Guide: which DAW has the best sound?

None.

All audio programs have the same sound quality

Audio software, given that it simply processes sound in a digital way, doesn’t have any physical limit that hardware may face. Like, for example, the usage of worn out or low quality components.

Digital sound processing is nothing but a huge series of fine mathematical calculations. And computers never miscalculate.

The fact that a sound is digitally processed doesn’t ruin its beauty

For each second of a recorded sound, at least 44100 calculations are made.

Which means that a digitally processed sound gets analyzed 44100 times per second.

With the accuracy of a mathematical calculation.

In other words: perfect accuracy.

The only difference in quality will be determined by which audio card we’re going to use

DAWs let us hear the results of their sound elaborations through the audio card.

We’ve created a guide to help you choose the right one: 14 rules to follow when buying an audio card.

If you want to know the reason behind the realization of these tutorials, you’ll find it here:

Our first post.

Also, on our website you’ll be able to listen to the products of our expertise.

Let us hear from y!

If you have found this post to be useful, share with us your experiences on our socials!
Maybe you could also add a link of what you’ve created, and by using the hashtag #lmkmprod we’ll be able to find all of you.

We’re looking forward to hearing from you!

Guide: the best outboard gear for audio recording.

There’s none.

During an audio recording, sound must not be altered in any kind of way

If you alter the sound while recording, you won’t be able to revert that change.

If, on the other hand, the recording has no alterations, you’ll be able to easily modify it in post production and, for example, delete an edit you didn’t like. Or make it better.

Eventually, you’ll also be able to take on that recording months later and find it completely unaltered, ready to receive new edits thanks to your improved skills.

The only acceptable outboard is a limiter, which is useful in borderline situations to avoid damaging the recorder

The limiter can be set up in such a way that the input audio signal doesn’t exceed the clipping threshold of the recorder.

It is useful in the event that you’re working with inexperienced technicians or in remarkably unstable situations, where you may lose control of the input signal volume.

If you want to know the reason behind the realization of these tutorials, you’ll find it here:

Our first post.

Also, on our website you’ll be able to listen to the products of our expertise.

Let us hear from you!

If you have found this post to be useful, share with us your experiences on our socials!
Maybe you could also add a link of what you’ve created, and by using the hashtag #lmkmprod we’ll be able to find all of you.

We’re looking forward to hearing from you!

Guide: how to recognise the quality of an audio recording

To recognise the real quality of an audio recording we need to make sure to be in a proper listening situation.

Technically speaking: provide ourselves with a reference listening system.

 

A reference listening system is a professional audio system, able to predict if an audio recording is going to sound good when being played by the majority of other audio systems.

A reference listening system will provide us with a quality playback.

Neutral, without any alteration.

(We’ve written a guide in which we explain what these alterations are: What is frequency response?)

Therefore, if somebody else is going to listen to our audio recording in a non-reference listening system, which will alter the recording sound quality, there’s going to be only one sound alteration: the one coming from their own system.

Hence, the name “reference listening system”: because it’s going to guide us during the creation and recognition of market ready audio materials.

(Also, if all of our listeners were inside a recording studio, they’d clearly hear any possible flaw in our audio recording)

To evaluate audio recordings from a non-reference listening systems means that we won’t be sure if it will sound good when played by all the other systems.

If we optimized the audio recording using a non-reference listening system, our following listeners will hear something that’s been fine-tuned in an altered system.

Something that already has an inner flaw.

Which is going to add to the alteration given by their non-reference listening systems.

Our upcoming listeners will hear a recording that’s been altered twice.

In other words: a garbage recording.

If you don’t have a reference listening system, you have nothing else to do but trust somebody who owns one.

Music production and recording studios invest time and money in the optimization of their listening systems to provide a crucial quality check service for audio recordings.

We’ve created tutorials on how to create reference listening systems.

In these guides we’ll explain how build a reference listening system, that will allow you to recognise and even create quality audio material.

If you want to know the reason behind the realization of these tutorials, you’ll find it here:

Our first post.

Also, on our website you’ll be able to listen to the products of our expertise.

Let us hear from you!

If you have found this post to be useful, share with us your experiences on our socials!
Maybe you could also add a link of what you’ve created, and by using the hashtag #lmkmprod we’ll be able to find all of you.

We’re looking forward to hearing from you!

Tutorial: how to create a good reverb

First of all, to create a good reverb, we need to understand what it is composed of:

  • Phase 1: early reflections
  • Phase 2: late reflections

Early reflections are the most complex part of a reverb: they provide us specific information about room size and sound position

Unfortunately, they’re also the hardest thing to set up about a reverb.

As the name suggests, they’re the early reflections caused by reverberation. They’re extremely swift: usually, they happen within 100 milliseconds after the sound is generated.

Due to their swiftness, they’re going to be perceived as part of the sound itself.

Late reflections result from the gradual energy loss of sound waves, reflection after reflection

This part is due to reverberation weakening. They’re called “late reflections” because, unlike early ones, they’re perceived as disconnected from the initial sound.

Depending on the material of the location in which the reverberation occurs, late reflections are going to comprise different frequencies.

Location size influences late reflections duration: the greater its size, the longer the duration will be.

Usually, modern devices are able to create quite realistic late reflections.

Let’s now see how to properly set up a reverberation unit.

For convenience, we’re going to refer to the reverberation space as “room”: reverberation units use the same terminology.

Mind that reverb is one of the most complex features of a sound: make sure that you have a great listening system before starting to edit one.

How to properly set up early reflections

First of all, lower to zero the volume of late reflections.
(Warning: the cheapest reverberation devices don’t allow for this. In such a case, you’ll have to find a work around).

Changing the size of the room: the more you increase early reflections delay (in milliseconds), the bigger the room will seem. Increase or lower the delay until it feels just right.

Changing the sound position: in the most sophisticated reverbs, you’ll be able to set up different millisecond values per channel. The more you lower the delay of a channel with respect to another, the more the sound will seem to come from that direction.

Protip: many reverberation devices tend to create early reflections with a very low volume. Do keep this in mind, in the event that you aren’t able to properly hear them.

Make sure that your reverb is creating realistic early reflections: unfortunately, many reverberation devices aren’t able to do that.
If you feel like, despite carefully choosing the milliseconds of delay, your reverb doesn’t sound realistic, there’s nothing else to do but change your device.
(Or keep a mediocre one).

How to properly set up late reflections

Leaving early reflections at their normal volume, you have two features you can act upon:
Changing the consistency of room walls: the thicker the wall material, the more they will be able to reflect frequencies.
This feature is usually managed by a low-pass filter (LPF) and a high-pass filter (HPF).
The more you rise the LPF cutoff frequency, the more you will feel like being in a room with metallic walls.
The more you rise the HPF cutoff frequency, the more you will feel like being in a room that absorbs low frequencies (this case is quite peculiar. E.g., a room that’s undergone acoustic treatments).

Changing the size of the room: the longer the late reflection duration, the bigger the room will seem. Increase or lower the duration (in seconds) until it feels just right.

If you want to know the reason behind the realization of these tutorials, you’ll find it here:

Our first post.

Also, on our website you’ll be able to listen to the products of our expertise.

Let us hear from y!

If you have found this post to be useful, share with us your experiences on our socials!
Maybe you could also add a link of what you’ve created, and by using the hashtag #lmkmprod we’ll be able to find all of you.

We’re looking forward to hearing from you!

 

What is frequency response?

Know your instruments.

Frequency response is the way audio equipment changes the frequencies we send to it.

In other words: it is the way its related instrument is going to “sound”; the way said instrument is going to alter the frequencies it receives.

Be aware: an audio signal is not characterized by frequencies alone, but by phases too. Which is going to be drawn into a different graph, called phase response curve.

It is a fundamental feature of sound. But, in case of music production, given the quality of modern equipment, it is usually unimportant.

The same concept doesn’t hold true in case of live audio systems, in which phase plays a crucial role due to the strong environmental interactions involved. Like wind, humidity, and the huge size of arenas, for example.

If you want to delve deeper in this topic, at this link you will find more information about it.

The most used method to illustrate a frequency response is the frequency response chart.

frequency-chart empty

On the vertical axis, the graph will show how the instrument is going to alter frequencies. That is, the dB value of its chromatic variation (“chromatic” in that the frequential components of a sound are going to define its “color”. In other words, its timbral capacities).

On the horizontal axis, the graph will show which frequency is altered.

The one above was an example of an empty graph.

Now, let’s see a frequency response curve within the graph.

Grafico risposta in frequenza

Frequency response chart.

The curve inside this graph is the frequency response.

Understanding this graph is very easy.

For example, we can infer that this instrument is going to subtract 20 dB to a 30 Hz soundwave, with respect to the original sound. For a 40 Hz soundwave it will subtract 10 dB. For a 300 Hz soundwave it will subtract about 3 dB.

Speaking about this example only, we’re debating about an instrument that’s quite reliable for sounds above 100 Hz, in that a variation of +/- 2 dB is considered to be acceptable for professional standards.

Let’s now see the graph of a remarkably reliable instrument.

Risposta in frequenza eccellente

Excellent frequency response.

Apart from a peak at 10 kHz, where 10 dB are added (something that can be easily corrected in post production, if needed), this instrument will not alter the original signal at all.

An instrument with such a linear graph is a reliable one.

In fact, it is the frequency response of one of our favourite microphones during orchestral recordings.

Let’s now see the graph of an average instrument.

Risposta in frequenza nella mediaq

Risposta in frequenza nella media

This instrument is not reliable below 100 Hz, and it shows some anomalies around 7 kHz that may result quite annoying.

But all in all, it’s a reliable instrument in situations where a complete tonal fidelity is not required. Like, for example, noisy live shows.

In fact, it is the frequency response of a quite common dynamic microphone.

Let’s now see a graph of an awful instrument.

Grafico risposta in frequenza pessima

Grafico risposta in frequenza pessima

If you see an instrument with a frequency response that shows so many pronounced anomalies, leave it on the shelf: it is not a reliable instrument.

That is, unless you didn’t come up with a terribly eccentric idea for your recordings.

Additionally, if you see graphs that show multiple curves for a single instrument, like this one for example

Effetto prossimità.

Effetto prossimità.

don’t worry: it’s nothing but the proximity effect.

Some kinds of microphones change their behaviour depending on their distance from the sound source being recorded.

These graphs show how the instruments will react depending on this distance, displaying different curves for different distances.

As odd as it may seem, this system will allow you to understand how an instrument is going to sound without ever listening to it at all.

Except for some uncommon exceptions, that we will cover in other articles.

If you want to know why this tutorial was made, you’ll find out more in this post:

Our first post.

And you got our entire website to hear if we’re talking about something that we can do.

We want to hear about you!

If you found this post useful, please: share your experience with us on our social pages!
Maybe together with a link to what you’ve created, and using our official hashtag #lmkmprod to let us find you all.

We’re looking forward to hear about you!

What is HRTF? (Brief explanation)

Let’s get to know psychoacoustics.

HRTF stands for Head Related Transfer Function.

In other words: the phase and frequency response of our head. In fact, a transfer function is a particular mathematical formula that groups both data sets.

Basically, it is the way in which our head changes the sounds that reach our eardrums.

These changes are dictated by the structure of our head: nose, forehead, mouth, hair, bone density, auricles… every feature of us that the sound hits before reaching our eardrums. And, in the event that the sound is coming from below, our shoulders too.

Every struck “obstacle” is going to faintly change the sound, altering frequencies and phases.

Depending on where the sound comes from (in front, behind, above, below), is going to encounter different obstacles. And different acoustic alterations.

Our brain has finely memorized these peculiarities, and it takes advantage of them to understand which direction the sound is coming from.

That is the reason for which, even with our eyes closed, we can still understand the position of a sound source.

The organ that influences these alterations the most is auricles: all their twists are needed to extensively characterize the auditory changes, by having the latter “clash” onto them.

(You’ve finally understood why ears have such a “weird” shape, instead of simply being flat.)

Here’s an example of front (continuous line) and back (dotted line) HRTF.

HRTF frontale e posteriore

If we apply the continuous line frequency response to a signal, our brain will understand that the source of the sound is in front of us.

If we apply the dotted line frequency response to a signal, our brain will understand that the source of the sound is behind us.

Needless to say that everyone of us has it’s own physical structure, and for this reason the HRTFs are never going to be identical. However, there’s a slight resemblance between all the HRTFs that allows our brain to effectively interpret signals that have been elaborated with other people’s HRTFs.

Moreover, are you aware that there are systems to record in HRTF, so that our recording isn’t just going to identify left and right (monodimensional), but also above, below and to the sides (tridimensional or binaural)?

Here’s one of the most common ones: a dummy head.

 

Dummy head (Neumann KU100)

Dummy head (Neumann KU100)

 

That is, a tool that simulates the shape of a human head, with microphones instead of eardrums. The attempt is to effectively record the HRTF information.

And it works pretty well: listening to a recording made with this tool feels like being on stage.

Software-wise, there are HRTF decoders that allow HRFT data to be linked to a signal, therefore giving it 3D spatiality.

Another interesting implementation of HRTF is to be found in almost every modern headphones: to try and avoid the “sound inside the head” effect, a frontal HRFT impression is imprinted into the headphones.

Which is also the reason that headphones frequency responses can’t be interpreted “with the naked eye”.

By the way, here’s another example of a professional pair of headphones.

http://www.headphone.com/

http://www.headphone.com/

Even if the curve is quite irregular, the auditory result is going to be reliable anyway, given that the anomalies are due to a particular frontal HRTF impression.

Actually. not exactly frontal, since the typical position of two speakers is a the vertexes of an equilateral triangle, with our head corresponding to the third vertex.

If you want to know why this tutorial was made, you’ll find out more in this post:

Our first post.

And you got our entire website to hear if we’re talking about something that we can do.

We want to hear about you!

If you found this post useful, please: share your experience with us on our social pages!
Maybe together with a link to what you’ve created, and using our official hashtag #lmkmprod to let us find you all.

We’re looking forward to hear about you!

 

8 steps to explain what “sound” is.

What is a sound?

  • A sound is a physical phenomenon

    An acoustic one, to be exact.

    In fact, if you place yourself in front of a powerful audio system, your body will start to vibrate, struck by the acoustic vibration.

    It becomes music when it is used for artistic purposes.

  • A sound is a vibration.

    And it gets detected by our ears.

    It is measured using two methods:

    • Vibrational frequency, in Hertz units (Hz)
      It will tell us the tune of said sound.
    • Vibrational intensity, in deciBel units (dB)
      It will tell us its volume.
  • Sounds are composed of frequencies. Frequencies have a “phase”.

    • Sounds, except for a specific one called “sinusoid” (also known as “pure sound”), are never made up of one single vibration.
    • Each vibration that makes up a sound is called “frequency”, and it has its own phase.
    • The sum of all these frequencies is called “spectrum”.

    In fact, spectrometry gets its name from the fact that it measures the spectrum of a sound, giving information about which frequencies compose it.

  • A vibration can be produced in many ways.

    For example, by the strings of a guitar. O by a drum struck with a drumstick. Or by a pipe of an organ-pipe, which vibrates thanks to a strong air stream, created by a compressor, blowing through it.

    Or even by a loudspeaker crossed by an electrical signal, like a synth one, for example, which vibrates according to the signal it receives.

  • Next, vibrations are transmitted through a medium.

    Through air, for starters: the body that generated the vibration hits the air molecules around it, which in turn hit other close-by particles, therefore producing a domino effect that brings the vibration to our ears.

  • Each transmission medium has its own characteristics

    Water, for example, transmits sounds much better than air does. In fact, when we’re underwater, everything seems much closer than it really is.

    However, we are unable to hear as well as when we’re outside the water mostly because water alters the functioning of our eardrums (and, if we dive too deep, it actually damages them).

    Walls are another example: they’re terrible acoustic conductors, as they greatly lower a vibration intensity (thereby softening sounds) and they suppress higher frequencies (making them more grave: the typical muffled sound that we may hear when standing outside a club.)

  • Sounds have no misteries, and can be reproduced.

    From our analysis, it is possible to understand that no sound has any mystery: every sound is measurable. If it’s measurable, we can analyze it.

    And if we can analyze it we can understand what it is made of, and therefore recreate it to our own liking.

    After all, a sound is nothing but a sum of frequencies. To reproduce it, it is sufficient to add all the frequencies it was made of. (Yes, this is the principle upon which the well-known additive synthesis lays its basis. We’ll write about it in another article.)

If you want to know why this tutorial was made, you’ll find out more in this post:

Our first post.

And you got our entire website to hear if we’re talking about something that we can do.

We want to hear about you!

If you found this post useful, please: share your experience with us on our social pages!
Maybe together with a link to what you’ve created, and using our official hashtag #lmkmprod to let us find you all.

We’re looking forward to hear about you!

5 advices to improve your room acoustics

Simple tips to fix the acoustic problems of your room.

In this tutorial we’re going to explain how to fix the most simple acoustic problems a room may have.

How to improve the acoustic quality of your room.

Almost every sound improvement technique moves towards a goal: to eliminate reflections.

Reflections are nothing but altered versions of the initial signal, that are going to mix up the final result, eventually ruining the audio quality.

Standing waves are another major problem, but they are way too complex to explain for a basic tutorial.

  • Don’t let any corner to be empty

    From an acoustic point of view, corners are quite peculiar: if they’re empty, they may cause troubles.

    You should try to cover them, for example using medium density mineral wool panels (do not choose them with density above 100 kg/m3).

  • Don’t have bare walls

    Bare walls create reflections.

    There are many ways you can cover them: fabric, composite mineral wool panels (one panel with density not above 50 kg/m3 along with another underlying panel with density between 70 & 80 kg/m3), foam rubber…

    Your goal is to prevent frequencies to bounce between the walls.

    Keep in mind that, the higher the frequencies, the lower the density of the cover material should be. We will show you in another article how to roughly calculate this ratio.

    Don’t overdo, otherwise you’ll have an excessively dry room. Usually, it is advised to leave at least one wall without any treatment.

  • Don’t have an empty ceiling

    As well as walls, even the ceiling can create reflections.

    Treatment methods are the same as wall ones.

  • Don’t let the room to be empty.

    Couches, armchairs, bookshelves… Anything that fills the room is able to improve its acoustics, dampening the sound waves and preventing them to frantically bounce off the walls, creating reflections.

    As always, don’t overdo, otherwise you’ll have no room to move around.

  • Use scientific methods to analyze the room, so as to understand what the issues are

    Do not rely on your hearing alone: to properly understand which frequencies are causing troubles in your room, you should make use of scientific devices for your measurements.

    For the brave ones, we’ll publish an article in which we’ll explain how to carry out a basic room analysis.

 

This tutorial is a brief guide aimed at solving some minor sound problems: in case of serious sound issues, the situation becomes extremely difficult, and the treatments required happen to be very expensive and intricate.

In the event that this tutorial is not sufficient to help you achieve a suitable room acoustics, our suggestion is to get in touch with an acoustic treatment specialist.

If you want to know the reason behind the realization of these tutorials, you’ll find it here:

Our first post.

Also, on our website you’ll be able to listen to the products of our expertise.

Let us hear from you!

If you have found this post to be useful, share with us your experiences on our socials!

Maybe you could also add a link of what you’ve created, and by using the hashtag #lmkmprod we’ll be able to find all of you.

We’re looking forward to hearing from you!

Tutorial: 9 steps to place well your studio monitors

10 minutes to properly place your studio monitors.

In this tutorial we’re going to give you 9 simple tips to accurately place your studio monitors.

This is a tutorial for near field monitors.

Premise: you’re going to need a 2 meters long rope/wire, a laser level and a microphone stand.

Any laser level will do the job: you can one for little money in any tool shop.

Like this one, for example.

www.bricoman.it

www.bricoman.it

  • Choose a room with the features listed below.

    The more of these features it has, the better your studio will be

    • Thick walls
    • Walls composed of dense material
    • Uneven walls (for example, natural stone ones)
    • No bare walls (wardrobes, bookshelves, armchairs, couches)
    • No bare corners (corners are, from an acoustic point of view, quite peculiar)
    • Far from bedrooms / highly populated areas
    • Far from noisy public transport systems (underground, tram, train, rocket launch base…)
    • No noisy neighbors
    • Dry
    • Not excessively high or low temperatures
  • Choose and place the workbench

    You should favour a workbench that’s solid and full (that is, without room for legs): they have a better acoustic output.

    Place its rear close to the widest wall, leaving 30-40 cm of empty space between the bench and the wall itself.

  • Place the monitors in line with the workbench edge

    Anything you put below the monitors is going to create reflections. And reflectioins alter the audio signal.

    Try to keep the space in front of the monitors as clear as possible, but there’s no need to overdo: a keyboard, for example, is not going to cause any harm.

    And yes, you got it right: placing the speakers behind a huge mixer with a gazillion of faders is not a good idea.

    Try to put a notebook below your chin while you’re speaking, and listen to the difference. And remember that the human voice has much less frequencies than an audio signal.

    Giusta posizione casse bordo.

    Correct monitor position.

  • Place the monitors so that their tweeters are at the same height as your ears.

    High pitched sounds are extremely directional: to hear them well you need to point them exactly at your ear height.

  • Set up the triangle

    Place the speakers so that they create, more or less, an equilateral triangle with your head as one of the vertexes.

    A rapid method is to use your arms as guides: open them out so as to form an equilateral triangle, having the speakers in front of the palm of your hands.

  • Refining the triangle

    Vertically extend the microphone stand, and place it where your head is going to be.

    Take the wire, measure the distance between the stand and the tweeter and, holding your finger on the rope marking the measured distance, use it to equate the distance between these points:

    • Tweeter 1 – Tweeter 2
    • Tweeter 1 – Stand
    • Tweeter 2 – Stand

    This way, you will obtain a perfect equilateral triangle.

    Like the one in this picture.

    Monitor da studio posizionati correttamente.

    Properly placed studio monitors.

  • Laser pointing.

    Place the laser level upon the monitor, exactly above the tweeter, turn the speaker until the tweeter is aligned with the microphone stand.

    Of course, don’t move around neither the stand nor the monitors: turn the speakers so that their tweeters are pointing towards the stand.

  • Success.

    Pop a bottle of champagne, turn up the volume, and play Lateralus by Tool in your now perfectly positioned system.

At this point, your system is going to be flawlessly positioned.

In other tutorials we will explain how to acoustically prepare the room in which you placed your speakers and how to solve possible sound problems that may ruin the reliability of your audio monitors.

If you want to know why this tutorial was made, you’ll find out more in this post:

Our first post.

And you got our entire website to hear if we’re talking about something that we can do.

We want to hear about you!

If you found this post useful, please: share your experience with us on our social pages!
Maybe together with a link to what you’ve created, and using our official hashtag #lmkmprod to let us find you all.

We’re looking forward to hear about you!

8 rules to choose the right audio monitors

Smart tips to choose the right equipment for your passion.

Another crucial step when creating music in a professional manner is the purchase of a pair of studio monitors.

Unfortunately, studio monitors are quite a complex piece of equipment, and in most cases it won’t be possible to take advantage of all its potentiality.

This guide is just a basic tutorial on how to choose your first studio monitors: for the ultimate audio setup, it is necessary for sound professionals to step in, as well as a good amount of money to spend in acoustic treatments.

  • Near field? Mid field? Far field?

    Absolutely near field.
    Mid field and far field monitors are extremely expensive and, even when affortable, they’re pretty hard to use for a basic setup like a home/project studio.
    Moreover, if you ever have to deal with monitors of such quality, there will be other people to set them up for you.

  • Room size.

    The size of your room is going to be crucial for a good sound quality: the bigger the monitor, the larger the room should be, otherwise they will not operate properly.

    If your room size is higher than about 4x4x2.5 m, you’ll be able to afford 8” monitors. A room that’s 5x5x3 m in size is perfect for 8” monitors.

    If you’re unlucky and do not have access to such a big room, you should purchase 5” monitors.

    If the room is much bigger than 4x4x2.5 m, you may have reverb issues.

  • 5” vs 8”

    5” monitors are the most convenient ones, but they won’t allow you to work on low frequencies. They’re meant to be used along with a subwoofer.

    8” ones are more expensive, but with them you’ll be able to work on low frequencies as well. They also have a good quality / price ratio.

    If you’re seriously looking to work with sound and music, our suggestion is to buy an 8” monitor.

    A 5” monitor + subwoofer is going to be much more expensive than an 8” monitor alone, and it’s also going to be more difficult to setup.

  • Multiple listenings?

    Useless.
    If you own good monitors, all you have to do is filter the input signal with an equalizer: you’ll achieve the same frequency response of whatever loudspeaker you desire.

    A single pair of great studio monitors is the best choice.

  • Which brand?

    It’s not that important.
    Brand by itself is not going to give much information about the quality of your monitors.
    There are monitors made by unknown brands that sound extremely good, and popular brand ones that sound just… “meh”.

    Speaking of reliability: almost all current monitors are, if properly handled, extremely reliable.

    The real method to understand what the best monitor is, is described in the following step.

  • Frequency response

    A monitor frequency response indicates its audio quality.
    The flatter it is, the better your monitor will reproduce the sounds you’re transmitting to it.
    The choice, in this case, is fairly simple: compare the different frequency responses of the monitors you’re browsing, and pick one with the flattest curve.

    If you’re lucky enough to be able to check it, this is the fastest and most reliable way to verify the quality of a monitor.

    However, if this information is not available, in the following two steps we’ll suggest you other methods to test your new monitors.

  • Music test

    If you really want to put a system under pressure, the best way is to have it reproduce Metal or orchestral music.

    In fact, these kinds of music have large dynamic and timbric excursion (they range from silence to massive volumes, and their sounds are extremely colorful).

    Minimal music pieces such as Blues, Jazz or even solo instrument ones are inadvisable to test sound monitors.
    They’re never going to put enough pressure on the timbric capacity of your new monitors.
    Moreover, given their low timbre (they do not have particularly colorful sounds), they will make the system sound better than it really would.
    This is because, given their low timbric demand, they will (relatively speaking) deliver the whole musical message, instilling in you the emotions they are supposed to.

    Hence, it’s time to pull out Brahms and Periphery.
    Or even LMK solo record: https://open.spotify.com/album/3zoTvDRS0ANwoNp8AFn36d

  • For the brave ones: spectrometric analysis

    The best way to test by yourselves the quality of a monitor, is through spectrometric analysis.
    Music, no matter its tones, is always different.
    Therefore, you should try out all the music in the world to have an accurate test of a monitor quality.

    …or

    send it a signal with every possible frequency (white/pink noise), and analyse the resulting sound with a high fidelity microphone (measurement microphone).

    In technical language: a spectrometric analysis.

    The result of this analysis is going to be a frequency response.
    Which is exactly what we have written about previously (5).

If you want to know why this tutorial was made, you’ll find out more in this post:

Our first post.

And you got our entire website to hear if we’re talking about something that we can do.

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