How do software equalizers work

Parametric equalizers are extremely flexible but can be harder to understand. Each has a set of fully configurable bands, allowing you to amplify or attenuate a certain range of frequencies, or everything before or after a certain frequency point.

The hardware units are usually restricted to professional audio environments, as they are costlier and much harder to use. The software versions however, are widely used and are the main tools for you as an audio engineer. We will cover the basics of parametric equalizers using MAutoEqualizer.

Please download the demo version for free, to follow the tutorial and use the integrated analyzer to actually show you what the equalizer does and how it works. Each band of the parametric equalizer has at least 4 parameters: frequency gain filter type Q Frequency - its use and meaning depends on the type of filter. For some types, it defines which frequency is affected the most; for others, it indicates the end of the range that the filter will influence.

Gain - allows amplification or attenuation of a certain range of frequencies. Filter type and Q are discussed below.

Filter type defines the shape of the processing curve, and therefore how each frequency will be affected. There are several standard filter types. We cover the principal ones below. In all cases, the curve in the graphs shows the relative effect on the frequencies. Peak filter has a typical bell type curve around the central frequency, showing the frequencies affected by the specified gain. These filters are the most commonly used.

For example, a snare drum, usually has its most dominant frequencies somewhere around kHz. Using this filter is a matter of finding the frequency and amplifying it. Simply take the band frequency point and slide it across the spectrum.

The hardest part is deciding where it sounds best ;. These are mostly used on bass or treble. For example, to make a track or even the entire mix 'brighter', just use a high-shelf filter and amplify everything above a certain frequency. Use this effect with caution - adding bass or treble often seems to sound better, however this may just be the 'hi-fi effect'. Well, apart from some theoretically perfect filters, that's not entirely true but they do attenuate particular frequencies a lot.

For example, you often need to remove bass content from vocals, piano, guitar and other higher pitched tracks, so that sonically, they don't collide with the bass and bass drum. You could use a low-shelf filter, but that would usually let you attenuate by only 24 dB, which may not be enough.

Additionally, there is generally no reason to keep 50Hz in an acoustic guitar track, as it is probably only hum and noise. A high-pass filter, can easily remove 80 dB from the frequencies you just don't want. Please note that this type of filter does not use a gain parameter as it has no need for it. For example, you may wish to remove some unwanted noise. It may be some electrical hiss, ambient resonances, or mechanical flaws on drums etc.

In most cases these appear on a constant, usually small, frequency range, so you just need to remove them. A band-pass filter will remove everything except the specified range. Slide the central frequency to the point that you hear the noise the most.

Then switch to the notch filter. The notch filter removes the unwanted frequencies that you targeted with the band-pass filter. Often called resonance, the Q parameter defines the steepness or width of each filter depending on its type. The width of the filter, or Q, becomes narrower the higher the number selected. With MAutoEqualizer for example, the Q ranges from 0. Please not that not all Q values are appropriate for all filter types.

As an example, if we try using a low pass filter set to a frequency of say 1 kHz with minimal width Q slider fully to the left, and maximal Q value of 20 , we create 'a huge hill' near the 1 kHz band. This is called resonance and is caused by the way the filter works and the physical limitations. What we have tried to do in effect, by specifying minimal width, is to create an extremely steep filter, which is unfortunately not that simple.

There are some advanced methods to create steeper filters, but they have other flaws ripple in the pass-band, post-ringing, instability etc. This is not a digital audio error, it is simply part of the design and is also present in hardware equalizers. This is basically the reason why the Q parameter is often referred to as resonance. If you listen to the sound produced by such a filter, it is similar to an object glass for example , resonating at a particular frequency, making it louder.

It's natural for a mechanical object to generate harmonics. If it resonates at 2 kHz for example, it is probable that it will generate a harmonic at 4 kHz as well, and then 8 kHz etc. MAutoEqualizer also contains more advanced parameters, such as harmonics control, to overcome this problem. This feature creates duplicates of the original filters at harmonic positions, behind the scenes This may initially sound complicated, but it's actually fairly simple.

If we recall using the notch filter to remove some noise, using harmonics control you could create the basic notch filter at 2 kHz and then use harmonics control to let MAutoEqualizer place other notch filters at 4 kHz, and 8 kHz etc. As you can see, parametric equalizers may be a little harder to understand, but are essential tools that, once mastered, will provide tremendous results.

They are currently the best compromise between flexibility, speed of work flow and audio quality. Finally there are dynamic equalizers, which extend parametric equalizers by detecting the input level and reacting with band gains in some way. Free form or curve equalizers are designed to let you draw in the frequency response that you want.

So if you need a perfect low-pass filter, draw it and it will be available. At least, that's the theory. MFreeformEqualizer is an example of this type of equalizer.

Other EQs, particularly parametric and multiband EQs, can be used for more precise control. The simplest types of EQs are single-band EQs, which include low cut and high cut, lowpass and highpass, shelving, and parametric EQs.

See Single-Band EQ. Maybe look at different structures like lattice, parallel, state-variable. Different topologies can have limitations that are tricky to understand.

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Learn more. How do software equalizers work? Ask Question. Asked 1 year, 3 months ago. Active 1 year, 3 months ago. Viewed 1k times. Improve this question. Kevin Sullivan. Kevin Sullivan Kevin Sullivan 5 5 bronze badges. I've seen implementations where coefficients for each filter are calculated in real time for each sample before processing the sample which is a real bad decision So, without knowing your implementation it's hard to say if that is your bottleneck.

Show 1 more comment. On the other hand, a female voice starts at Hz and can extend up to 11 kHz. When you look at the lower end frequencies, you would have to use different techniques for the male vocals as they have a lower end. This is the same for instruments as well.

As a music producer learning where each instrument should be in a frequency spectrum will make the process much simpler. This is one of the reasons why experienced producers can make calculated changes to an instrument by just judging the song by ear.

Experience sets the value of a music producer rather than how many hits he has in the music industry. Equalization is one of the areas in music production where without any experience you would feel like you are a fish out of water. Equalization, which started with faders and hardware equipment, has grown so much in the last 10 years.

We have software equalizers that are capable of showing every intricate detail in a song. Equalization first began with the hardware equipment. It slowly dominated the music recording studios in the s and made its way into all hardware sound systems. The hardware equipment also made its way into mixer boards by the end of the s, causing a huge shift in the execution of mixing and mastering in the industry.

Today, hardware equalizers are still used in recording studios for live musical recordings and performances. The advantage of using a hardware equalizer is that it offers the ability to adjust and manipulate a specific group of frequency bands rather than an individual frequency. Hardware instruments add warmth and texture to the track which cannot be attained using a software instrument.

Hardware instruments are easy to handle when compared to software instruments as they have fewer options and functions to deal with. When you look at all the available hardware equalizers, there are usually only three to four knobs.

Generally, you would have a knob around to Hz to control the low end. The next knob would be for the mid frequencies around kHz and there should be another one for the higher frequencies from 15 kHz. Some of the more advanced hardware comes with 6 to 7 knobs that control different bandwidths as well. There is always a question about whether you should use a hardware or software equalizer and the answer is pretty simple when you look at what software equalizers offer.

Even though using hardware equalizer will offer warmth and more tonal harmonies, software equalizers make up for this in the manipulation abilities they offer. Software equalizers were first brought into audio production software around and made a lot of progress in the 10 years that followed. These innovative modulations changed the way equalizers are used by producers in the studio and in audio production software.

The introduction of side chaining in the equalizer made all the difference in how the low end in songs was optimized. The thumping effect of the bass that we hear in songs is produced with the help of side chaining with the equalizer.

The modulation, such as the filters which were introduced in the software equalizers, lead to advanced manipulation of instruments and vocals. The ability to make a choir out of even a single vocal was made possible with such filters in audio production software.

Multiband equalization which was introduced in the software equalizers also makes a great difference in how the songs are manipulated in the audio production software. This is a great change in the way equalization has been done throughout the years. The advantage of using a software equalizer as opposed to a hardware equalizer is justified here.

They always had to use separate analog hardware to route the compression. This was eliminated with the introduction of software equalizers. Software equalizers ensured that the compressors worked even inside the software.

Even hybrid software equalizers with compression have been developed. Multiband compression and equalization techniques were introduced to the industry as separate techniques used in both compression and equalization.

Software companies brought them together. Multiband compression along with equalization reduced the problems faced during the shift from analog to the digital software. The warmth that was missing in the software instruments was restored. There are some guidelines that you can follow to make sure that you minimize errors when you are doing your first equalization project. When it comes to vocals, you should avoid making massive changes, and rather focus on the low-end roll-off.

The audio visualizer in the equalizer will show you the frequencies of the particular vocal. Simply put, roll off is cutting down everything below the starting point of the vocal. Some male vocals start at Hz and some female vocals start at Hz. This completely depends on the person who is singing.