The Sound Tech Hearing Aid

What is this?
My idea of the perfect hearing aid for sound technicians - including standard mixing board functions, custom settings for different situations in everyday life, maximum user control, and a feature to sample noise with the affected ear(s) in mono or stereo.

 An acquaintance of mine recently got a hearing aid and complained about the loudness. Let's face it: many of us, if we did not take care of our ears well enough, may need hearing aids at some point in our lives. So, why not seize the opportunity to make a fun and useful gadget?

In my opinion, the ideal sound tech hearing aid would come with the following features:

1. A decent mic, maybe e. g. a DPA omni capsule or similar.

2. A pad button (e. g. for rock concerts; optional)

3. A gain button

4. A noise gate (to sort of mimic the cocktail party effect; or an EQ preset for better voice recognition, maybe enhanced by AI if the noise is of the same frequency spectrum as the voice(s), plus giving the voice(s) a +6 dB-ish boost. The AI feature could be an optional add-on, like in a lego toolkit)

5. A compressor (e. g. if you are listening to a conversation where a speaker always goes quiet at the end of their phrases) with automatic makeup gain

6. A mute button for when you want to be left in peace

7. An EQ with at least 6 frequencies - for example, give the higher frequencies a mild boost if you lose them as you get older. Also, some people have individual "valleys" in certain regions. You could establish an individual preset in collaboration with a doctor and/or hearing aid tech based on the results of your hearing test, with different settings for each ear. Usually, you would want to go for a clean dB(A) curve (or maybe less bass for when you are not working / listening to music) - taking the Fletcher Munson curves into account (iirc, they mostly look similar at different sound levels, but maybe the best response would require a real-time analyser, and somewhat shifting EQ curves based on the current avg dB level? On second thoughts, this would likely feel weird, as I would prefer one fixed setting - probably best to settle on a compromise).

8. various FX (optional, for fun)

9. A signal splitter for sampling interesting noise where the 2nd pathway goes to an external data storage device (for example: smartphone memory card) and saves the signal in a lossless format (pre or post eq, fx; I would prefer .wav). The data can then easily be extracted into your OS / DAW of choice (Maybe, to prevent abuse of this feature or spying, the hearing aid should feature a red blinking LED, like an earring, whenever it is recording)

It could e. g. be controlled by a smartphone app that is linked to the device via bluetooth and a password.

*It has since been brought to my attention that devices like this exist, but most of the features seem to be automated. I would like to be able to operate them manually, as I may need a different preset when I am at a party where I would like to listen to people vs. when I am at a classical concert and would like to experience a wide dynamic range (no filters, everything flat). Also, sadly, hearing loss can worsen, so adaptations to the settings may have to be made. A sound tech could do this independently between doctor / hearing aid tech appointments.

One more idea:

HRTF simulation with two e. g. hypercardioid mics

Imagine that you have two mics, one per ear. Let's assume that both are facing forward like your eyes, and thus have less reception from behind, similar tothe way human ears work. 

Due to time and loudness (ITD, ILD) differences between the ears, plus the fact that sounds from the front will always sound sharper than from behind, it should be possible for an algorithm to place the sound correctly in a 360 degrees circle (let's start with the 2D, 3D may be more complicated or require a third mic) with the human brain as the center. Once this position is established, there could be a postprocessing filtering curve that corresponds to the average human HRTF for both left and right ear (the curves may differ from ear to ear), thus simulating binaural hearing, making the sound more natural and allowing the user to precisely locate any sound, just as before with the "natural (g)ear". There could be a lowcut for this feature, i. e., it need not be applied below 500 Hz. As the processing requires calculating power, the bar could be set higher, maybe starting from e. g. 1 kHz - this would have to be tested.

Make Dante Controller - Device safety pincodes global

What is this?

A simple idea to improve the safety of Dante infrastructure for small, independent, low-budget, and/or state-funded theatres and other venues like small concert halls.

 Imagine the following situation:

You run sound infrastructure at a theatre, broadcast company or other venue which often hosts guests. Your Dante infrastructure consists of two types of devices:

a) fixed devices with patched connections that should not be touched by anyone (amps, loudspeakers, a Vivace system, wireless mics, a stagebox, a Sennheiser MobileConnect station, etc)

b) devices which other people may want to, or need to, connect to (mixing board, maybe a Mac laptop with a DVS, maybe a video beamer, etc) in order to either stream their content (backing tracks, videos etc) or make a multitrack live recording

How do you make the system as safe as possible while allowing for enough freedom to use it as you and other people see fit, without any limits to creativity?

The expensive solution is the Dante User Manager, which is not affordable for smaller theatre venues.

If you have a Dante Controller, there is an option to lock devices which should not be messed with by typing in a safety pincode. If you want to change any patches on those devices, you first need to type in the pincode, and only if this pincode is correct do you get access to the Dante device.

So far, so good. However, if Dr. Evil tags along, has a Dante Controller installed on his laptop and patches his network cable into any of your Dante switches, he can, again, access ANY of your Dante devices on his own Dante Controller, from his laptop - with the potential to wreak havoc on your entire infrastructure.

There would be an easy workaround:

Make the safety pincodes of locked Dante devices global. If any Dante device is secured with a pincode, no one can access it again without entering said safety pincode. (Perhaps there should be a smart reset function as well, in case people forget passwords, like resetting the pincode by sending an automatically generated reset pincode to the e-mail account registered with the Dante Controller.)

That way, people could patch their connections e. g. to the mixing board, but you would feel safe knowing that they cannot access e. g. the main amps and change the output.

Bonus feature (maybe for a small fee): Enable locking of individual channels, globally. This way, a guest may be able to patch their backing tracks e. g. on a Stereo LR track channel 31 and 32, but not on, say, channel 1 and 2 which maybe goes directly to the Main LR PA.

I believe in open source technology wherever it is accessible, and while I am grateful that Audinate provides free training for their certificates, it would be great to provide a downscaled version of the User Manager for free or a smaller fee which is affordable for smaller venues (like an annual subscription for ~50 USD - maybe, since theatres are public venues, with a sponsoring partnership, i. e. putting the Dante logo in some official theatre publications, or similar).

To make sure that this feature is only used by venues that fit the criteria mentioned above (but e. g. no corporate companies), there could be an application form where the applicants motivate why they would like to have this feature, give an overview about the public value of their house, state funding if existent, and, depending on the extent of the affordable infrastructure provided by Dante, special sponsorship contracts could be established (where the Dante logo is e. g. printed on the tickets; or the venue may be asked to participate e. g. in a Dante promotional video).

Do share this with other people if you think the idea is useful.

The MIDI driving wheel / car pedals drumset

 What is this?

A design for a MIDI controller for car drivers who get stuck in a traffic jam, take a break somewhere, or have some unexpected free time in the car for any other reason.

A car drumset... ?

A friend of mine, Martin P, shared this photo on facebook (source unknown). This led me to the following product idea:

The MIDI driving wheel drum set

Could be produced by: Doepfer et al.

How it would work:

The MIDI controller itself is a cloth with MIDI buttons that can be placed over a driving wheel / over the pedals. Regarding the pedals, light tapping would have to suffice to trigger a sample. No pedal function on the brake, for safety reasons.

This controller would be linked to a set of drum samples which are individually mappable and can be customised / replaced at any time (e. g. for left-handed drummers, custom drum setups people are used to, special cymbals or other special percussion elements that aren't standard).

The audio signal would then be fed to the car's stereo. The MIDI signal could be fed to a MIDI-capable interface to record any performance to a DAW.

There should be a global mute activated whenever any gear and the motor are active, so that playing would only be possible if the engine is turned off or idling. Additionally, incoming phonecalls would need to have priority over the drumset.

As some smart people have pointed out, wheel rotation is an issue. The most convenient workaround could be a bluetooth device to transfer MIDI data to a bluetooth receiver - and from there, MIDI could go to an interface / DAW, and the sound would go to the car speakers, similar to mobile phone calls. Latency could be an issue. However, I am confident that this will get better with newer encoding standards and technology in general improving all the time.

I see a big market here - for people who love drumming but don't bother to build this by themselves.

If you use the idea, please give me credit and show me the product when it is in its final stage.

Thanks.

Universal wireless audio signal codec / data transmission protocol

What is this?

A suggestion to introduce an universal wireless audio signal codec to increase capability between wireless audio systems by different companies and reduce waste / dependence on one system - comparable to MIDI or the EU law to unify mobile phone chargers and charging ports.

For the most part, wireless audio signal transmission applications are provided by two competing companies starting with the letter S.

Each has their benefits and drawbacks. But now, let's have a naive dream.

Make them talk to each other.

Let's have a universal wireless audio signal codec / data transmission protocol, similar to the MIDI standard of synth communication.

There are many situations when this would come in handy. As the market shares seem to be about 50/50, people who use S1 will often bump into S2 users, and the logical thing would thus be to enable both systems to talk to each other.

Examples:

A theatre uses a S1 in-house system. Guest artist who visits brings their S2 headset and system, but wants to have a spare headset provided by S1 as a back-up. To facilitate a quick headset change, it would be smart if the S1 was running the entire time and the S2 could dial in to the S1 system. Granted, thorough frequency management would probably be sufficient, but in times of narrowing frequency bands, more efficient use of available frequencies, by using the same codec / transmission standards if needed, could help make things run smoothly.

A low budget theatre inherits four transmitters from S1 and four from S2, plus a receiver unit from S1. Maybe the S2 unit happened to break down or was sold off. Now, they have a production with seven leading roles who require headsets. It would be beneficial to be able to use all the transmitters with the S1 receiver.

Another theatre has been using S1 gear for a long time, but due to some special features which are only provided by S2, they want to switch to S2. For sustainability reasons, however, they still want to use their older S1 gear in combination with S2. Instead of having to throw everything away, they could use the universal codec mode.

Obviously, this can only be applied to future devices. Those should have

- their individual firmware (either S1 or S2) to ensure downwards compatibility
- the new universal transmission standard
- the ability to send both signals in an encrypted mode
- cross-compatible re-chargeable batteries. It has happened that guests with an S2 system had short-lived batteries with no replacement at hand. If those guests could be helped with long-lived re-chargeable batteries from an S1 system, that would be a big help.

How could this be achieved?

- The new compatibility mode would require both companies (and any other competitors) to send standardised data packages which transmit signal / battery information etc.
- Encryption would have to be standardised, and used.
- An Universal ID allocating / syncing system would be required.
- In compatibility mode, any device needs to recognise and allow "foreign" devices to send or receive signals.
- Ideally, I would envision a universal protocol that can run on different codecs and automatically recognises which codec is used, so that multiple devices can talk to each other. For example, if you are using a state-of-the-art transmitting unit, the Universal Protocol should run on, and recognise, 16-QAM. However, if any of the devices that need to talk to each other has an older / less complex codec, like only using phase shifts, the Universal Protocol should automatically switch to this older standard for the oldschool transmitter, and use an algorithm to avoid any data transmission conflicts.
- Latency should be 3 ms max., as usual.

Obviously, the Universal Mode would limit the range of special functions that could be provided by system-specific applications. The sound may be different, as different encoding systems apparently make a difference. However, I can see thousands of potential situations where compatibility between devices from different companies would be incredibly beneficial. In the age of the climate crisis, any approach to increase longevity of existing systems and use fewer resources / produce less new material is a big plus. Green Technology will be increasingly embraced by Early Adopters and politics, e. g., in the EU, by governmental bodies, and so on - it could be possible to raise substantial funding for this idea.

I do not think that this idea would threaten either S company's sales. Both systems have their advantages, both work reliably and safely, and happy customers will usually stick with one of the two. It will remain best practice not to combine the two (or more) if it can be avoided. However, it would be a shame to miss out on the advantages an Universal Mode could offer. Most importantly, increased flexibility of frequency use at a time when available frequency bands are under constant threat of further narrowing - especially considering the fact that wireless audio signal transmission purposes are classified as Secondary Users in Austria, with Primary Users and their interests having priority over us.

Looking at mobile phone chargers, the EU has issued legislation to standardise chargers and charging slots, which has made enduser life a lot more convenient, yet hardly affected mobile phone sales. A similar form of legislation could require the Universal Protocol to be put into practice. Ideally, I would imagine it to have global compatibility. Similar to the MIDI Association, the basic foundations could be laid out at the next WRC.

To conclude, I hope that some smart brains consider this idea and will put it into practice. My main motivation behind this suggestion is to see more of a Green Event Industry taking forms, to make the AV industry more sustainable wherever possible.

The 'fake acoustic' synth

What is this?

An approach to integrate synthesizers in an orchestral / acoustic instruments setting, plus a few thoughts on universal synthesizer music sheet notation to ensure accurate reproduction of sound which is independent of the synth model (using e. g. MIDI values and mathematical functions).

This is an idea of a specific synth sound setup for e. g. orchestra ensembles in classical music halls. I had this idea while working at Musikverein (as a stagehand, to avoid any confusion) at night once. I was wondering what it would be like to play a piece for synth and orchestra there someday. Some fresh air, so to say. 

This idea has maybe been around and even put into practice in some keyboard models for a while, I guess, but maybe not in good enough quality... and I haven't really seen it in use in top performing orchestras / chamber music ensembles yet. Could be worthwhile to give it a new shot - with better gear.

Standard settings:
Synth to DI to mixing board to PA, end of story.

'Fake acoustic' synth:

Imagine that a synth player wants to perform a modern classical symphony with an orchestra ensemble in a concert hall with marvellous acoustics.

Rather than wiring the synth to a PA, I think it would make more sense to equip the synth with loudspeakers directly on the instrument in such a way that it mimics e. g. a piano, with sound levels being similar to a classical piano. The sound would come from two loudspeakers, L and R, that are installed at either end of what would be the area where the wing of a classical piano is opened and the strings of the piano can be seen (Fig. 1).

Fig. 1. The 'fake acoustic' synth (doodle). Keys, patch section and pitch bend wheel (or whatever you like, drumpad keys, etc. Here, it is just plain chaos, because I couldn't be bothered to draw it realistically. It is just a generic representation. It could also look totally different, too). In the area where a classical piano would have a wing and strings, L and R panning. Signal flow: from keys/controls to a (very fast) FFT analyser (because you can pitch bend...) to ensure correct panning at the loudspeakers, to speakers. For noise samples, maybe positioning at the predominant frequency makes sense, or the panning parameters for complex samples could already be saved in the database, and then played accordingly.

Each key of the synth would be panned at the correct position (but fretless, of course) so that each frequency that it can produce could be heard at the same position as it would on a classical piano covering the same octave range.

Advantages of this setup:

1. The instrument would fit in well with orchestral pieces of work from previous centuries, where PA systems were not known yet. It would feel more 'natural'.

2. Easy localisation of the instrument by other players and the audience.

3. By emitting sound this way, the instrument could use the naturally good acoustics of good concert halls, like the rest of the orchestra does. For example, the Musikverein in Vienna.

4. It could also work better in a 'chamber music' setting with e. g. theremin (why not, while we're at it) or string instruments.

The quality would, of course, strongly depend on the quality of the installed speakers, which could be limited by speaker size. Probably some sort of studio monitors with soft membranes and neutral sound would be best.

Please note that this would only make sense in rooms and concert halls with decent acoustics.

Addendum:

Another way to realise this would be to patch the signal via a DI to one or two active wedges facing the audience, approximately at keyboard/synth height, to fake an acoustic signal. Since this should not be any louder than other orchestra instruments, the SPL exposure of other musicians should be negligible. If there is enough space on stage, this could give a more realistic impression of the sound's origin than classic PA panning.

As a follow-up thought, I was wondering how to write this sort of thing in a sheet music version. As far as I know, no one has really bothered to come up with a standardised notation yet. It would also require a total absence of "colouring" by various synth models, to reproduce everything the same way every time, over the centuries. That might only be possible to do with a software emulating synths, like puredata. Could be one hell of a nightmare to even get started. BUT. If we want synth music to last, it would be worth the effort. Synth models may go out of production, reproductions may be inaccurate. But if people find a workaround (like just writing down mathematical operations, like in programming code), the sound will stay the same over centuries. Might be worth investing more time into thinking about methods of accurately reproducing synth sounds, by introducing a unified sheet music notation system and maybe software standard, just like MIDI facilitates communication and control.

Left handed piano

What is this?

Thoughts and an outline for musical training to increase ambidexterity in playing the piano (e. g. switching melody hand and chord hand, which are often attributed mainly to the right hand / left hand).

All my piano life, I have been learning new songs mostly in the following "setup" (let's call it setup 1):

Left hand: Chords, arpeggios, small movements

Right hand: Melody, improvisations

Very rarely, however, have I encountered songs written the other way round (let's call it setup 2):

Left hand: Melody, improvisations

Right hand: Chords, arpeggios, small movements

Conclusion: The piano, as it is taught in many places in the world, is a concept by a majority of right-handed people.

However, to obtain, practice and maintain full dexterity in both hands, I would suggest that any and all piano lessons should be structured the following way:

Setup 1 - Setup 2 - repeat (50/50 balance)

(Later perhaps: two melodies, like in canons, etc. - this is just a suggestion for early practice)

Of course, you shouldn't break your arms, but it is perfectly possible to play bass lines as individual melodies with the left hand. In jazz songs, for example, I think it would be easy to switch hands because the accompanying chords of the voicings are very free, anyway, so you can rotate them any way you like.

The result of the current way of teaching piano is:

Most people, at a beginner or intermediate level, are much more agile in their right hands.

Only if they practice switching "chord hand" and "melody hand" will they practice "ambidexterity". I think that, especially for styles like jazz, this should give any piano students more flexibility in playing e.g. impros and increase overall piano "fluidity".

TRS - unbalanced stereo input in mixing boards

What is this?

How to avoid conversion of stereo to mono when feeding analog audio signal from a laptop (unbal TRS out) to a mixing board.

Update: Fixed the issue with the help of a 3.5 mm out to double (stereo) XLR converter. At my most recent workplace, we ordered two of those to be prepared. It works fine - the only drawback is that you need two XLR ins for one signal source, which may be an issue when working with a limited amout of XLR ins (smaller mixing boards). 

Whenever possible, I prefer digital communication between devices, as it usually reduces latency while maintaining or improving signal quality. Regarding laptops, the two most common solutions are USB or Dante. For other mixing boards, AES is also a good option.

Imagine the following situation:

There is a spontaneous decision to feed music into a mixing board via a laptop. In this situation, some people do not have a USB interface at hand to route the signal to the mixing board digitally.

What happens instead? They use the headphones out, which classically comes in a TRS configuration - unbalanced L and R out.

Once you feed this into a line in of the mixing board (usually via a 3.5 mm jack to 6.3 mm jack converter), the board combines both channels to mono. I have encountered this exact issue working with an Allen & Heath SQ-16.

As we all know, people can forget adapters... For this reason, it would be nice to have one line in per mixing board for a 3.5 mm jack input from a laptop out. The unbalanced TRS signals would be recognised as two different audio signals. These would then be routed to two separate but paired and (hard)panned audio channels.

Dwarf keyboard for jazz musicians with tiny hands

What is this?

A design of a piano layout for people with smaller hands, like children and women. Downscaling the keys widens the handspan and enables them to reach more notes. With this improvement, it will be easier to play more advanced chords, e. g. with ninths. 

I would like to have a keyboard with

- white keys half the size 
and
- black keys 2/3 the size

of a Novation UltraNova keyboard. It should span 5 octaves and have both MIDI and 6.3 mm balanced L+R jack outputs.

Case in point:

E7b9.

With my tiny hands, I cannot play this on an UltraNova without playing at least one unintended extra key and risking hand damage (see picture). In my opinion, this is yet another case of gender bias in product design.

I am the T-Rex of jazz music

To those who claim that this is doable with more practice: It is not really practical, since I do not have the required anatomy! Everything on a keyboard with dimensions for standard male hands and players is easier if you have big hands and becomes very tedious when you have lady hands. On the Ultranova, my hand spans a ninth without cramps, a tenth is only possible with cramps and will not sound good because I cannot play it with ease. However, if I intend to use all five fingers at once, the maximum interval span gets shorter because I need to crook all my fingers to use all five keys. To be limited by design flaws is not amusing.

I have been told that some Casio models have smaller keys. I have yet to find out if any keyboard according to my wishes exists. If not, I would like to see it made, to make jazz music life for children and people with small hands easier.

Off switch for household intercom speakers or doorbells

What is this?

An installation of an off switch for doorbell speakers, for myself and my neighbour. 

All household intercoms should be made with an off switch for the speakers. Alternatively, a potentiometer for the volume would not only allow one to switch off the speakers at any given time, but also adjust the volume to a lower level suitable for cat and sound tech ears.

I manually installed two off switches for my own and a neighbours' intercom. Nowadays, I usually leave it on when I ordered takeaway food or expect mail delivery, but keep it turned off at all other times. I also want to attach a resistor to decrease the volume to a fixed level, since installing a potentiometer would require adjustments on the motherboard, which may make it more difficult to return to the default settings of the intercom, if I would like to let the flat to someone else someday.

Advantages of an off switch:

- Allows you to sleep in peace after a night shift. Usually, post employees will ring the doorbell in the morning hours, hoping that one of the neighbours of the receiver of a package will be at home. The intercom at my flat has also had periods of malfunctioning and/or there have been weirdos ringing doorbells in the middle of the night, repeatedly.

- Cats who are easily scared by foreigners and doorbells, like mine, have a less stressful life. This is also why I would like to install the resistor.

- No disruptions during rehearsals and/or home studio time.

- The sound is not very appealing, so less of it increases one's overall quality of life.

The technical heart of our intercom. A moment of triumph as I detached the loudspeaker from the wiring. All of a sudden, my quality of life had increased significantly.

Intercom off-switch that I installed for my neighbour. Off-switch kindly provided by my workmate Elmar, who is a genius at playing around with various Raspberry Pi smart home applications that he designed by himself - for watering plants, switching off lights etc.

Plugin to convert (mono and poly) synths with mono out to stereo

What is this?

Some thoughts on how to create a plugin to convert mono synths to stereo, on a solid mathematical basis. 

As someone who enjoys playing old synths, you may have run across this problem before. Your synth only comes with an audio mono out (like the Casio VL-tone).

For monophonic synths, the conversion should be quite easy to write e. g. in Max4Live or puredata etc.

Values that need to be set before use:

1) Highest note of synth (100 % R)

2) Lowest note of synth (0 % R)

These are the endpoints of a linear graph y = k . x on the Oct/Volt system.

If you play a note, an analyser (e.g. FFT) reads the Hz value, which is then fed into a sidechain for the panning. Depending on where you want to hear the synth, you may want to create the illusion that the synth player stands e.g. to the right, so you would use the entire synth note range in 50-100% R of your soundscape. Or, if you want to make the synth appear very broad or only e.g. use one of three octaves, you may want to pan the synth up to 0-300% width, for example.

This (monophonic) solution works for any frequencies within the specified range, which is nice if you have specific, or not well-tempered, tuning settings, like in DAFs Der Räuber und der Prinz.

For polyphonic synths, the conversion gets more complicated. The general settings are the same as previously.

Ideally, you would then need to make a sample dump from your synth onto your DAW of preference or, if the synth only has a 3.5 mm jack out, find e. g. a MIDI instrument that is an exact copy of your synth's sound. In that case, the commands would need to be translated e. g. to MIDI CC messages.

When a chord is played, you would send it to an FFT (this function is available e. g. in puredata). The Hz value is then used for calculating the panning for each note, as above. The amplitude is used to recreate the correct intensity. Each signal triggers the corresponding sound from the sample dump. Latency may be an issue, but could be fixed after playing by moving the sidechain track just a tiny bit forward (which an algorithm would be smart enough to do at the precise position).

The polyphonic solution would be synth-specific and limited to the discrete Hz values of the synth you are playing, unless you recreate the entire s o u n d, e. g. in puredata, without any limitations on which Hz you feed the algorithm.

There may be better solutions if the synth can speak MIDI. However, there are a few synths out there which are polyphonic with an audio-only mono out that are covered by the outlined strategy. Another issue may be that some synths existed before the dawn of MIDI. In this case, sample dumping may provide tedious.

If you would like to start working on this plugin, I won't stop you.

Of course, there are less elegant ways to create an illusion of stereo. For just a few notes, another idea is to record each note separately and then arrange them spatially wherever you like. But there is some joy in working in a 'stilren' way, as the Swedes would say.

Four-stroke engine with organ pipes as an analog sequencer

What is this?

An abstract art concept to generate and modify a rhythm.

This may be a totally impracticable idea, but here goes:

I recently listened to an old Steyr-Puch tractor from the 1930s. It provided a good four stroke metronome. The strokes originate from cylinders. Organ pipes resemble cylinders up to some point, so perhaps it should be possible to run an engine with cylinders that generate notes. Then, as you would in a sequencer, perhaps you could "program" / "set" gradual differences, e.g. in pitch, over time.

Obviously, not the most environmentally friendly idea. Just a nerdy thought for some industrial band out there like Die Krupps...

Adjust EQ settings by drawing EQ curves by hand on a touchpad

What is this?

An idea to quickly adjust EQ curves. 

Currently, many mixing boards work like this:

You manually select some frequencies, then adjust the filters at said frequencies (shelf, notch, dynamic EQing, bell, whatever...). This can be tedious.

To speed things up, if you know more or less how your instrument / mic / ... behaves and what you would like to do, it could be useful to use a pen and a touchpad as input devices, and set a new EQ curve by simply d r a w i n g the entire graph at once. For a bell curve or a shelf, the exact position matters less than for a notch filter. Still, it should be possible to draw a notch filter by hand and finetune the frequency of the absolute minimum later. Since there may be at least 4-ish annoying frequencies you need to take care of, if you know the behaviour of your gear, this could save some time.

Use a balanced XLR out as an unbalanced L/R TRS out

What is this?

An idea to send more signals (risking more noise) from a mixing board with very limited space, by using a flexible configuration. 

The following may be a familiar situation:

You are running out of outputs and e. g. the camera person / streaming person / ... asks you for a main L/R sum out.

One 'dirty' way to achieve this if there is a lack of space on the mixing board would be to be able to switch a balanced XLR out to an unbalanced L/R TRS out, as in old headphones, where T/S is L and R/S is R, for example. It is not a perfect solution, but it could be useful to have, say, four of these as optional extensions in medium budget mixing boards with limited space. For example, 16 outs can become too little pretty fast. 4 switchable outs would result in a total of 12 balanced outs plus 8 unbalanced outs.