feedback Compressor. Fabien tried to make the sound of feedback compressor more controllable and found that the best way to achieve this is just to change the topology to feedforward one. It’s better to say, Feedback Compressor led to Kotelnikov. Also the interesting fact, early version of Kotelnikov had also additional feedback mode but I asked Fabien to remove it because it was the most boring compressor sound I ever heard. I mean if you add more control into feedback circuit, it just ruins the sound.

Must have been a challenge to obtain such a smooth sound from a feed-forward topology. In general, what do you think makes a dynamic processor stand out these days especially but not limited to mastering?

I think, it’s an intelligent control over reactions. For example Kotelnikov has some hidden mechanisms working under the hood, users don’t have access to them but they help to achieve good sound. I don’t think it’s good idea to expose all internal parameters to the user. There must be hidden helpers just doing their job.

I so much agree on that! Do you see any new and specific demand concerning limiting and maximizing purposes? I’m just wondering how the loudness race will continue and if we ever going to see a retro trend towards a more relaxed sound again …

I think even in perfect loudness normalized world most of the music is still consumed in noisy environments. The processing allowing the quietest details to be heard and cut through background noise, to retain the feel of punch and density even at low volumes is in demand these days. Loudness maximizers can do all this stuff but in this context they act like old broadcast processors. In my opinion, the loudness war will continue but it’s not for overall mix loudness anymore but how loud and clear each tiny detail of the mix should be.

Can we have a brief glimpse on what you are currently focused on, DSP development wise?

You may take a look at Tokyo Dawn Labs Facebook posts. We shared a couple of screenshots some time ago. That’s our main project to be released someday. But also we work on a couple of dynamic processors in parallel. We set high mark on the quality of our products so we have to keep it that high and that’s why the development is so slow. We develop for months and months until the product is good enough to be released. That’s why we usually don’t have estimation dates of release.

Related Links

What loudspeakers and audio transformers do have in common

Or: WTF is “group delay”?

Imagine a group of people visiting an exhibition having a guided tour. One might expect that the group reaches the exhibitions exit as a whole but in reality there might be a part of that group just lagging behind a little bit actually (e.g. just taking their time).

Speaking in terms of frequency response within audio systems now, this sort of delay is refered to as “group delay”, measured in seconds. And if parts of the frequency range do not reach a listeners ear within the very same time this group delay is being refered to as not being constant anymore.

A flat frequency response does not tell anything about this phenomena and group delay must always be measured separately. Just for reference, delays above 1-4ms (depending on the actual frequency) can actually be perceived by human hearing.

This always turned out to be a real issue in loudspeaker design in general because certain audio events can not perceived as a single event in time anymore but are spread across a certain window of time. The root cause for this anomaly typically lies in electrical components like frequency splitters, amplifiers or filter circuits in general but also physical loudspeaker construction patterns like bass reflex ports or transmission line designs.

Especially the latter ones actually do change the group delay for the lower frequency department very prominently which can be seen as a design flaw but on the other hand lots of hifi enthusiast actually do like this low end behaviour which is able to deliver a very round and full bass experience even within a quite small speaker design. In such cases, one can measure more than 20ms group delay within the frequency content below 100Hz and I’ve seen plots from real designs featuring 70ms at 40Hz which is huge.

Such speaker designs should be avoided in mixing or mastering situation where precision and accuracy is required. It’s also one of the reasons why we can still find single driver speaker designs as primary or additional monitoring options in the studios around the world. They have a constant group delay by design and do not mess around with some frequency parts while just leaving some others intact.

As mentioned before, also several analog circuit designs are able to distort the constant group delay and we can see very typical low end group delay shifts within audio transformer coupled circuit designs. Interestingly, even mastering engineers are utilizing such devices – whether to be found in a compressor, EQ or tape machine – in their analog mastering chain.

A more realistic look at the Pultec style equalizer designs

One of the few historic audio devices with almost mystical status is the Pultec EQP-1A EQ and a lot of replicas has been made available across the decades. Whether being replicated in soft- or hardware, what can we expect from a more realistic point of view? Lets have a closer look.

Some fancy curves from the original EQP-1A manual
  • In the top most frequency range a shelving filter with 3 pre selected frequencies is offered but just for attenuation. Much more common and usable for todays mixing and mastering duties would be an air band shelving boost option here.
  • Also in the HF department there is just one single peak filter but this time just for boosting. It offers 7 pre selected frequencies between 3 and 16kHz and only here the bandwidth can be adjusted. However, the actual curves could have been steeper for todays mixing duties.
  • There is no option in the mid or low-mid range at all and also no high pass option. Instead, there is a shelving filter for the low-end which allows for boost and/or attenuation around four pre selected frequencies between 20 and 100 Hz.

All in all, this appears to be a rather quirky EQ concept with quite some limitations. On top of that, the low frequency behaviour of the boost and cut filters is rather unpredictable if both filters are engaged simultaneously which is exactly the reason why the original manual basically states “Do not attempt to do this!”.

Nowadays being refered to as the “Pultec Bass Trick” the idea is that you not only boost in some low end area but also create some sort of frequency dip sligthly above to avoid too much of a boost and muddiness in total. In practise, this appears to be rather unpredictable. Dial in a boost at 3 and an attenuation at 5, just as an example: Does this already feature a frequency dip? And if so at which frequency exactly? One has no idea and it even gets worse.

Due to aged electronics or component variety one has to expect that the actual curve behaviour might differ and also to see each vendors replica implementation to be different from another. In practise this indeed holds true and we can see the actual bass frequency dip at a much higher frequency within one model compared to another, just as an example.

… the more I boost the EQ the more it makes me smile …

A reviewers statement misguided by simple loudness increase?

Fun fact: Like the original device, all current (hardware) replica models do not have an output gain control. Also they increase the overall signal level just by getting inserted into the signal path.

So, where is the beef? Its definately not in the curves or the overall concept for sure. Maybe I’ll take some time for a follow-up article and a closer look into the buffer amplifier design to see if all the hype is justified.

Further Links

Not really demystifying but fun to read:

In the VoS plugin line you can find some Pultec style low end performance within NastyVCS: https://varietyofsound.wordpress.com/2010/05/07/nastyvcs-released-today/

Also interesting to read and hear: https://www.sweetwater.com/insync/pultec-shootout-with-sound-samples/

Introduction to SlickEQ and SlickEQ GE by Dan Worrall

tips&tricks with SlickEQ

SlickEQrouting

Note: Some of the tips rely on features from the GE version.

Mixing against HP/LP combo

A good generic practice when EQing several tracks in a mix is too start by dialing in HP/LP combinations by an  appropriate level and then do further EQing/mixing against those settings. Also using the tilt filter is a good idea to apply very first and rough tonal corrections and then working out the details afterwards with the three EQs.

Preserving low-end energy when high-pass filtering

A cool trick to preserve some low-end energy when high-pass filtering is applied is to boost the low-end while using the EQ-SAT feature. As you can see in the routing diagram the HPF comes after the main EQs and EQ-SAT. This way, harmonic overtones are generated based on the fundamentals before the HPF is applied.

Decoupling the low-end

The low-end EQ features a “Phi” option switch which allows to decouple the low-end by an allpass filter network. The crossover can be freely adjusted with the normal frequency control in this band while the gain control does not have any effect in this mode. This may work great for that mellow bass drums just as an example but in other cases it might loose some definition as a trade-off.

Compare different settings

SlickEQ contains two effect settings slots, A and B. Use them in combination with the automatic output gain control to AB test different settings. Within the plugin you can move settings between A and B but also copy&paste is there to freely copy settings between different plug-in instances. Also, undo/redo comes in handy here.

Adjusting precise values

The gain/frequency displays can also be used to enter specific values and also shortcuts are accepted, e.g. “5k” can be entered to set a value to 5000. And did you know that SlickEQ has mouse-wheel support?

 

 

processing with High Dynamic Range (3)

This article explores how some different HDR imaging alike techniques can be adopted right into the audio domain.

The early adopters – game developers

In the lately cross-linked article “Finding Your Way With High Dynamic Range Audio In Wwise” some good overview was given on how the HDR concept was already adopted by some game developers over the recent years. Mixing in-game audio has its very own challenge which is about mixing different arbitrary occurring audio events in real-time when the game is actually played. Opposed to that and when we do mix off-line (as in a typical song production) we do have a static output format and don’t have such issues of course.

So it comes as no surprise, that the game developer approach turned out to be a rather automatic/adaptive in-game mixing system which is capable of gating quieter sources depending on the overall volume of the entire audio plus performing some overall compression and limiting. The “off-line mixing audio engineer” can always do better and if a mix is really too difficult, even the arrangement can be fixed by hand during the mixing stage.

There is some further shortcoming and from my point of view that is the too simplistic and reduced translation from “image brightness” into “audio loudness” which might work to some extend but since the audio loudness race has been emerged we already have a clear proof how utterly bad that can sound at the end. At least, there are way more details and effects to be taken into account to perform better concerning dynamic range perception. [Read more…]

interview series (9) – D.W. Fearn

Doug, when and how did you arrived in the music business?

I have had an interest in electronics ever since I was a kid growing up in the 1950s and 1960s. I built a crystal radio  receiver when I was 8 and my first audio amplifier (tubes, of course) when I was 10. I passed the test for an amateur radio license when I was 12 and that experience of communicating using Morse code was excellent training for  learning to hear. I built a lot of my own radio equipment, and experimented with my own designs.

The high school I attended had an FM broadcast station. Most of the sports and musical events were broadcast, and I learned about recording orchestras, marching bands, choirs, and plays. Friends asked me to record their bands, which was my first experience working with non-classical music.

Another major factor was that my father was a French horn player in the Philadelphia Orchestra. As a kid, I would attend concerts, rehearsals, and sometimes recording sessions and broadcasts. I learned a lot about acoustics by walking around the Academy of Music in Philadelphia during rehearsals.

It would seem logical that my musical exposure and my interest in electronics would combine to make the career in pro audio I have had for over 40 years now.

I was a studio owner for many years before starting the D.W. Fearn manufacturing business, which started in 1993. [Read more…]