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You can ask me questions in the chat during my Studio Streams on Twitch which are almost weekly and free to watch - please mention that you bought the Masterclass 2022 and you have a question: Florian Meindl Twitch Studio Streams



Key Points from chapter 2.2

1. Spread your budget wisely

Find a direction where you want to go with your studio, do you want to do mixing and even mastering? In this case the acoustics and monitor speakers need to be of outstanding quality but also everything in your mastering chain needs to be of equal high quality. If you have one bad device or connection in your whole signal chain you are reducing the overall quality down to the weakest link in the chain.

If you want to only make electronic music and record your synthesizers, then the acoustic treatment can be left out in order to have budget for the synths etc.

2. Try to build things yourself if possible

If you are a talented craftsman, try to build the acoustic elements yourself as well as the furniture.

3. The sound sources before you go into the audio interface are a style decision and therefore don't need to have highest possible audio quality.

If you record a great sounding detail rich poly synth with a cheap audio interface and wrong recording settings you are giving away quality, however if you record a wild lofi sounding and noisy guitar stomp box effect with a great audio interface that's ok, because that's a style choice and even if one layer of your recording sounds lofi it might make sense in the context of a whole song.

4. Take sufficient time to plan your new studio or an upgrade of your existing studio.

  • It's hard enough to plan a new studio but to actually do it will result in several unexpected problems - so consider taking enough time for your project.
  • If you want to go for good acoustic treatment, check the consistency of the room first if it has got hard concrete walls or if there is plasterboard with rockwool already. A room with concrete walls will require much more acoustic treatment.
  • Also the shape of the room is important, make sure it's rectangular and not a square.
  • Check if you need to improve the electricity and if it is safe with a sufficient amount of fuses etc. Companies who install electrics sometimes have long waiting times, never do it yourself if you are not an electrician.
  • Make sure you can get fast internet if you move to a new place. If you are very far outside of civilization you could use a Stark Link Internet which will get you around 100Mb/sec.
  • Find a good listening position, which gives you a well balanced bass sound and place your furniture accordingly if possible.
  • Check if you can be as loud as you want to or if you could get problems with neighbors.
  • Prepare tools you will need, if you are dealing with concrete walls then you might need an impact drill for mounting light or truss etc.
  • In case you build a studio from scratch think of the order your items should arrive: 

1. Tools you might need for the walls and a vacuum cleaner etc.

2. Electric lines

3. Carpet or floor

4. Acoustic Elements

5. Furniture and a light concept

6. Computer, gear, synthesizers


Don't neglect the light concept because you want to have the right mood in your studio!



Key Points from chapter 2.3

Free software to analyze the room response and other important parameters of your studio room: ROOMEQWIZARD (REW)

You can do a measurement before you improve the acoustics but it will not help much, because the room modes can be calculated with the distances between the hard walls (not from plasterboard wall to wall). 

I would recommend to build large bass traps in the corners from top to bottom and as large as you can. Fill them with rockwool (leave it inside the plastic warp, except it's compressed in there) and build a cover construction with wood and textile cover. 

Then reduce the early reflections from the side walls and from the ceiling with mid/high absorbers and then you can do the first measurement and take further action if necessary.


1. The aim is to have as much direct sound as possible and to reduce reflections from the walls and room modes

Direct Sound

2. The RT60 should be around 0.4 seconds in a music studio, for mastering or very small rooms it can be even shorter

For rooms where you primarily play your instruments and synthesizers it's maybe even nice to have a longer reverb time but if you want to mix your songs or master something then you should go for a short reverb time, because you need to judge artificial reverbs which you use in your mix for instance. A dry room also improves the stereo field.

RT60 is the time it takes the sound to go down in sound pressure by 60db compared to the initial sound.

The more constant this time is over the frequency spectrum the better! It's easy to get a short reverb time at 10khz but at 50hz it's difficult and might require large bass absorbers.

3. The frequency response at your listening position should be as flat as possible

This means, that no frequency from 20hz to 20.000hz should be too loud or cancelled out at your listening position in order to judge the volume levels of your mix.

4. Bass problems: Room Modes

Below around 200hz you will have room modes if the studio is not acoustically treated - they are determined by the distances of the hard walls. Room modes are cancellations and exaggerations of sound pressure at certain frequencies and certain points in your room.

You will notice that if you walk around in your room listening to a 50hz test tone. In all corners the sound pressure will be at a maximum and in the room you will find so called nodes (minimum amplitude or sound pressure level) and antinodes (maximum amplitude). On a different frequency, for instance 60hz you will notice, that those points of cancellation and peaks are located somewhere else! 

Bass traps (large velocity absorbers in the corners) will reduce this problem and even out, or flatten the frequency response so that the nodes and antinodes get weaker.

You should choose a listening position, which is in a flat frequency response area in the bass frequency spectrum. This way you don't need extremely large amounts of bass traps.

Place one speaker or your subwoofer in the corner and play music which has different bass notes going on or a test tone which goes from 30hz to 100hz up and down and walk around in the room - wherever the frequencies are kind of equally loud you have the flattest possible natural frequency response.
The rest can be treated further with bass traps.

Room Modes

5. Mid / High frequency problems: Reflections and resulting comb filter effects as well as long reverb times.

Porous absorbers reduce the velocity of the moving air at around 1/4th of the wavelength of the sound wave. If the velocity gets slowed down, the pressure is equally reduced. This way the reflected sound has got lower pressure and therefore less comb filtering effects occur and shorter reverberation times. Which results in more direct sound and a flatter frequency response at your listening position.

Mid frequency absorber


Key Points from chapter 2.4

Electricity can be dangerous so you should talk to an electrician about how to set up your power supply.

Avoid non music devices like heaters or coffee machines in your circuit because they draw a lot of energy and can cause fluctuations in voltage.

Also avoid having too many devices on one power plug, in Germany the maximum is around 3600 watt which is calculated 230 volt times 16 ampere - but I highly recommend to keep significantly lower than that.

If you have a lot of analog gear you might have a hum noise in the audio signal, this is because the ground of the audio signal is connected with the ground of the power signal and if this becomes a loop it acts like an antenna.

Hum Noise

Dirty Power: It depends on the country and if you are in a power grid where machines are switching on or off or if you are near a train station which operates on high voltages and interferes with your grid. 

If you have problems you can use power conditioners, use one for all your computer and hard drives etc., other ones for each section of the studio.

It also catches peaks in the voltage in case of a nearby thunderstorm and older electric grids might let some of those through to your gear but the electricity has to pass many fuses to this is mostly no problem.

For mastering I would recommend good power conditioners because you will have a bit more headroom in the signal - I only have two power conditioners: One for the computer and most attached hard drives and one for the mastering chain.

An UPS (uninterruptible power supply) which is a battery buffered power supply - it can stabilize your power output if you have problems. 

If you want to bring your analog gear to the next level you can also use power leads especially designed for audio gear but that’s the optimization of the last percentage. 



Key Points from chapter 2.5

The monitor speakers are obviously a very important part of your studio and the more you go into mixing and especially mastering, the more important they become because you need to make important judgments and need to hear every detail.

You should definitely choose designated studio monitor speakers and not home hifi speakers for music production because they need to have a flat frequency response, they should have no phase issues between the drivers or distortion. 

Frequency response of EVE Audio SC407:

Frequency Response

Good monitor speakers pay off because you want to hear what you are doing in the most honest way possible. If they give you too little sub bass, your track will maybe have way too much on a large club sound system for instance.

Also the resolution and detail in the mid and high frequency domain will enable you to make better decisions in your music production like how loud the reverb needs to be mixed in.

Before you go for new studio monitor speakers you should check if you can turn them loud without getting problems with neighbors.

If so, then consider small speakers with maximum 6” woofers plus studio headphones where you can check the bass.

Popular models are from Beyerdynamic or AKG and if you have a high budget maybe go for the  Fostex TH-900. If you cannot turn your speakers loud then it’s maybe worth considering an expensive headphone but I recommend to test them and choose one which feels comfortable to wear for hours and which doesn’t play the bass or any frequency too loud - the flater it plays the better you will be able to mix! I would recommend choosing a model with a low impedance (lower than 200 ohm, better around 50 ohm because some headphone amps might be too quiet. Usually the lower the resistance (ohm) the louder they will play.

If loud music is no problem and you even want to simulate club speaker levels I would highly recommend to go for horn loaded speakers in addition to your studio speakers because in Clubs and on festivals mainly horn speakers are used too and they give you that signature sound. I recommend you to only use them to check how mastered tracks sound because it’s hard to produce music on club speakers - they are often too wild and too dynamic

For project studios where you produce and mix your music I recommend using active studio monitor speakers, because a passive construction is only better if it’s designed very well and usually super expensive.

The speakers should have an appropriate size according to your room and how much space you need for other things. I tend to go for larger speakers in order to cover the whole frequency spectrum and have enough headroom for loud peaks. You only need to make sure, that you comply with the minimum distance you should have to the speaker, otherwise phase issues might occur.

I’m a fan of subwoofers because it allows you to have relatively small satellites and on the floor, where the subwoofers stand they are mostly not in the way of something. You also have the ability to place them on different spots and find out where they give you the flattest frequency response. 

A subwoofer also takes away the very low shaking frequencies from your satellite monitor speaker so the heights get a little more defined because the speaker is shaking less.

You need to make sure to have the same distance from the subwoofer to your ear like the satellite speakers, otherwise the bass might come too late. I would say 50cm difference is still ok because that corresponds to 1.4 milliseconds which is no problem in my opinion.

For high end mastering with the best gear and best acoustics only, they tend to use large main monitors, which have the subwoofers in one enclosure with the mid and high drivers. I guess because they want to exactly maintain the distances from the various drivers to the ear.

Make sure the subwoofer is active and has got a similar price range or is from the same brand like your satellite speakers and choose one with a built in crossover so the subwoofer can pass on the signal without deep bass to the satellites.



Key Points from chapter 2.6

There are a few guidelines and recommendations but you will have to try around a little bit in your room.

  1. The focal point needs to be on ear height and the minimum distance to the speaker needs to be adhered to. 
  2. Also Avoid 1/4th of room length, width or height to place the speaker because that’s exactly the area where the woofer can couple to a velocity climax of a standing wave and cause more cancellations and peaks in the low frequency spectrum.
  3. Avoid the same distances from the speaker to the walls and measure from the cone of the woofer.
  4. The speakers should form an equilateral triangle towards the sweet spot where you are going to sit. Make the triangle rather smaller than too large because this will encourage you to mix more excitingly in the stereo field.



Key Points from chapter 2.7


Start with what you have  and switch to affordable office tables or when the budget is high enough: upgraded to a so called session desk in order to have your rack gear in front of you and in a round shape - if you work a lot with keyboards then use a keyboard drawer.


Racks keep your 19” gear in order and give you a tidy look - I prefer ones with fixed holes for the rack screws (you can use longer screws for them too),.

They also come with floating nuts so you can adjust where the gear sits but I prefer the other ones. The disadvantage is, that they don’t allow custom distances between gear and if the thread is worn out it cannot be used anymore.

Maybe you need to leave spaces between gear which gets very hot, then the ones with the floating nuts are better - otherwise you need to leave 1U (height unit) between the gear which is maybe too much.

Make sure to check out rack elements like:

  • Drawer to store little things
  • Power On/Off switches
  • Rack Lamps (The colored ones are nice)
  • Power distributors (if possible incl. USB)


Make sure to have cable channels which can hold enough cables between your racks and the different areas of your studio - it’s simply more safe and looks much more tidy. If you have unbalanced cables don’t run them in parallel to power cables because the magnetic field of the power cables will cause noise and hum in your audio cable.


Speaker Stands

Speaker stands should bring the speaker to ear height when you sit and need to be be stiff and heavy, I would not take those budget metal stands because they don’t fulfil these criteria - rather build your own and fill them with sand or enforce them inside.

Additionally place some isolation pucks, which are usually made of some hard material and rubber, under the speakers - you want to avoid transmitting vibrations from the speaker into the stand and the speaker should also not move - if the speaker is shaking, then the tweeter will not reproduce high frequencies accurately!

I personally placed my speakers on little stands attached to the desk because I want to be able to walk around my studio - however I placed them on isolation stands which decouple it from the desk of course. 

There are even isolation stands called “voice coil stabilizers” which kind of stick to the speakers bottom side with a heavy metal plate and below is some soft decoupling material - this way the tweeter cannot shake and will produce the high frequencies very accurately.


Studio Carpet

Normally music studios have hard and reflective floors - maybe to avoid having the sound too dry and because it’s easier to clean - but I prefer carpets in the studio because I like to have the high frequencies dry and I tend to take my shoes off, so it feels better on the carpet.



Key Points from chapter 2.8

  • I personally have my gear in a circle so the ways are as short as possible
  • Accessibility on backside of gear
  • Light from the floor up
  • Sum areas of your studio together and connect them section by section to avoid chaos




Key Points from chapter 3.2

Keep the signal flow as simple and minimal as possible. Make some decisions about what you want to do and what you don’t need to do.

I found a compromise in my new studio by implementing only my mastering chain into a patchbay and not the insert effects for example - and I decided to use my mixer only as a group mixer and effect device - so I cannot make a mix of something coming from my computer for example.

Example of a common setup:

  1. DAW sends MIDI clock or MIDI notes to synthesizers and drum machines
  2. Synthesizers and other sound sources are sending audio signals (mostly unbalanced) to an analog mixer via patchbay and multicore audio cables
  3. In the mixer send effects are added as well as insert effects
  4. Mixer sends balanced audio signals through bus or group channel insert effects (mostly compressors and equalizers) to the audio interface and DAW records it
  5. Monitor speakers play back the incoming signals from mixer
  6. Audio outputs of audio interface are connected to the analog mixer inputs in order to mix layers coming from the DAW
  7. Final mix is being send from analog mixer to the audio interface (in this case you need to make sure to listen to only the stereo signal which is coming back from the mixer - you can find this setting in the audio interface software or hardware)

More layers can be added step by step 



Key Points from chapter 3.3

  • Line level is for synthesizers and other audio effect devices like compressors
  • Dynamic microphones work on line level
  • Condenser microphones need phantom power (usually 48 volt)
  • If you drive the gain too loud on a line level input you can create pleasant distortion, but be careful with old mixers and devices you could damage the electronics.

Eurorack synths for instance have slightly higher peak level of around 10 volts peak to peak so you can clip a line level input which usually starts clipping at around 3.5 volts peak to peak.

Most hardware synths have stereo outputs because there are stereo effects inside so don’t forget to pan the left and right signal in the mixer!



Key Points from chapter 3.4

Total recall is another big strength of software, for hardware this can be very difficult and I quickly gave it up to be honest.

I personally decided to have only total recall over a few stereo group recordings which I record into the computer, because this makes the mixer signal flow much easier and if the group or bus recordings are done cleverly, you can still do a lot in the DAW when arranging the track.

In order to recall an analog mix you need a mixing console with auto faders or you need to take a large picture with all the settings - but I would rather spend that time on something else.

I only do that with my mastering chain because sometimes I do another version and want to recall the settings - but I try to avoid that of course.



Key Points from chapter 3.5

Balanced / Unbalanced  Cables

Balanced cables have 3 wires inside and one of them is with inverted polarity, this way any noise interference is being cancelled out. Only devices which work with balanced signals will use all 3 wires inside an XRL cable - otherwise it's simply an unbalanced connection.

Cables with three wires inside can not only be used as balanced cables but also as stereo cables, because they can hold the ground and two separate signals.

Also so called “insert cables” have three wires because the two signal cables are used as in and output - but be aware that Y insert cables are unbalanced.


XLR connectors come in two sizes as well as in female and male.

The small size is only for portable microphones so what we use in the studio are the normal large ones. XLR is usually balanced because it has three wires and three connection pins - you can find it mainly on microphones as well as 19” audio gear like compressors and equalizers as well as for active monitor speakers.

Jack / TRS (tip ring sleeve) or TS Tip Sleeve

Jack connectors come in three sizes and in the studio the large ones are used by far the most. The medium size is called “tiny telephone” and is for small patchbays in mixing consoles as well as for some exotic modular synths.

The small size is mainly for stereo headphones and the version with only tip and sleeve is for Eurorack modular synths.

The large jack connectors come as TRS and TS, synthesizers are almost always unbalanced - but balanced cables can still be used, it’s just not balanced then. If the cable is very long out of your synth or guitar then you can use a DI box which generates a balanced signal - if you then plug a balanced cable into its output and connect it with the audio interface or into a device with balanced inputs then there will be no noise interference.

RCA / Phono

There is also phono or RCA for hifi devices and turntables, however the turntable signal is phono level which needs to be pre amplified to line level. That’s why DJ mixers have a designated phono input. RCA can also be used to transmit S/PDIF which is a digital audio format by the way.

Small TS Jacks

For modular synthesizers you have the small TS mono jacks, which also come as stackables so you can take a copy of the signal by simply plugging another cable into the stackables input.

MIDI cables

MIDI cables are also balanced and can therefore be quite long, however the limit is 15 meters. You need to be careful when you go with one midi cable into a device and out of its thru connector and into the next device - if you do this once it’s mostly no problem but if you do this in series a couple of times you might get timing issues.

Optical cables

With optical TOSLINK cables, which are only 1mm thick you can transmit 8 audio channels at 44.1khz if you use the ADAT connection. ADAT is one direction by the way, so if you want to send and receive you need two optical cables.

BNC connector cables

This kind of cable is used to send wordclock from audio interfaces to other AD or DA converters because they need to be in sync with the sampling rate.



Key Points from chapter 3.6

A patchbay is a very simple device but you can use it in multiple ways so it can be a bit complex. 

Decide what the purpose of the patchbay should be and if it's worth it, because you will need more cables in your system. Maybe only use it for certain devices and not for all.

Modes (Purposes)

1. Normal

In normal mode, you can plug in your synthesizers as well as audio interface ins and outs etc. to the rear in and outputs of the patchbay and unless you plug in something at the front output, the signal is "normalized" or connected at the rear. So by pluggin in a cable at the front output of the patchbay you break the normalized connection and you can route it somewhere and bring the signal back at the front input. You can also simply plug a sound source into the front input and it will be connected to the rear output.

2. Half-Normal

The Half-Normal mode is similar to the normalized mode except if you tplug in something at the front output you get a copy of the signal and the original signal is routed to the rear output unless you stick something into the front input. this is helpful if you need a copy of the dry synth signal for instance and you want to record the reverb onto another channel. Or if you want to record a dry guitar signal on channel one and the distorted version on channel two.

3. Thru

The thru mode is useful if you want to use the patchbay as an input and output device like a stage box. You can have your Mixer inputs at the other end of the studio or in another room. 

Some patchaby allow individual switching between the modes per channel - but it is advised to write down the names of all the in and outputs to avoid chaos - most patchbay come with a piece of paper to be sticked onto the patchbay.





Key Points from chapter 3.7

MIDI (=Musical Instruments Digital Interface)

  • 5 Pin DIN MIDI (Normal MIDI cables)
  • MIDI over USB
  • MIDI over WIFI

Each MIDI connection can hold 16 channels, so you can play 16 MIDI devices like synthesizers and drum machines with one MIDI cable. In order to do that you need a MIDI splitter box which has got one input and up to 16 outputs, where you plug in your devices and set the same channel in the synthesizer settings and your sequencer.

However I would recommend to use a MIDI Interface if you have many devices in order to avoid timing issues.

A MIDI channel can transport time clock, notes, pitch bend, filter frequency and many other information like program changes - so you can change the preset of your synth with MIDI (if the synth allows it).

Also MIDI controllers like MASCHINE from Native Instruments for example sends MIDI commands to the software.

Midi Interfaces

Almost every audio interface has got one MIDI in and out, but if you want to do complex MIDI routings you should go for a designated MIDI interface like the Mio XL where you can route all incoming and outgoing MIDI signals in a software.

If you do a lot of analog jam sessions I can recommend the Multiclock, which should be the master clock and has got 4 MIDI outs, which can be started and stopped individually as well as put back and forward in timing and a really cool feature is the shuffle per channel so you can shuffle your 303 for instance!

A Midi Keyboard is essential if you like to play melodies and if you want to quickly hear a sound from your expander-synths or even drummachine (hardware and software), it is usually connected to the audio or MIDI interface, where the MIDI is routed either to a software synthesizer or to your hardware synthesizers via the MIDI output of the interface. 

In this case I would recommend using the USB MIDI. Most modern synthesizers are so called "USB class compliant MIDI devices" and don't even need a driver.


Key Points from chapter 3.8

The purpose of an audio interface is to play back the audio in high quality and give you a balanced output for your monitor speakers as well as headphones.

Depending on the audio inputs it lets you record microphones and line level signals (balanced and unbalanced) in high quality.

Audio interfaces can do even more, depending on their feature set you can send digital audio via word clock or SPDIF to another interface, which converts it to analog audio.

I use ADAT for instance to have 8 more audio inputs at the other end of my studio - all the data is transferred via an optical TOSLINK cable! This way I have only one 1mm thick cable instead of 8 balanced audio cables.

You have to choose a sampling rate. I would recommend 44.1khz for electronic music, which is also the standard for mastered tracks. If you record instruments and voices you can try higher sampling rates - this could slow down your computer though in case you reach the CPU limits easily. 

Choose an interface with enough inputs for what you want to do and see if it comes with the desired connection like USB, Firewire or Thunderbolt.

I would go for a USB connection on the interface because it's the most common and usually sufficient.

For analog mastering you can use an additional digital analog converter with high end specifications to reach best possible quality. Also an external additional clock source can be used to provide the most stable clock for the sampling rate of your converter.




Key Points from chapter 4.1

Make sure to have a diverse mixture of sound sources for the sound you want to make. Try to avoid having only monophonic synths or only polyphonic synths and maybe have some software and some hardware sound sources. The best of both worlds is mostly the key to a healthy sound palette.

Don't use all your budget on an expensive vintage classic except you really want exactly this original sound. Often clones can give you almost the same sound or maybe an even better sound for your purpose.

Use stomp box effects like reverbs, distortion, delays, chorus etc. because they are often cheap but very effective - especially if you use them in a chain. If you clip with a loud signal into a stomp box it often gives you a wild organic sound which is impossible to get from software.

Take maintenance costs of vintage gear into account, which can be very expensive and some parts are impossible to replace because they are simply not existing anymore.

Another thing to consider is, how to connect the different devices together - old vintage gear often doesn't even have MIDI or it has got DIN SYNC MIDI which does not work with normal MIDI clock. So you need converters and everything gets more and more complicated.

If you are really into analog synthesizers you can consider a modular synth but only if you are already familiar with how an analog synth works - otherwise it’s a waste of money in my opinion.

Try to get the best out of the gear you have, even if it's only one synth, because certain limitations will spark your creativity and the really outstanding sounds often come from less-known features and tricks.

The time it takes you to learn another new synth is often better invested in making music with the gear you already have and know. However new gear can spark your creativity too because it will bring fresh possibilities - I recommend you to find a balance here.


Thanks for watching!

You can ask me questions in the chat during my Studio Streams on Twitch which are almost weekly and free to watch - please mention that you bought the Masterclass 2022 and you have a question: Florian Meindl Twitch Studio Streams



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