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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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:

If you want to learn about Audio Engineering, you can take my Audio Engineering Fundamentals online course, where you'll learn about sound systems, connections. More info and a sample lesson here.

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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.

You can learn more about Sound Engineering in my Audio Engineering Fundamentals online course. You can find more info and take FREE sample lessons here.

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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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Introduction to Digital Audio

What is Digital Audio? 

Nowadays it's very common to find the word digital everywhere. The "digital revolution" is reaching every part of our lives: information, video, audio, images, money, etc. In the world of audio, there is a constant debate between digital and analog audio about which one is best. Each one has pros and cons and we actually need both of them when producing music. This time I am going to explain what digital audio is and why it's important.

First we have to understand that most audio signals begin as analog signals. For example, the electrical signal coming out of a microphone, guitar, synthesizer or MP3 player is an analog signal. This means that it's a voltage that varies with time. This is why we can represent it as a waveform.

Fig. 1. Sinusoidal waveform.

The amplitude changes (Y axis) represent voltage variations. X axis represents time. In the image we see that the voltage goes up and down as time goes by. The amount of variation will depend on the kind of sound that this signal represents. Thus, the waveform will be different if it's a high pitched sound, a voice, a drum, etc.

A digital signal, on the other hand, is a series of pulses. Usually, these are electric pulses, although there is technology that handles pulses of light instead (photonics). These pulses represent coded information and they can have two values only: 0 or 1. So we use a binary system to represent the coded information. Each one of these binary values or digits are called bits (binary digit). For example, in the image we see a two-bit signal, in this case we use 2 bits to represent different amplitude levels.

Fig. 2. Digital signal

This chain of bits only makes sense if we know how they are coded, so we can decode it later and extract the information. The most common coding system used for audio is called Pulse Code Modulation (PCM). This is the kind of modulation used for WAV or AIFF files, although there are other kinds of less common modulations.

100% digital?

Loudspeakers need an analog signal to work. This happens because its cone movement is proportional (is analog) to the electric variations of the signal. This means that any digital signal needs to be decoded and converted into an analog signal before reaching the loudspeaker. It also means that there are not a 100% digital audio systems. Likewise, we cannot simply asume that a digital system is better than an analog one.

Both kinds of systems have its pros and cons, depending on the application, goals and budget. A basic audio system can be digital on the recording, processing (editing, mixing, effects) and transmission stages but it has to be analog on the power amplifying stage (the signal going to the loudspeakers). Anyway, it's very probable that with technology development, we will have more audio stages on the digital domain.

Digital Signal Processing (DSP)

We could say that digital audio is just a bunch of numbers. That's why one of its main advantages is the use of Digital Signal Processing (DSP). This process consist on a series of complex calculations that can modify the original signal in several ways. This is what happens inside our computers or smartphones when we edit video or images too. Modern computers and devices give us the opportunity to achieve things not possible with analog audio. DSP applications are vast in many fields. In the case of audio, through DSP we can:

• Modify the pitch or duration of a sound or tune a singer automatically (sound familiar?)
• Create different effects like reverb, echo, chorus, etc.
• Process the audio with filters, EQ, compressors, etc.
• Reduce background noise or restore old recordings
• Analyze the components of an audio signal, its frequency, phase, amplitud, etc.
• Recognize patterns (voice and music)
• Many more things yet to come!

Fig. 3. Analyzer using DSP.

Digital audio is and will be part of our daily life, not only for the audio professional but for everyone. Its advantages for storage, transmission and its impressive processing potential make it a must for any music producer or sound engineer.

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Music Technology and Pure Data (PD)

Music Technology

This time I'll talk about a very interesting topic in the world of modern music: music technology. When we talk about music technology we mean all the tools that can be used for music creation, analysis and synthesis. We are living in an age where music technology grows everyday and the most interesting discoveries are yet to come.

Among the available music technology tools, we can find music programming languages. Here it is a list with the most popular languages:

• Csound. This is one of the oldest languages but still popular. It has tools for music composition, synthesis, live coding and more. It's based on written code, it is free and it can be used on any computer platform.
• Supercollider. Another code-based, free and multi-platform language. Supercollider is an object oriented language that allows real time performance, synthesis, algorithmic composition and many more.
• ChucK. This is one of the newest languages yet it has become very popular. This is because its simplicity and great synchronization features, live coding, synthesis, composition, etc. It is code-based, free and multi-platform.
• Reaktor. Unlike the other examples, this is not a "real" programming language, but a graphic developing platform instead. With Reaktor you can build synths and effects in a visual way. It is a commercial software and runs on Mac and Windows. Its graphic environment allows the development of creative tools and projects in an intuitive way and with excellent sound.
• Max/MSP. Another graphic environment where you can connect "blocks" and do many processing operations. These include synthesis, MIDI and even interaction with video or images. It's a commercial software available for Mac and Windows. It's very popular due to its great integration with Ableton Live. 
• Pure Data (PD). This one is like the "open" version of Max/MSP. It was created by the very same person: Miller Puckette. It's very similar to Max/MSP in its basic functions but it's open source, free and runs on Mac, Win and Linux. You can create synthesis, composition, audio, MIDI and control tools.

In future articles we will talk about working with PD. With this platform we can build the music tools we may need, for example:

• Synthesizers
• Audio signal processors
• MIDI processors (arpeggiators, splitters, etc.)
• Movement, camera or sensor controllers
• Interactive art tools
• Educational tools
• Algorithmic composition tools

All of this looks very interesting but we must understand that PD gives us the basic blocks to build all these tools. It's a kind of LEGO set for audio and MIDI processing. Anyway, this doesn't mean that building this kind of creative tools is a simple task. You need imagination, knowledge and persistence to make the things work.

Finally, we will use PD for many reasons. The first one is that PD is free and runs on any system. The second is that, being a graphic environment, it doesn't use a complex syntax nor code lines. You need to learn the language rules though, but the learning curve is so much more intuitive. The third reason is that you can use PD to work along with any DAW or virtual instrument. Another advantage is that it doesn't need a lot of processing power, so you can even use it with old laptops or even some netbooks, whether live or in the studio.

Music Technology is an essential tool for the modern musician or producer, that's for sure. These technologies will allow today and tomorrow's musicians to build the creative tools the may need. In this way, there will be almost no limits for the creative process and imagination in the world of music.

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How audio systems work?

The audio systems

We can understand an audio system as a chain of interconnected "blocks" or stages. The purpose of a system can be either sound recording or sound reinforcement. We will find many of these "blocks" in any audio system. Examples of systems are: a home studio, a live sound system, karaoke systems, school P.A, etc.

Mic and Inst levels

All systems begin with the audio inputs. The input can receive a signal from a microphone or from any other audio source, such as electronic instruments (keyboard, guitar), processors or other devices. These elements generate very small electrical signals (mV) so they need amplification. The levels of these tiny signals are called Mic-level or Instrument level.

Fig. 1. Mic level.

Whatever the input, the first stage in the system has to be an amplifier. This is called preamplifier or preamp. The preamp is an amplifier that takes a signal from a mic or instrument and amplifies it until it reaches a higher level. This level is called Line-level and it's about 1 V. Almost all consoles and audio interfaces have built-it preamps.

Line level

Once the signal reaches line-level, we can process and manipulate it in various ways. For example, we can equalize it, compress it or apply effects like echo, chorus, etc. Also, line-level signals can be mixed with other signals without noise problems. The line-level signal may be digitized too, so all these processes can be either digital or analog. We can also record line-level signals. This could be done in an analog medium (magnetic tape) or in a digital one (hard disk).

Fig. 2. Line level processors.

Speaker level

When the line signal has been processed and/or recorded, then we need to amplify it even more so we can hear it. This is done through another amplification stage called power amplifier. Power amplifiers raise the energy of the audio signal from line-level to speaker-level. This level can reach up to 100 V, so it must be handled with caution. That's why we have to use thick cables for this kind of signals. These signals are connected to a loudspeaker. The speaker will convert the electrical variations into movements of the cone. Then, the cone will move the air and will generate a sound wave.

Fig. 3. Power amplifier (speaker level).

This chain of stages is valid on all systems, but often we do not realize it because the stages are integrated within the gear. But even with digital developments and new technologies, we will always need:

• Audio inputs
• A preamp
• Some kind of processing (EQ, effects)
• A power amplifier
• A transducer (speaker or headphones) that generates the sound wave.

Fig. 4. Mixer with processors and power amplifier included.

It is important for the sound engineer to supervise that the signal goes from a small voltage to a larger one. You should always consider the amount of energy at each stage when interconnecting equipment. That is, a microphone signal should be connected to a preamplifier, this one should feed a processor and so on without skipping stages. Not following these guidelines may cause noise, distortion or damage to equipment. For example, we should not feed a power amplifier with a microphone signal because it is too small for the amplifier. Nor we should drive the input of a processor with a speaker-level signal, as we could damage it!

In short, there are many possibilities, but the important point is to remember that we must always follow the order: Mic - Line - Speaker levels. This way we will avoid many problems for us... and for our wallet!

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Balanced and unbalanced lines

What are audio lines?

Hello, this time we will talk about an important topic in audio. It is related to how to interconnect equipment, this is: balanced and unbalanced lines. When we talk about a line (transmission line) we mean the cable used to carry the electrical signal. The cable type we use can make an important difference in the transmission quality. This is due to the electrical characteristics of the conductor's material and the signal's frequency.

So we must know the application to use the best audio cable for the job. Different cables are needed for: microphones, guitars, speakers, digital signals, etc.

Unbalanced lines

Unbalanced lines are the simplest and thus, the most common found in lower-cost or semi-professional equipment. It consist of two wires: positive (+) and ground (GND). The most common connector for this kind of line is the TS (tip, sleeve) plug or "mono" plug.

TS plug

In these connectors the tip is positive. Then we find a plastic strip and then the ground, which is the sleeve. The cable only needs two poles for unbalanced signals. Another type of unbalanced connector is the RCA. This is used for video and audio signals. While the audio quality is not degraded with good quality RCA connectors, it is recommended to use TS whenever possible.

RCA

The main disadvantage of an unbalanced line is that the signal is more vulnerable to interference and noise. This could come from motors, lights or radio signals. An unbalanced cable is only useful for distances less than 5 meters (15 ft.). For longer distances the risk of noise increases. Anyway, these lines are still very common. They are the standard connections for electric guitars, keyboards and devices such as tablets, mobile phones or MP3 players. This is because the typical distances for these devices are not large.

Balanced lines

These lines use cables with three poles: positive (+), negative (-) and ground (GND). The signal travels in both the positive and the negative wires, but it does so with reversed polarity in the negative one. For proper operation it requires special audio inputs in the equipment. These inputs need a circuit called differential amplifier that can receive both signals, positive and negative. This feature of receiving two equal signals with reversed polarity allows so much more immunity to noise and interference. On the other hand, its implementation a bit more complex and expensive. The connectors used for balanced signals are the TRS plug (stereo) and the XLR (Cannon).

TRS and TS plugs

In the case of TRS, the tip (3) carries the positive signal. The ring (2) carries the negative signal and the sleeve (GND) closes the circuit. The TRS connector has two insulating strips (4), while the TS has only one. It is important to notice that, as in the TRS two signals can travel at the same time, it is often used to carry an unbalanced stereo signal. This means that the right and left signals are sent on the same cable. An example of this application is the headphone cable. In the case of the mini-plug (3.5 mm) the connections are the same as the standard plug. The mini-plug connector could be both TS or TRS versions.

The XLR (Cannon) also has the same functionality as the TRS, but it is more robust and secure. In this case each pin is numbered and has a standardized function: Pin 1 ground (GND), pin 2 positive (+) and pin 3 negative (-). The numbers are indicated on the connector itself.

Cannon

As we said, balanced lines are much more effective against noise. A balanced line can easily reach 50 meters (150 ft.) without noise problems. This is why they are the best way to connect pro audio gear. Finally, we must clarify that it is not possible to balance a signal just by changing the type of connector or cable. For example, to put a TS at one end of the cable and an XLR at the other. This merely serves to adapt connectors but the signal remains unbalanced at the end of the XLR.

The correct way to balance a signal is using a "direct box" (DI Box). These boxes have internal balancing circuits (usually transformers) that do the balancing job. Using DI Boxes can lead to an important investment, but it is worth it if good audio quality is what we are looking for. We can find active (amplified) and passive (without amplifier) DI Boxes.

DI Box

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What is a preamplifier?

The Preamp

One of the most important concepts in an audio system (and also most ignored by many operators) is the preamp, also called pre. The preamp is perhaps the most critical stage for handling an audio signal. It is at this stage that our signal can increase its amount of energy even up to a million times!

We do not have that amount of amplification at any other stage. It is crucial to understand how to operate a preamp in a right way. Many of the most common problems in audio systems, such as distortion, noise and feedback are usually due to poor preamp adjustment. In fact, in my courses and seminars I always start with the preamp setting before moving on to the rest.

The first thing to understand is that the preamp is an amplifier that is usually found at the inputs in the consoles or mixers. Its function is to deliver the amplification needed to connect various signals to a mixer. These signals could come from: microphones, electronic and electroacoustic instruments, CD players, effects processors, etc. All these signals are different and need a different amount of amplification.

The preamp takes care of that so that all signals entering the mixer will have a similar level. In this way, they can be mixed better and we reduce the chance of distortion or noise at the mixer's output. The preamp's gain knob is generally found on the top of each channel and is usually labeled as Gain or Trim. Some gear also has a label next to the knobs showing the amount of gain in dB's.

Fig. 1. Here the preamps are the red knobs on the top

The key point is to know how much gain we need on each channel. This is what is known as gain structure and it's a topic to be discussed later. However, in general we can mention the following procedure:

1. Turn the knob to the minimum before connecting a microphone or instrument. Notice that placing the preamp in this position doesn't mean that we have "zero volume", but rather that the amplification is low. In other words, some microphones can send enough signal even with the knob at the minimum.

2. Connect the microphone or instrument and ask the musician to sing or play. This is known as line check. (Note: they will not hear themselves right away as the faders are down).

3. Press the PFL (pre fader listen) button in the channel and check the signal's level on the console's main meters. Usually there will be a little level and only the lower LEDs will light up.

4. Adjust the preamp so that the signal level fluctuate around 0 dB in the main meter. The amount of gain required will depend on several factors such as: musicians volume, microphone type and distance, vocal techniques, etc. We can see that if somebody else uses the microphone we will have to make adjustments.

5. Turn the PFL off. The output volume will depend on the fader's position and the power amplifier's gain setting. The faders are usually set around 0 dB. The power amplifier's gain needs a deeper explanation that we can have other time.

Fig. 2. The master section

Finally, it is important to mention that there are also external preamps. These are often used in professional studios to get a different sound. These are usually installed in a rack and can cost up to $5000 each! Using these preamps depends on the sound quality that the engineer wants. As several of these preamps base their operation on electronic tubes or valves, these devices give a 'color' to the signal. Many producers and engineers have their favorite ones for each particular situation. In other cases, they should use the preamp included in the mixing board.

Fig. 3. Outboard preamp with EQ

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What are the decibels (dB's)?

The decibel

In the audio world we often encounter people using concepts and terminologies that are not so common in other areas. In this sense, one of the key concepts that we should understand are the decibels (dB).

We need to make clear that the dB's are not a unit of measurement. They are an indicator of the amount of change that has a signal's power. For example: 0 dB means zero changes, not zero energy. That is why when we adjust a fader or EQ at 0 dB, what we are doing is to leave the signal's power intact, neither increases nor decreases. In fact, in most audio processors, it is common to initialize them at 0 dB.

Fader set at 0 dB (no changes)

A +3 dB, +6 dB or +10 dB setting represents a boost of energy, while negative values represent signal attenuation (-3 dB, -6 dB, etc.). An interesting case is infinite attenuation, represented by the symbol of infinity (∞). It is usually found in the lower part of a fader. At this point, the attenuation is the highest and the signal does not continue to flow in that path.

Another function of the dB is to represent physical values, e.g: voltage, power or sound pressure. In these cases the name dB is accompanied by other letters. The most common are:

• dBu. Used to represent volts and found in professional audio equipment. 
• dBV. Represents volts in home or Hi-Fi equipment. 
• dBm. Indicates electrical power (milliwatts). 
• dB SPL. Indicates sound pressure level (Pascal). 

They all mean different things and should not be confused, i.e: 0 dBu = 0.775 V 0 dBV = 1 V 0 dBm = 1 mW 0 dB SPL = 20 Pa (micropascals) So 0 dB (no change) means something different than 0 dBu (0.77 V) or 0 dB SPL (20 micropascals)! As we can see, each type of referenced decibel means an already standardized amount. Furthermore, we note that 0 dB in these cases represents a physical value. For example, it is the same to say 1 volt or to say 0 dBV. 

The reason for using dB's instead of conventional units such as volts, watts or pascals is because decibels represent logarithmic changes that resemble how our ear perceives changes. For example, a sound of 0.02 pascals (60 dB SPL) is perceived as a moderate volume. A louder sound would need 100 times more pressure, that is 2 Pascals (100 dB SPL). It's easier to handle 40, 85 or 100 dB SPL than more complex numbers as 0.003 or 0.00007 Pascals.

Our hearing system requires large amounts of energy to hear changes. The decibel is a tool that allows us to manage values in a more compact scale for greater simplicity. We need to explain the dB's and its applications in a more detailed way, but for the moment, we have a better idea about  this important concept.

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An introduction to Audio Equalization

What's the equalization?

Equalization is one of the most popular and important processes in the world of audio. With EQ we can improve sonic details that we could not avoid when we recorded something. It is important to say that equalization is not a "magic tool" that you use all times. In fact, many engineers say that it is preferable not to have to use equalization or use it only when necessary.

The name equalizer (equaliser in UK) refers to making something equal to other thing. That is, its original function was to fix the sound captured by a microphone. This was necessary due to the technical deficiencies that many old microphones had. This is why we use the equalizer, for example, to increase high frequencies that the microphone did not capture well. Currently, equalizers are still used in this way in professional audio.

They are also used to create special effects, such as the "telephone" effect, old radio, megaphone, small speakers, etc. EQ can compensate shortcomings of the sound sources and not necessarily a microphone. For example: a person wants a deeper voice and uses equalization to add bass and get that effect.

One of the most important applications of the EQ's is to attenuate or filter out certain frequencies that make the sound "dirty". I don't mean removing noise (there are other more specialized tools for this task), but to reduce resonances and frequencies that are annoying to the ear. I think we've all heard those cymbals or electric guitars that are unpleasant or even hurt.

In my opinion, any audio signal that disturbs our ears is either bad equalized or recorded. Good audio should be a delight and an experience that invites us to listening to it, not to want to run away from it! I mention this regardless of the genre, because, although there is music that is aggressive in its interpretation, the sound should be clear, definite and strong, but not unpleasant.

Learning the art of equalization: what and how equalize, requires a lot of expertise and a lot of practice. In this regard, I have published a video course explaining every detail of the process and with many examples. With this link, you can take it with 50% off!

So, equalization is a very important deal for audio engineers and musicians, but with practice, guidance and good ears it is not an impossible task.

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