How to get Pro Tools for free in 2024

 How to Get Pro Tools Legally and for Free in 2024

Pro Tools is the most popular Digital Audio Workstation (DAW) software in professional recording studios. Originally created by the company Digidesign, it became so popular that after a few years, it was acquired by the video and multimedia company Avid. For many years, it has been considered the "audio standard" in many recording studios, and now you can have it on your computer for free!

Advantages of Pro Tools

Pro Tools is famous for being easy to use, stable, and quite powerful in terms of audio functions. It is excellent software for recording and editing audio, as it allows both functions to be performed with ease and speed.

It is also perhaps the most widely used program for mixing music, film, and television. Most film, TV, and recording studios have a Pro Tools system. Its main advantage is that it is a standard in audio production.

Therefore, it is very important for audio professionals (or aspiring ones) to learn how to use Pro Tools and have a version on their personal computer.

Disadvantages of Pro Tools

However, Pro Tools is not without its drawbacks. One of the main disadvantages is that it is a somewhat expensive system. This has been the case since its early versions many years ago.

The reason for this is that Pro Tools was not just software; Digidesign designed it to be a comprehensive system that included both software and hardware. This involved using digital signal processing (DSP) cards that had to be installed in the computer, as well as special interfaces.

Classic Pro Tools cards and interfaces

This situation was necessary in the 90s (when Pro Tools started) because computers of that time were not capable of efficiently processing digital audio. Therefore, Pro Tools was the first system to allow for a digital audio studio through a computer.

Returning to the present, both Pro Tools and computers have evolved, so it is no longer necessary to have any special hardware to run the program. This has made it much more accessible for everyone... even for FREE!

Versions of Pro Tools

So now we can have Pro Tools on our computers (Windows or Mac) without having to pay for the software license. All completely legal, although with limitations. Avid offers various options according to the user's needs. Here are the options available in 2024.

Pro Tools Intro

This is the best option for those starting with Pro Tools, as it is the completely free version. Its most important features include:

• 8 audio tracks and 8 MIDI tracks
• 4 audio inputs (depends on the interface used)
• Many editing functions and tools, such as Elastic Audio
• 36 processing plugins
• Resolution of up to 96 kHz, 32-bit floating point
• Does not support working with video

As we can see, Pro Tools Intro is a good option for those starting in audio work or needing to work on simple projects. In reality, the main limitation is that it does not allow importing video, so these types of projects are completely ruled out.

Pro Tools Artist

This is a more complete version and includes features such as:

• 32 audio tracks, 64 MIDI tracks, and 32 instrument tracks
• 16 audio inputs (if the interface allows)
• 32 folder and routing tracks
• Over 100 plugins
• Special version of Melodyne for professional voice tuning
• Audio-to-MIDI, Beat Detective, and Clip Gain functions

This version is ideal for musicians or producers who require more tracks and processing tools. The cost is $10 USD per month.

Pro Tools Studio

It is a more complete version, designed for audio and music professionals. Among its additional options are:

• 512 audio tracks
• 512 instrument tracks
• 1024 MIDI tracks
• 128 auxiliary tracks
• 128 VCA tracks
• Complete collection of plugins and processors
• Option to work with video
• 64 audio inputs
• Surround system handling and Dolby Atmos

Its price is $30 USD per month.

Pro Tools Ultimate

This is the most complete version of Pro Tools, although many may not require all the options it offers. In addition to what the studio version has, this version includes:

• Over 2000 audio tracks
• Handling up to 256 audio inputs
• Use of HDX hardware
• Advanced recording and editing functions
• Support for AAX DSP plugins
• 64 video tracks

As we can see, this version is ideal for the most demanding professional situations. Like working with HDX hardware, which allows processing audio using additional cards (as with the early versions of Pro Tools), relieving the computer's CPU and allowing for running huge and complex sessions. It can also work with multiple video files simultaneously for audio post-production.

The price is $100 USD per month, and there is access to a special price for students and teachers.

Requirements

The minimum computer requirements for Pro Tools Intro are:

Windows

• Intel i5 CPU or higher
• Windows 10
• 8 GB of RAM
• 15 GB of free disk space
• Internet connection
• It is recommended to have an audio interface that supports ASIO

Mac

• Intel i5 CPU or higher, or Apple Silicon
• MacOS 10.15.7 or higher
• 8 GB of RAM
• 15 GB of free disk space
• Internet connection

To have Pro Tools Intro, you will need to go to the Avid website and create an account. There you can download the installer according to your operating system or the Avid Link software, which manages installations and updates.

You will also need to create an account on iLok.com to authorize and use the software. This is also a free account that will allow you to use Pro Tools via iLok Cloud, which is a cloud service that monitors licenses to prevent illegal use.

In this sense, it is necessary for your computer to be connected to the internet while using Pro Tools, as the iLok Cloud system needs to be online when using the software.

So now you know the basics to have Pro Tools on your computer without spending a single cent. Although Pro Tools Intro has some limitations, for many users, it won't be a big problem, and you can start using this excellent DAW right away!

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Choosing a computer for music production in 2024

How to choose a good computer for Music Production

 When we enter the world of music production, one of the most common questions is figuring out what type of computer (PC or Mac) is recommended for optimal work. 

In this article, we will seek to address the most common doubts on this controversial topic, especially for beginners.

PC or Mac?

The first question we need to answer is whether we need an Apple computer, a Windows PC, or perhaps an option for Linux.

Let's clarify regarding Linux first. Linux is undoubtedly an interesting system in many respects, however, in terms of music production, the Linux ecosystem is quite limited compared to a PC or Mac. The primary limitation lies in the available Digital Audio Workstations (DAWs) for Linux. While there are some good options like REAPER, Mixbus, or Bitwig Studio, the variety of virtual instruments, plugins, libraries, etc., falls short compared to what other systems offer. Additionally, hardware setup and compatibility tend to be more complex.

For this reason, our first recommendation is to avoid Linux systems for music production, unless you are a programming geek passionate about Linux. It's definitely not a good option for beginners.

Apple Computers

Mac computers are undoubtedly a great choice for both beginner and professional music production. Among the main advantages are:

1. Operating System: MacOS is one of the most stable and secure systems. While it can fail, like any other system, it is highly reliable, clean, and often prevents many issues such as viruses and corrupt files. MacOS is designed to work with audio efficiently, using Core Audio technology, making it an excellent choice for demanding professional applications.

2. Native Hardware: All Apple computers come with good hardware for audio playback and even recording. You can work with your Mac's built-in outputs without needing to buy an audio interface, and the sound quality will be quite good. While you'll eventually need an interface for more demanding tasks, it's not a necessity when you're just starting.

3. Apple Ecosystem: Consider that with a Mac, you enter or expand your Apple ecosystem. If you have an iPhone or iPad, the Mac allows practical and seamless integration with those devices. This can benefit you when producing music, as you can use mobile apps as virtual instruments, MIDI controllers, or extensions of your DAW.

4. Native Software: All Macs come with Garageband, a friendly and comprehensive DAW that allows you to start working without acquiring another program. For users needing something more advanced, Apple offers Logic Pro, one of the most popular and powerful DAWs. If you've learned to use Garageband, transitioning to Logic Pro won't be complicated.

In addition to this, practically all popular DAWs are available for Mac. Apple computers are definitely a great option to start in the world of music production.

Their disadvantages usually revolve around the higher cost compared to Windows PCs. However, with the new generation of Apple Silicon computers (M1, M2, etc.) that Apple has released, costs are much more competitive and powerful, making a Mac worth considering.

Windows PC

Windows PCs are the most popular computers globally. While initially not the best candidates for audio work, they have improved significantly over the years and are also a great option for both beginners and professionals.

Factors that can help us decide if a PC is our best option include:

1. Preferred DAW: While most DAWs run on both Mac and PC, some may only work on PC. This should be the first consideration when choosing a computer.

2. Budget: PCs can be more economical in their basic form. Carefully examine this aspect, as a PC suitable for audio may have a higher cost than a basic PC for school or office use.

3. Operating System: Windows has improved considerably in recent years. If you love Windows or feel comfortable with this system, you can definitely continue using it as a base for your music production software.

Technical Specifications

If you are evaluating a PC's capability to work with audio and music, we recommend the following minimum specifications:

- Windows 10: This is definitely the most stable and recommended version for most programs.

- Intel Core i5 or AMD Ryzen multi-core processor: This is a crucial part, as the processor handles audio processing operations. A low-capacity processor will make your work too slow and frustrating. While a more advanced processor, such as an i7, is better, these minimum processors will allow you to do general work without issues.

- 8 GB RAM: RAM allows you to run multiple programs and plugins (audio processors or virtual instruments) simultaneously. As your projects become more complex, you may need to increase RAM to 16 GB, 32 GB, or even more, depending on your experience.

- Free Disk Space of 10 - 40 GB: Storage allows you to save more projects on your computer. Additionally, each DAW requires a certain amount of space for installation. It's worth noting that using SSD (Solid State Drive) storage is highly recommended over hard drives. SSD transfer speeds are much faster than any hard drive, ensuring excellent performance for audio programs.

Unlike Apple computers, typical PC audio hardware is not of high quality. Therefore, it is practically mandatory to get an audio interface to work properly from the beginning. Make sure your interface has ASIO drivers for correct functioning and high-quality audio.

As you can see, there are many options to find a good computer and start creating music. Often, it's not necessary to make exorbitant expenses initially. It's better to start with a basic but high-quality setup and increase your computer's capabilities as your level of expertise and needs grow.

If you want to start in the world of music production from scratch, you can check out my online courses, where you'll learn step by step in a comprehensive and clear manner. Click on the following links for more info:

- Audio Engineering Fundamentals.

- Introduction to DAWs with REAPER.

- Principles of Acoustics for the Home Studio.

I'm confident they will greatly help you on your journey as a producer. See you next time!

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Understanding sound absorption

 Sound Absorption: A Key Element in Acoustic Design

In our previous articles, we explored the acoustic characteristics of different spaces, specifically delving into the distinction between reverberation and echo. The reverberation time of a space is closely tied to the sound absorption properties of its materials. In simple terms, a space adorned with carpets and curtains will exhibit less reverberation compared to one with tiles and concrete surfaces. This is because materials like curtains and carpets are more effective in absorbing sound than hard surfaces like tiles or concrete.

Defining Sound Absorption

Technically, sound absorption refers to the conversion of acoustic energy into heat. As we know, energy is neither created nor destroyed; it simply transforms. When sound waves encounter a surface like a carpet, their energy doesn't vanish but rather transforms into another type of energy—in this case, heat. However, it's crucial to note that the power of this energy is quite low. To put it into perspective, the energy generated by an entire stadium cheering for a goal would barely be sufficient to heat a cup of coffee. Absorption is just one of three phenomena that occur when a sound wave encounters a different medium; in general, three phenomena take place:

1. Reflection: Some of the energy reflects back.

2. Absorption: Some of the energy is absorbed by the material.

3. Transmission: Some of the energy passes through the material.

For example, in the illustration above, sound energy emanates from a source (S). While in the air, a portion of it gets absorbed (E). Upon reaching a wall or obstacle, some of the energy reflects (A), and another portion transmits and gets absorbed in the material (F and H).

Measuring Absorption

Each material has the ability to absorb a varying amount of sound energy, a property measured by the absorption coefficient. The absorption coefficient is a number indicating the percentage of energy absorbed by a material, and it has no units. A coefficient of absorption equal to 1 implies 100% absorption with no reflection, while a coefficient of 0.25 means 25% absorption and 75% reflection. Generally, porous materials tend to have higher absorption coefficients than hard materials. For instance, curtains have a higher absorption coefficient than concrete.

It's important to note that the absorption coefficient is not constant across all frequencies. It changes with frequency, leading to tables displaying absorption coefficients at different frequencies.

In the image, various absorption coefficients are represented by the Greek letter alpha (α). As observed, gypsum board absorbs lower frequencies better than higher frequencies. Understanding these coefficients is crucial for planning the acoustic treatment of a space, and comprehensive tables exist to guide the selection of materials commonly used in construction.

It's also worth noting that different materials can yield similar acoustic effects. Believing in a "magic solution" is a misconception; achieving the desired acoustic outcome often requires a well-thought-out combination of both absorbent and non-absorbent materials.

Practical Solutions

Sound absorption finds practical applications in various scenarios. For instance, when recording, portable acoustic panels can assist in controlling sound. These panels, crafted from foams with high absorption coefficients, prove to be excellent choices for home studios lacking additional acoustic treatments.

When aiming to acoustically treat a room, the recommended approach involves using acoustic panels like these. In larger spaces, covering a greater surface area becomes essential, necessitating the use of larger panels.

It's crucial to emphasize that the quantity of acoustic material or the surface area to be covered should be determined by an acoustic expert. A thorough study of the room's acoustic characteristics is essential for satisfactory solutions. Filling every wall and ceiling with acoustic material is a mistake, as excessive absorption can lead to a sound that is overly dull and lifeless.

If you want to delve deeper into the world of sound, consider exploring my Introduction to Audio Engineering course, where you can learn the fundamentals of this captivating field!

Also, you can also expand your knowledge of Applied Acoustics by taking my Acoustics online course. More information and free lessons are available here.

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Common acoustic problems in the Home Studio

 The most common acoustic problems

Acoustics is one of the most common issues that a music producer faces in the home studio. Acoustics plays a crucial role in what we hear and how we hear it. 

It doesn't matter how great your audio interface is, or how expensive your studio monitors are, if you listen in a room with bad acoustics, the overall sound quality that you get will be less than optimal.

The world of acoustics is complex. Sound waves behave differently according to their frequency. So, low frequencies behave very differently than mid or high frequencies. This also means that the solutions to control each frequency range are different as well. 

The two main acoustic problems

In most rooms, there are two main "problems" or areas that need to be addressed. Each one needs a different solution. 

Sound Treatment

This issue has to do with the way the sound behaves inside the room. When a sound is produced in a room, it gets reflected back and forth on the different surfaces, walls, ceiling and floor. This creates all kinds of echoes and some reverberation.

These effects must be controlled to have a good listening experience. Without proper acoustic treatment, the sound of recordings or mixes can be confusing. There may be cancellations and colorations that will affect the real sound.

Among the solutions for sound treatment, we have absorbent panels, diffusers and bass traps. We can also use curtains or certain materials on the walls. 

All this treatment will improve the listening experience, but they will do a little to tackle the other main problem: soundproofing.

Soundproofing 

This problem has to do with sound isolation. We want to avoid that some external sound will come into our room or vice versa. Soundproofing has to do a lot with the materials of the walls, but also with the sealing of doors and windows.

Low frequencies are the most difficult to control. This has to do with the wavelength, as large wavelengths are more difficult to block. 

Glass windows are also the weakest links. Glass is not very effective to stop low frequencies, so, it's a good idea to avoid too many windows if possible. 

The most important point that we need to understand, is that each problem has a particular solution. A good, all-around acoustic work needs to fix both problems. So, in reality, there's not a simple solution to solve all the acoustic problems of your studio or any other room.

Here, you can watch a video with more info about this.

If you want to learn more about acoustics, please check out my online course Principles of Acoustics for the Home Studio. You can see the syllabus and watch sample lessons here.

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The Acoustics of the Home Studio

Learn the applications of Acoustics in the Home Studio. 

A home studio is a room designed to record and produce music at home. In recent years, home studios have become more common, as technology allows us to create music with a moderate budget. Nevertheless, the acoustical treatment of a studio is usually overlooked.

Acoustics

The days when the only way to record music was in a professional recording studio, are now gone. Nowadays, almost anyone can do a decent job recording music or voice at home. You just need an audio interface and a decent microphone, right? Well, not entirely. 

Even if you have some nice equipment, you have to consider the importance of acoustics in the overall sound production. 

What is acoustics?

Acoustics is a branch of physics that studies vibrations in general. Sound is a vibrating phenomenon, so acoustics deals with sound as well. There are many sub-fields in acoustics. For example, the study of musical instruments, noise control, vibrations in buildings or cars, and how to optimize a room for better sound. This last field is known as architectural acoustics. 

The home studio

The typical home studio has two main problems to deal with: how to isolate the sound and noise that enters or gets out of the room, and how to improve the listening experience in the room.

The first problem is solved with soundproofing. This is a very important area of acoustics that helps to keep your studio quiet from external sounds and also keeps your neighbors happy! 

The second problem is solved with acoustic treatment. This area helps to have a better listening experience in your room by controlling the sound reflections and bass resonances. 

It's very important to understand that those are two different problems that need different solutions. In other words, a solution that helps to soundproof your studio, won't be very effective to improve the sound within it and vice versa. 

This misunderstanding has created lots of myths around acoustics for years. For example, if you want to improve the sound isolation of your studio, you need to use special doors and windows, as they are the weakest link in the soundproofing chain. 

On the other hand, if you cover your walls with absorptive material (i.e. mineral wool), that will control the internal sound reflections (echoes) and may improve your listening experience, but it will do almost anything to control how much sound is escaping out of your studio.

Knowledge is power

Although acoustics is not an easy field, you can learn its basic principles to improve the listening experience in your home studio! I've created an online course that covers all the important topics:

• Sound behavior
• Control of high, mid and low frequencies
• Control of echoes and reverberation
• Speaker positioning
• Use of sound diffusers
• Selection of the "sweet spot" in your studio for best monitoring performance, and more!

All of this explained in a clear and easy-to-follow way. Without complex math and with practical applications. You can find more information here.

So, it's possible to improve the listening experience in your home studio. It doesn't have to be very expensive. You can even use materials that you already own! It's only a matter of desire to learn. 😄

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How to use a 3-band equalizer

Basic analog audio mixers (or DJ mixers) have limited equalization options. Many of them only have a treble and bass control, which are known as shelving equalizers. In other cases, there can be a third knob for the mid frequencies. 

Mixer with 3-band EQ (green knobs)

This kind of 3-band equalizer is good for its simplicity, but unfortunately, it's not very good for professional applications. When you are mixing several instruments or microphones, you have to adjust specific frequencies and this basic EQ is not the right tool for the job.

Anyway, sometimes that EQ is all you have and you'll need to do your best. In this video, I show you an alternative way to use a 3-band EQ. With this technique you can get different EQ curves using only the 3 available bands. 

Enjoy!

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The acoustic diffusers

What is sound diffusion?

When a sound wave travels and finds a surface, different phenomena occur. Sound absorption converts part of that energy into heat. Part of that energy is transmitted into another room and another part of the sound energy gets reflected as echo or reverberation. But there's another interesting form of reflection: the diffusion.

To understand diffusion and diffusers, we need to first understand that the sound waves follow the Law of Reflection. This law says that the incident angle of a sound wave is the same as the reflected angle.

Fig 1. Law of Reflection

In the picture, we see that if the wave gets on the surface with a θi angle, it will be reflected with a θr angle. Both angles will be equal if the surface is flat. This behavior is similar to a pool ball when it bounces on the edge. The sound waves will reflect in this way as long as the reflecting surface is larger than their wavelength. This kind of reflection is named specular reflection.

Wavelength

Wavelength is represented by the letter lambda (λ). It indicates how long the wave is. The higher the frequency, the shorter the wave. Audible frequencies, which can go from 20 Hz up to 20 kHz have wavelengths as large as 17 meters and as small as 2 cms (less than 1 inch).

As low frequencies have large wavelengths (more than 10 feet), it's difficult to see the Law of Reflection in those cases. With high frequencies, it's easier to "see" this behavior as they are smaller. Actually, you can think of mid and high frequencies as "sound rays". In the field of acoustics, this behavior is called Ray Theory.

Fig. 2. Different wavelengths

Diffusion

When the reflective surface is irregular, the Law of Reflection applies in almost every little spot of the surface, as seen in the picture.

Fig. 3. Sound diffusion

The result of this scattering is called diffusion. A specular reflection follows the Law of Reflection in a more evident way, you will hear that as echos. A diffuse reflection, on the other hand, generates a field of sound in all directions, you will hear this as reverberation.

Fig. 4. Specular reflection vs. Diffuse reflection

To diffuse or not to diffuse

The most important point is to know if it's better to use diffusion or not. For the most common situations, it's recommended to have a good amount of diffusion in the room. The more diffusion we have, the better the sound distribution and dispersion around the place. In other words, the sound will be more homogeneous in the room.

For example, in the case of a recording studio, the diffusion helps to make the sound that the microphones sense as more balanced. In this case, diffusion helps to avoid undesirable coloration and comb filters due to echos.

In the case of a mixing or mastering studio, diffusion helps to avoid early reflections on the sweet spot. These reflections can create image problems in the mix. Also, diffusion can create the sensation of a larger space even in a small room.

In the case of a concert hall or auditorium, diffusion is very important for the sound to reach more zones in the venue.

Acoustic diffusers

But, how can we create diffusion? We can build or buy systems designed just for that. In this regard, there is a great variety of designs. For example, the QRD (quadratic residue diffuser) systems, which are made of wells of different depths and dimensions. The size of these elements will depend on the frequencies that we want to scatter; larger dimensions, lower frequencies.

Fig. 5. QRD diffuser

Convex surfaces can also work as diffusers, but concave surfaces do not (i.e. church domes). An example of a "natural" diffuser would be a wall made of stone. The irregular surface of the stone has a diffusing effect on high frequencies.

Finally, we can also use a bookshelf as a good diffuser! If we gather books of different sizes and widths and we place them in a "random" way, we can achieve better results.

As we can see, the general idea is to avoid flat surfaces. This kind of surfaces will generate rapid echoes (specular reflections) that will have a very noticeable effect on the sound. Diffusion is very important for acoustic treatment in a TV room or home theater as well. We can use the resources available to us to improve our listening experience by using sound diffusion almost everywhere.

If you want to learn more about acoustics, please check out my online course Principles of Acoustics for the Home Studio! More info here.

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Logic Pro Scripter

Introduction to Scripter

Logic Pro X includes a very nice tool for the musician that loves music technology: Scripter, the MIDI programming script editor. This is a MIDI plugin that allows us to program new MIDI plugins by using JavaScript code.

Perhaps Scripter is the least understood plugin in Logic. The documentation is somewhat limited and you need some experience coding with JavaScript to push it to its limits. Anyway, it's worth it to give it a try and start working on it, as the possibilities are endless!

Fig. 1. Scripter main window.

But, what is it good for? Well in general, you can manipulate MIDI data in many flexible and powerful ways. For example, you can:

• Change the pitch, volume, panning, etc. of certain notes.
• Create new notes and play them back at different time intervals.
• Create automatic Control Change (CC) messages or transform them.
• Generate random notes or create notes based on probability (algorithmic music, this is your call!).
• Filter, transform or delay MIDI events based on our rules.

With this tool, we can create things like:

• A MIDI effect that emulates guitar strumming techniques.
• An algorithm to create automatic melodies or rhythms.
• Totally customized arpeggiators or harmonizers.
• MIDI Sequencers or drum machines.
• User interfaces with menus and sliders for our plugin.

Fig. 2. Scripter and the editing window.

As we see, we can create many interesting tools to unleash our musical creativity. It's important to mention that we can use Scripter right away, without writing a single line of code! Scripter comes with a variety of interesting presets that we can use immediately. Anyway, let's see how to start writing our own scripts or how to customize the presets according to our needs.

Let's begin with the basics: JavaScript

JavaScript (JS) is a programming language that was born along with the Web browsers. Regardless of that, it has become one of the most popular languages in recent years. Its applications reach areas like: the Internet of Things, mobile applications, virtual reality and of course, music.

This is because JS is a robust language, it's easy to learn and there's a huge community of developers around the world that use it. It's important to mention that JavaScript is not the same language as Java. The name looks similar but they are very different languages. Mozilla is the group that maintains and develops JS.

JS has a specific syntax that we should learn to write code. We will learn the most essential parts to keep our focus on what we want more: the music.

Functions

A function is a block of code that can be reused. Programming languages include basic functions and the programmer can create his own functions. In Logic, we have different functions that are defined in the Scripter API.

The simplest example of code in Scripter is a program that "resends" the MIDI data that it receives. Think of it as a virtual MIDI Thru. Here it is the code:

function HandleMIDI(event) {
    event.send();   //Resends the event that is received.
}

The word function is a keyword in JS. It indicates that we will use a function named HandleMIDI(). This function is part of the Scripter API and it will receive and process MIDI data. Then it will store it in the variable named event. This variable could be named in any other way if the programmer desires.

When the function HandleMIDI() is called (when the plugin runs), the instructions between the curly brackets will be executed one at a time, from top to bottom. In this example, there's only one instruction: event.send(). The text after the double slash (//) is known as a comment. It helps the programmer to understand the code. It's just for clarification purposes as it is ignored when the program runs. Although it's optional, it's recommended to use.

The command event.send() reads the data in the variable event and sends that data to the MIDI channel where the plugin is inserted in Logic. With this code, you can hear the notes played on that channel. Not a big deal, as it's the same effect as using no plugin at all! Each command or instruction ends with a semicolon.

Let's add another command to make our program more interesting:

function HandleMIDI(event) {
    event.send();
    event.trace();  //Prints the data in the editor console                
}

The command event.trace() prints a log in the Scripter console (lower part of the window). Here we can see the information inside the event variable.

Fig. 3. Scripter Console showing the MIDI data received.

In the image, we see that the console shows a list of MIDI messages received by the Scripter plugin. In this case, there are NoteOn and NoteOff messages, along with their parameters: channel, pitch, and velocity.

This code is very simple, but it is useful to understand the logic behind Scripter. Later we will see other more advanced commands to manipulate the MIDI data.

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Types of microphones and polar patterns (part 2)

Polar patterns

Last time we saw that microphones can be categorized in many ways. One of these ways is by its polar pattern. This pattern is represented by a polar graph. These graphs show us the points (in degrees) with most and least sensitivity.

The basic patterns are the omnidirectional, which captures sound from all directions, and the bidirectional, which senses sound from its front and back. Now we will see what happens when we combine these two patterns.

Cardioid microphone

If we create a 50% omni and 50% bidirectional microphone, we get a cardioid pattern. The name comes from the "heart-shaped" resulting graph.

Fig. 1. Cardioid pattern

This kind of microphone is also called unidirectional, as its maximum sensitivity is on its frontal axis (0º). On the other hand, there's almost no sensitivity on its back (180º). This is the most used microphone, as we can easily control what we want to capture with it.

Also, we can see that this microphone has some sensitivity from the sides (90º and 270º), although it's lower (-6 dB) than the on-axis response. This is why it's very important to look for the optimal mic position. Most of the time, it's best to point the axis of the microphone directly towards the sound source.

Fig. 2. Correct microphone technique

Hypercardioid microphone

When we combine 75% bidirectional and 25% omnidirectional patterns, we get this result: 

Fig. 3. Hypercardioid pattern

This is the hypercardioid microphone. As we can see, the pattern has little sensitivity from the back and it has two cancelation points at 120º and 240º. We also see that the lateral sensitivity (90º and 270º) is lower than the cardioid mic. This is why these microphones are very popular for live sound applications as they reject a good amount of ambient noise.

There's another similar pattern, the supercardioid. This pattern is somewhere between the cardioid and the hypercardioid patterns and it's very popular too.

All these microphones have the proximity effect. This is a low frequency boost when the microphone is close to the source. This is why all directional microphones change its tone or "color" as we move them closer or farther to the sound source. If we put them too close, the sound will have more bass (up to 15 dB as seen in the graph!). This can be used as a tool for sound shaping. It's like having an EQ integrated in the microphone! Anyway, if we want the purest sound, then the omni mic would be the best option, as it doesn't have proximity effect.

Fig. 4. Proximity effect (blog.shure.com)

Subcardioid microphone

This is the oposite pattern to the hypercardioid. It's formed with a 75% omni and a 25% bidirectional.

Fig. 5. Subcardioid pattern

We can see that it looks almost like the omni microphone but it captures a little less from its back. The main advantage of this microphone is that it captures a natural sound because of its little proximity effect but, at the same time, it gives us more ambient noise control than an omni microphone. Anyway, this pattern can produce more feedback problems than the cardioid or hypercardioid. This is why it isn't very popular for live sound applications.

Acoustic marvel

You may wonder, how can all these combinations be done? One way to do it is through acoustic tunnels. These tunnels are cavities that are designed for each kind of microphone, brand and model. The ducts change the phase of the sound wave from different directions and generate cancellations in different points. All this creates different and specific patterns. That's why its very important to avoid blocking the ducts and to hold the microphones in the right way.

Fig. 6. The vents and ducts create different polar patterns

Another way to get different patterns is using two microphone capsules, one omnidirectional and one bidirectional. Then, an electronic circuit mixes both signals and create different patterns. These patterns can be selected by the user. These microphones are called multi-pattern and they are very popular in recording studios because of its versatility.

Fig. 7. Multi-pattern microphone

Well, we have covered all the most important microphone patterns out there. This information should help the engineer or producer to select the most appropriate kind of mic for a particular application.

You can learn more about microphones in this video:

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Types of microphones and polar patterns (part 1)

The World of Microphones

A proper use of microphones is essential in audio engineering and music technology. Nowadays, there is a huge variety of microphones in the market. You can get into any online audio store and you'll see that there are a lot of brands and models available. So, the decision about what kind of microphone you should buy is not that simple. This question is quite common for audio engineering students and musicians as well.

Fig. 1. Typical handheld microphone

What is a microphone?

Many times, microphones are the first stage in the audio chain. They are transducers that convert air pressure variations into voltage variations. This conversion is analog, which means that the resulting voltage signal has the same variations than the original air pressure variations.

There are many ways to achieve this conversion or transduction. In this sense, we can classify microphones as:

• Dynamic. They use electromagnetism as their basic principle and they don't need batteries.
• Condenser. They use the capacitor's principle to operate. They need batteries or Phantom Power to operate.
• Ribbon. They are another kind of electromagnetic microphones, somewhat more delicate than the dynamic ones.
• Carbon. They use the electrical properties of carbon to achieve transduction.

We will talk deeper about these transduction forms in another time. For now, we will explain how microphones are categorized by its polar patterns. This means how microphones can capture sound from different angles.

Omni microphone

This microphone has a uniform sensitivity no matter the angle. This means that it's equally good for capturing sound from any direction (360º). In practical terms though, it doesn't behave completely omni at high frequencies (above 5 kHz), as it becomes more directional above those frequencies. It is also known as pressure microphone, as it senses the pressure variations around it. In a polar graph this pattern looks like a circle. The 0º point represents the frontal axis of the microphone.

Fig. 2. Omni pattern.

This mic works great as an ambient microphone, as it captures the sound from all directions. The omni pattern can be very useful for studio applications, as it adds very little "color" to the sound. On the other hand, it's not very popular for live sound applications because it can capture more ambient noise than other kinds of mics (remember your hands-free device?). This type of microphone is also common for acoustic measurement applications.

Fig. 3. Omni microphone for acoustic measurements.

Bi-directional microphone

This pattern is also known as figure-of-eight. This microphone has its highest sensitivity at both 0º and 180º, in other words, its front and back. Although the sensitivity is greatest at these points, the polarity is opposite. These microphones are also called pressure gradient microphones. That's because its operation depends on the pressure differences between the sound from the front and the back.

Fig. 4. Bi-directional pattern.

Its practical applications depend on its positioning. We can see that at 90º and at 270º there's a theoretical zero sensitivity point, so we have the greatest sound rejection there. If we compare this with the omnidirectional mic, the bi-directional only captures around 1/3 of the ambience captured by an omni. This is why we have more control of what we want to capture with this kind of microphone.

Fig. 5. Bi-directional microphone.

But, what would happen if we could combine these two patterns at the same time? Actually we can do it, and the result is a very interesting (and useful) pattern. Here it is the resulting polar graph. We will explain this and the remaining patterns next time!

Fig. 6. Resulting pattern from the combination of an omni and a bi-directional pattern.

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The new Logic Pro 10.3

Logic Pro 10.3.1

Logic Pro is one of the most popular Digital Audio Workstations out there. The recent free update from Apple (10.3) has many interesting features. I'll talk about those that I found to be the most interesting ones.

Installation

As usual with most Apple products, the update process is quite simple. The update is done through the App Store. The download size is around 500 MB, so it shouldn't take too long for most connections. The process is automatic and only requires your Apple account password. You don't need to reboot the system when the installation finishes, so you can start using Logic right away.

User interface (UI)

The first thing to notice is the change in the UI. This update has a "flatter" design, which is a common trend these days. It made me recall a little bit of Ableton Live.

Fig. 1. New look for the user interface.

Among the new features of the UI, we find that we are able to change the background shade. You can choose either a darker or a lighter appearance. This could be useful when lighting conditions change. For example, when using a MacBook outdoors.

Fig. 2. Background appearance option.

Another practical option is the horizontal auto-zoom. This allows you to see the entire session horizontally and auto-adjust the zoom level as the song becomes longer. Logic already had this feature for the vertical zoom, so this is a very nice addition in my opinion.

Fig. 3. Auto-zoom options activated.

Another very useful feature comes when we edit a region's size. When we place our pointer on the region's edge and trim the region, we can see a visual guide showing the whole file. This is a very practical feature for the editing process.

Audio functions

In this sense, we find nice functions that we were asking a long time ago. The first one is the possibility to use an internal 64-bit summing bus. This option is superior to the former 32-bit bus as it gives us more headroom while mixing and preserves more dynamic range. This feature puts Logic in the same league than systems like Pro Tools HDX.

Fig. 4. Internal 64-bit summing bus.

Another excellent option is that now we can use Stereo or Dual-mono audio plugins. If we are using a  surround track, we can use Multi-mono plugins as well. When we use a Dual-mono plugin, we can process the left channel with a different setting than the right channel. In other words, we have independent control for left and right parameters. This opens a lot of opportunities for creative processing while mixing or mastering.

Fig. 5. Dual-mono plugin.

Other new features

• Support for the new MacBook Pro's Touch Bar.
• 256 auxiliary buses.
• Region's effects rendering. 
• The MIDI plugins can control audio plugins parameters.
• Simultaneous fade adjustments in different tracks.
• Track alternatives menu.

Fig. 6. Track alternatives menu.

Now you have an idea of the new options in Logic Pro 10.3. The update is definitely worth it. Logic is a great DAW and this update makes me remember why it is my favorite one! :D

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What is Audio Mastering?

What is audio mastering?

Audio mastering is perhaps one of the most mysterious topics in the audio field. This may happen because mastering is done by a specialized engineer. Another point would be that the client is not usually involved in the mastering process and so, the kind of work done while mastering is not too obvious for most people.

Fig. 1. Audio console

But, what is mastering? We could say that mastering is the last stage in the music production process. That is, the last stage where we can make decisions and take actions from a creative point of view (sound processing). After mastering, there are no chances for any additional change, neither audio processing nor editing. In mastering we get the final master.

The master is the final disc which is the basis for all replications. Technically speaking, mastering is the manufacturing process of this master disc. Yet, we usually call mastering to the creative audio process too. In this stage we can improve the overall audio quality, spectral balance, dynamic range, etc. In our digital era, we can call "master" to the final audio file which will be distributed through an Internet platform: iTunes, Spotify, YouTube, Soundcloud, etc.

The mastering studio

The place where mastering should be done has very unique characteristics. Perhaps the most important one is the acoustic treatment. An engineer's decisions depend on what he can hear. Studio acoustics play a fundamental role in this sense.

Excellent studio acoustics can be very expensive. This is why it's not easy for everyone to have the optimal place to master a project, at least not at the top profesional level. In this sense, the mastering studio should have a delicate balance between sound absorption, diffusion and reflection. The soundproofing should be excellent as well. Besides of that, there shouldn't be large objects or surfaces that could affect the sound waves in a negative way. An example of this could be a big mixing board or studio windows.

Fig. 2. Mastering studio Eternal Midnight

The mastering gear

Another important characteristic in a mastering studio is the audio gear available. Mastering requires the best audio processors and converters (analog to digital and digital to analog) and this represents another very important investment. It's common to find very expensive equipment in a professional mastering studio. For example, monitoring systems above $10,000 USD, premium or vintage audio processors (equalizers, compressors) around $4000 USD each and software mastering suites that cost above $1500 USD.

Fig. 3. Typical mastering monitors

Nowadays, it's usual to find mastering projects for surround sound too. These projects could need 5.1 or more complex audio formats. In these cases, the monitoring requirements increase clearly (at least five monitors and a subwoofer). Furthermore, the electric installation and audio cabling must have excellent quality too to preserve the HI-Fi audio chain. All of this gives us an idea of how specialized a mastering studio should be.

The mastering engineer

The main recommendation is that mastering should be done by a professional with trained ears and a "fresh" approach to the project. This means that mastering engineer and mixing engineer should be two different persons. The advantage of this is that a different engineer could be able to listen to details that the mixing engineer could have missed. Many times, after long hours of mixing work, the ears can "deceive" the engineer because of ear fatigue. Then, the mastering engineer can help by balancing the spectral or dynamic levels from a different point of view, or should I say... hearing!

Proper mastering requires a lot of knowledge and skills from an experienced engineer. The ears play a crucial role for this. Many good quality productions can be done in home studios and then enhanced by a professional mastering engineer. The final result of this combination can be excellent.

Fig. 4. Mastering engineer Dan Millice

Back on Earth

Mastering is a fascinating area in the audio world. Nowadays, we have the opportunity to use accesible tools to get a very good quality master. All we need is a computer and a couple of software processors to begin with. There are also mastering systems based on AI. This is taking audio mastering to the machine learning frontier.

It is possible to learn basic mastering techniques and improve the audio quality of your productions too. If you are one of those who love to learn, experiment and develop their ears above the average, then mastering could be not so mysterious after all!

Online audio mastering course

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Harmonics and sound

What are harmonics?

When we talk about sound or music, we hear the word "harmonics" many times. Actually this word is related to music harmony. If we want to understand what harmonics are, first we need to understand that sound is based on vibrations. That is, sound originates with a vibrating object and then, this vibration gets to our ears though the air. After that, our eardrum also vibrates and then our inner ear generates an electric signal (bioelectric signal). Then, this signal travels through our nervous system to our brain, where it's interpreted as sound.

Fig. 1. Vibrations in the air forming waves.

Vibrations have an oscillation frequency which tells us how fast this vibration is. For example, a heavy object vibrates slower than a light object. The heavy object vibrates at a low frequency and the light object vibrates at a high frequency. We use Herz (Hz) or cycles per second to measure that frequency. Generally speaking, the human ear can sense vibrations between 20 Hz and 20,000 Hz (20 kHz).

To have a better idea of these vibrations let's think about a hummingbird. This bird can flap its wings at around 50 times per second. So this movement generates 50 Hz vibrations which we can perceive as a very low "hum".

Fig. 2. A hummingbird flapping generates 50 Hz vibrations.

On the other hand, a mosquito flaps its wings so much faster: around 600 times per second! That's why the mosquito generates a 600 Hz vibration that we perceive as a higher pitched sound... and a really annoying one!

Multiple simultaneous vibrations

We find vibrations in nature all the time, but this vibrations are not constant nor simple. Almost every sound we hear is made of different frequencies occurring at the same time. For example, when we pluck a guitar string, the string will vibrate at different frequencies at the same time. Yet, there will be a predominant frequency which usually sounds louder. We can see that in this slow motion video:

The lowest frequency of vibration is called fundamental frequency. The other frequencies of vibration are called overtones. When these overtones are multiple frequencies of the fundamental, then we call them harmonics, phew! Let's see an example to make this clearer.

The heaviest string of a guitar has a fundamental frequency of around 80 Hz (E note). The other vibrating frequencies (overtones) in that string are:

• 160 Hz
• 240 Hz
• 320 Hz
• 400 Hz
• 480 Hz
• etc.

All these overtones are multiples of 80 Hz, hence they are called harmonics. Harmonics usually have different intensities, some of them are louder than others. This depends on many factors, like the string material, playing style or technique and point on the string where the string is plucked. In the next figure, we can see the harmonic spectrum of an A string. The peaks represent the different harmonics, each with different amplitude.

Fig. 3. Spectrum of a string with fundamental frequency at 55 Hz (note A).

A very interesting fact is that strings produce all these harmonics in a natural way. This is why the most popular musical instruments are based on strings, like the guitar, piano or violin. Anyway, there are other vibrating systems which are also very good for generating harmonics, like the pipes or bars.

The amount and intensity of harmonics determine the timbre of a sound. Harmonics are also fundamental to perceive a sound as pleasant or unpleasant. This is applied when composing, arranging and performing music, and even while mixing or mastering audio!

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