When studying and practicing music production or audio engineering, you will definitely come across high-pass filters. High-pass filters are powerful tools that are used in equalization and in general audio design.
What is a high-pass filter? A high-pass filter (HPF) “passes” the high-frequencies above their cutoff frequency while progressively attenuating frequencies below the cut-off frequency. In other words, high-pass filters remove low-frequency content from an audio signal below a defined cut-off point.
In this article, we’ll have a detailed look at high-pass filters, covering how they work, how they’re designed and how they’re used, not only in EQ but in other applications pertaining to audio as well.
The study of electronic filters goes deep. In this article, we’ll focus on high-pass filters in the context of audio as much as possible in an attempt to keep the post at a reasonable length. Please refer to the table of contents to jumps around to the information you need.
Table Of Contents
- What Is A High-Pass Filter?
- Analog Vs. Digital High-Pass Filters
- Active Vs. Passive High-Pass Filters
- Mixing With High-Pass Filters
- Other Uses Of High-Pass Filters In Audio
- Related Questions
What Is A High-Pass Filter?
The introductory answer paragraph gave us the gist of what a high-pass filter is in the context of audio. However, there’s much, much more to learn about this kind of filter. Let’s begin with the basics.
Thus far we know that a high-pass filter “passes” high frequencies above a certain cutoff point and attenuates of filters out frequencies below this point. High-pass filters are sometimes referred to as low-cut filters and both titles refer to how the filter affects the frequencies of the signal.
The Ideal High-Pass Filter
In an ideal world, a high-pass filter would cut all frequencies below a defined cutoff frequency and allow all frequencies above the cutoff to pass completely unaffected.
This ideal filter is sometimes called a “brickwall” filter and is unobtainable in practice. In theory, an ideal high-pass filter would resemble the following frequency-amplitude graph:
In this graph (and many of the graphs to come), we have frequency along the x-axis (measured in Hertz or “Hz”) and relative amplitude along the y-axis (measured in decibels or “dB”).
Frequency is measured in Hertz, which refers to cycles per second. The universally-accepted range of human hearing is defined within 20 Hz and 20,000 Hz. Therefore many audio mixes have content within this range. Of course, audio signals (which are AC voltages in analog form and binary representations in digital form) can have frequency content outside of this range. However, infrasound (< 20 Hz) and ultrasound (<20,000 Hz) frequencies will be imperceivable upon playback and are typically removed.
Relative amplitude is measured in decibels (tenths of a Bel). This relative unit of measurement expresses the ratio of one amplitude to another on a logarithmic scale. When it comes to signal amplitude, a 3 dB difference will produce a doubling/halving of power quantities (power and ultimately sound intensity). A 6 dB difference will produce a doubling/halving of root power quantities (voltage/current and ultimately sound pressure level).
In the graph above, which shows an ideal high-pass filter, we have a cutoff frequency at 1 kHz (1,000) Hz. All frequencies above 1 kHz are passed perfectly with no alteration and all frequencies below 1 kHz are completely eliminated from the output.
Though impossible to achieve by analog or digital means, there are ways to approximate this type of ideal/brickwall high-pass filter.
With analog HPFs, increasing the filter order will move us closer to the steepness of an ideal filter around the cutoff frequency.
In digital LPFs can also be programmed to approximate such an ideal “brickwall” filter.
Real World High-Pass Filters
Unlike the ideal HPF, real-world high-pass filters will have some sort of transition range where the attenuation will roll-off below the cutoff frequency.
A typical high-pass filter can be easily visualized in the following EQ chart:
Let’s note a few things as we look at the image above.
The first thing to notice is that the HPF will attenuate frequencies below a certain point by a steady slope. As the frequency goes down below the cutoff, so too does the amplitude.
This cutoff frequency, denoted by fC, happens at the -3 dB point and not at the exact point where attenuation begins. This -3 dB cutoff frequency is standard across nearly all filters including, of course, high-pass filters.
Remember our previous discussion on decibels. The -3 dB cutoff frequency point represents the point at which the power of the signal in reduced by half.
HPF Passband, Stopband & Transition Band
Now that we know that real HPFs have a transition band and not only a passband and stopband, we should probably discuss these three distinct frequency ranges/bands.
Starting with the passband, a high-pass filter will technically have a passband from its cutoff frequency to infinity. Of course, this is largely constrained by the components of a circuit of the limits of a digital sample rate. It’s also typical that an audio signal will not have much information above 20 kHz (if at all).
The passband is effectively the band that gets passed through without attenuation. That being said, the typical HPF will have some affect on the signal amplitude and phase in the passband near the cutoff frequency.
The stopband of a high-pass filter will be at some point below the cutoff frequency once the attenuation reaches a sufficient point. This can often be defined as -50 dB point but other attenuation points can also be used depending on the application. Technically speaking, the stopband will extend to 0 Hz at the low-point.
HPFs will generally have a transition band between the stopband and passband as the filter rolls off the amplitude below the cutoff frequency. The bandwidth of this transition range depends upon the slope of the roll-off (determined by the type of HPF and the order of the filter) and the stopband attenuation threshold.
High-Pass Filter Order
Electronic filters are often described by their order. High-pass filters are relatively simple filters and their order defines the slope of their transition bands (also known at the roll-off rate).
The order of a filter is a positive integer. With analog HPFs, the order is technically defined as the minimum number of reactive elements (namely capacitors) required by the filter circuit. Note that, although capacitors are most common, inductors may be found in some passive HPFs.
With the standard Butterworth HPF, each integer increase in order increases the steepness of the roll-off rate by an additional 6 dB/octave or 20 dB/decade.
With digital high-pass filters, order still plays a role in determining the slope of the filter though it obviously doesn’t count the number of reactive components in the circuit (since there are none).
An octave is defined as a doubling (or halving) of frequency and a decade is defined as a tenfold increase (or decrease) in frequency.
An increase in order in any HPF type will effectively steepen the roll-off rate.
In the image below, I’ve presented 5 different Butterworth high-pass filters with orders 1 through 5:
While the cutoff frequency (-3 dB point) of each filter is matched at 1 kHz, we can see that the roll-off rate steepens as we move from the first-order filter (red/1) through to the fifth-order filter (pink/5).
We can see that as the order increases, the low-pass filter gets closer to becoming an ideal filter and that the transition band becomes narrower (let’s take the -50 dB threshold).
High-Pass Filter Q Factor
Note that some high-pass filters will have a Q factor control. This is particularly the case with parametric EQ plugins and digital EQ units, where the filter is not designed as any particular type (Butterworth, Bessel, Chebyshev, Elliptic, etc.).
Though the Q factor does have a technical definition, it’s often ignored, making the parameter somewhat arbitrary. Many manufacturers will have their own “version” of Q.
That being said, in general, increasing the Q factor of a HPF will steepen the roll-off slope while causing a resonance peak to form at and below the cutoff frequency.
Conversely, decreasing the Q factor of a HPF will increase the attenuation at and below the cutoff frequency while making the roll-off slope more gradual.
The EQs that will offer a Q factor control on the high-pass filter will typically have a graphic to show you, graphically, how the filter is affecting the signal so as to avoid confusion.
High-Pass Filters & Phase-Shift
It’s important to note here that high-pass filters do not only affect the amplitude of the signal but also affect the phase of the signal. In most filters (and all analog filters), there will be some amount of frequency-dependent phase-shift between the input signal and its output signal.
Generally speaking, each reactive component in an analog filter will introduce 90º of phase shift in the signal. For analog high-pass filters (and the digital filters that aim to recreate them in the digital realm), this means that there will be 90º of phase-shift per integer increase in filter order.
With standard Butterworth high-pass filters, half of the total phase-shift will happen by the cutoff frequency. If the total phase shift is 90º (like a first-order filter), the phase-shift at the cutoff frequency will be 45º. If the total is 360º, the cutoff frequency will have 180º of phase-shift.
Here is a visual representation of a first-order Butterworth high-pass filter with both amplitude-frequency and phase-frequency graphs:
In the section on digital high-pass filter, we’ll discuss linear phase EQ/filters that are designed to eliminate phase-shift while maintaining the same amplitude filtering of typical high-pass filters.
Analog Vs. Digital High-Pass Filters
As their names would suggest, analog HPFs filter analog audio signals and are designed with analog components (resistors, capacitors, operational amplifiers, etc.) while digital HPFs filter digital audio signals are either embedded in digital chips or programmed into software/plugins.
Let’s have a more thorough discussion on both analog and digital high-pass filters in this section.
Analog High-Pass Filters
Analog filters are relatively simple to understand compared to their digital counterparts. A basic understanding of electrical circuits and mathematics should be able to get us through a lesson on analog audio high-pass filters.
This is not necessarily the case for digital HPFs. In fact, digital circuits and coding are beyond my abilities and so I won’t go into too much detail there. Rather, I’ll explain how analog high-pass filters are designed and how they work.
Many digital high-pass filters are simply designed to recreate the effect of their analog counterparts, anyway, so an understanding of analog HPFs should be somewhat translatable, at least for general knowledge.
With that said, let’s get into analog HPFs. I’ll add plenty of equations and diagrams to help with the explanation.
To really understand the basics of how a high pass filter works, we can study a simple passive first-order RC (resistor-capacitor) LPF. This filter can be visualized with the following image:
The circuit above can be thought of similarly to a voltage divider:
In the above schematic, we derive the following formula:
With a voltage divider, we have a situation where, as R2 increases, Vout increases (assuming R1 remains constant). Remember this as it will translate to the simple RC high-pass filter circuit.
If we were to simply swap the resistors of the DC voltage divider with the components of the RC HPF circuit, then R1 would become the capacitor and R2 would become the resistor.
When dealing with simple DC voltage, we only need to be concerned with electrical resistance. When we move to AC voltage (which is how analog audio signals are represented), we must use impedance, which is a combination of DC resistance and AC reactance.
So if we have, let’s say, an audio signal at Vin with frequency content between 20 Hz and 20,000 Hz (the human range of hearing), then we have an AC signal. AC signals are subject to impedance, which has both phase and magnitude and is made up of the resistance and reactance of a circuit.
In an ideal world (which we’ll use to understand RC low-pass filters), the reactance of a resistor is zero and the resistance of a capacitor is zero. Therefore the resistance component of the circuit impedance will be solely from the resistor. Similarly, the reactance component of the circuit impedance will be solely from the capacitor.
Let’s revisit our simple passive analog RC high-pass filter schematic:
From this circuit diagram, we can swap some variable of the voltage divider equation to come up with the following:
• Z is the overall impedance of the circuit
• R is the resistance of the resistor
Remember that the impedance is made of the resistance and reactance components of the circuit. The typical impedance formula is:
Where XL is the inductive capacitance. Because there is no inductor in the RC circuit, XL is equal to zero.
Let’s quickly rewrite our RC output voltage with this new information:
Look familiar? It’s nearly the same as the simple voltage divider.
Alright, so we can effectively compare the simple voltage divider to a simple first-order RC high-pass filter. Assuming R remains constant (a safe assumption), it’s the capacitive reactance (XC) of the capacitor that will affect the Vout as compared to the Vin. More specifically, as the XC increases, the Vout will decrease.
How does this apply to a high-pass filter? Well, the reactive capacitance of the capacitor decreases as the frequency of the input signal increases. The formula for this is as follows:
• f is the frequency of the signal
• C is the capacitance of the capacitor
So with that we have the following rules of the RC high-pass circuit:
- As the frequency increases, the capacitive reactance decreases
- As the capacitive reactance increases, the output signal level decreases relative to the input signal level (assuming the resistance of the circuit remains the same)
As the capacitive reactance increases (as the frequency decreases), more of the signal is sent to ground rather than to the output.
So then, as frequencies get lower, the high-pass filter circuit will attenuate the output more and more.
There is a point (the passband) at which any increase in frequency will not increase the output and the Vout will be (ideally) equal to the Vin.
The cutoff frequency, as we’ve discussed before, is the point at which the high-pass filter attenuates the signal by 3 dB and anything below the cutoff frequency is part of the transition band and, eventually, the stopband (below a certain attenuation threshold).
The equation of the cutoff frequency in our simple RC high-pass filter is:
• R is the resistance of the resistor
• C is the capacitance of the capacitor
As an additional equation, we can calculate the aforementioned phase-shift of an RC high-pass filter with the following equation:
That’s about as simple as it gets. I hope it all makes sense.
Analog high-pass filters can get more and more complex as the order increases and active components (such as op-amps) are included. The higher the order, the closer the filter performance gets to that of the “ideal filter”. Note that many digital filters (including EQ plugins) emulate these analog filters.
Increasing the order steepens the roll-off rate and shortens the effective transition band.
There are plenty of filter types to be aware of. Thus far, we’ve focused largely on the popular Butterworth filter. However, there are 3 main filter types (among the many) that we should be aware of when it comes to audio. They are:
- Butterworth filter
- Bessel filter
- Chebyshev filter
These “types” of filters are dependent on the values of the components used in the filter design and the damping factor that comes with the filter design. The study of individual high-pass filter schematics is beyond the scope of this article but these popular types are worth knowing about.
What is a Butterworth filter in audio? A Butterworth filter (maximally flat magnitude filter) is a linear analog filter designed to have a frequency response as flat as possible in the passband. Butterworth filters do not offer an overly steep roll-off and are often used in low/high-pass and low/high shelf filters.
What is a Bessel filter in audio? A Bessel filter is a linear analog filter with a maximally flat group or phase response to preserve the wave shapes of signals within the passband. Bessel filters provide a gentle frequency roll-off beyond the cutoff frequency and are mainly designed for linear phase response with little overshoot.
What is a Chebyshev filter in audio? A Chebyshev filter is a linear analog filter designed to have a very steep roll-off at the expense of passband ripple (type I) or stopband ripple (type II/inverse).
Note that the elliptic filter (also known as a Cauer filter) is a linear analog filter with equalized ripple in both the passband and the stopband. It offers a very steep transition band. It is achieved by combining a high-pass filter and a band-stop/notch filter.
Digital High-Pass Filters
Digital signal processing (DSP) and the continual improvement of computer processing power has made it possible to make very capable and flexible digital high-pass filters. These HPFs are typically much more precise and versatile that their analog counterparts.
They’re also typically more cost effective (especially in the form of software) and are less effected by temperature and humidity.
Some digital HPFs are designed to emulate analog LPFs (only on digital signals). In this case, we’ll often find Butterworth, Bessel, Chebyshev and other HPF filter types in digital processors.
These digital filters, as mentioned, can be coded into software/plugins or embedded in digital chips. They do not have analog components.
A digital low-pass filter will fit into one of two camps:
- Infinite Impulse Response (IIR)
- Finite Impulse Response (FIR)
What is an infinite impulse response filter in audio? An IIR filter is a linear time-invarient analog type of filter (that has been digitized as well) that works with an impulse response that continues indefinitely, never becoming exactly zero. Butterworth, Chebyshev, Bessel and elliptic filters are examples of IIR filters.
What is a finite impulse response filter in audio? An FIR filter is a filter (analog or digital, though nearly always digital) that works with an impulse response of finite duration, settling to zero within some amount of time. It lends itself well to linear phase EQ.
Speaking of linear phase EQ, these specialized equalizers are worth mentioning here as well.
A linear phase EQ (which will almost certainly always have high-pass filter options) effectively eliminates any phase-shift within the audio processor.
Recall in the section High-Pass Filters & Phase Shift how we discussed the inevitable phase-shift of analog LPFs (90º of phase-shift for every reactive component in the circuit).
A linear phase EQ (and high-pass filter) uses digital signal processing (DSP) to analyze the frequency content of a signal and apply gain to the appropriate frequencies via FIR (finite impulse response) filters in order to eliminate any phase-shifting that arises.
The Fabfilter Pro-Q 3 EQ (link to check the price at Plugin Boutique) is an EQ plugin that has a superb linear phase mode:
Recap On Analog & Digital High-Pass Filters
Here’s a short table to recap what we’ve discussed in this section.
|Analog Audio HPF||Digital Audio HPF|
|Filters analog (continuous-time) audio signals||Filters digital (discrete-time) audio signals|
|Made of analog components||Embedded in digital chips (with adders, subtractors, delays, etc.), or;
Coded into software
|Limited in functionality & adaptability||More versatile in programming|
|More sensitive to environmental changes||Less sensitive to environmental changes|
|Analog components introduce thermal noise||Quantization introduces digital noise|
|Higher manufacturing cost||Lower manufacturing cost|
Active Vs. Passive High-Pass Filters
Another big categorical difference we’ll find with high-pass filters is that of active filters and passive filters. Active filters have active components (such as op-amps) and require power to function while passive filters only have passive components and do not require power.
Labelling a HPF as active or passive typically only applies to analog filters. Digital filters are, by design, always active, whether they’re embedded in hardware chips or coded into software.
Though most audio EQs (and high-pass filters by extensions) will be active units, it’s worth mentioning the differences between the two.
Passive High-Pass Filters
We’ve discussed the basic passive RC high-pass filter at length in the section on analog high-pass filters. Therefore, we should already have a good understanding of such circuits.
Allow me to repost the basic schematic of a first-order passive RC high-pass filter:
As previously discussed, we can effectively steepen the roll-off rate of the passive HPF by increasing the order (adding capacitor-resistor pairs) of the circuit. However, with no amplification, this will come at the expense of losing signal amplitude across the entire frequency spectrum. It will also come at the cost of a worsened signal-to-noise ratio as the signal must pass through more components without any buffer or amplifier.
That being said, HPFs don’t necessarily require any amplification. After all, they’re only tasked with cutting frequencies and not boosting frequencies. Including active components is likely to improve performance and is, therefore, common practice in audio high-pass filter design.
Passive high-pass filters are still found in certain applications and there are even passive EQ units on the market that, by definition, will have passive HPFs (if they include a high-pass filter).
Note that, with passive EQs, there is an amplification stage for “makeup gain” after the filter circuit(s). It’s just that there are no active components within the filter circuit(s).
Active High-Pass Filters
Indeed, most audio high-pass filters are active.
Analog active HPFs typically use operation amplifiers in their designs. Op-amps can be used in unity gain filters to maintain signal amplitude without increasing signal amplitude. They can also be used to offer gain to the circuit.
By including an active op-amp, HPF designers can steepen the roll-of slope without losing overall signal amplitude at the output.
In addition to gain, op-amps greatly improve the output/source impedance of the filter.
Op-amps can effectively bridge impedances at the output of the filter so that the filter can properly drive the load (the input of the following audio device). Op-amps can also be used within higher-order filters to buffer the signal within the circuit, thereby improving the signal-to-noise ratio, gain and overall performance.
Here’s an example of an active first-order RC high-pass filter with unity gain:
Note that it looks very similar to the aforementioned passive RC filter. The main difference being, of course, the op-amp. In this case, the op-amp does not offer any amplification to the signal. Rather, it maintains unity gain and allows for an appropriate output impedance for the high-pass filter circuit.
Now let’s have a look at a look at a simple first-order RC high-pass filter that does offer amplification:
The gain AV of the non-inverting amplifier is calculated by the following equation including the feedback resistor (R2) and its corresponding input resistor (R1):
The gain of the overall circuit is frequency-dependent (as the high-pass filter attenuates lower frequencies). This gain can be defined with the following equation:
With this equation, we can observe the following:
- At low frequencies (f < fC): A = Vout/Vin < AV
- At the cutoff frequency (f = fC): A = Vout/Vin = AV/√2 = 0.707 AV
- At high frequencies (f > fC): A = Vout/Vin ≈ AV
If plug 0.707 AV into the following equation for decibels, we can confirm that the cutoff frequency is indeed at -3 dB from unity:
If we look at a second-order active high-pass filter (in a simplified schematic), we’d have the following:
When dealing with second-order filters (and higher), we have a damping factor in the circuit. The damping factor of this simple Sallen-Key filter topology is:
So the RF and RI values are involved in determining the gain and damping factor of the circuit. The RF and RI also determine whether we have a Butterworth, Bessel or Chebyshev filter. Note that the following only applies to a second-order filter:
- RF / RI = 0.586
- DF = 1.414
- AV = 4 dB
- RF / RI < 0.586
- DF > 1.414
- AV < 4 dB
- RF / RI > 0.586
- DF < 1.414
- AV > 4 dB
Let’s now have a look at a sixth-order RC high-pass filter below:
In this case, we can see that there are 3 op-amps (1 for every RC pair). These ope-amps help to buffer the circuit and maintain proper gain staging throughout the filter circuit. By buffering within the circuit, we maintain proper impedance bridging between the pairs of RC components. Adding gain (and damping factor) with each op-amp helps maintain signal strength and a proper signal-to-noise ratio.
With a sixth-order setup, the roll-off slope of this high-pass filter would have a slope of 36 dB/octave or 120 dB/decade below the cutoff frequency (assuming it’s a Butterworth filter).
The HPF could take on the Butterworth, Bessel, Chebyshev or any other possible low-pass filter “type” given the topology. The various RF / RI ratios between the 3 sets would be different than those defined above for the second-order filter.
This is about the extent that we’ll go to in this article. There are plenty of more in-depth articles (and electronic engineering courses) on filter design. The information shared here is more than enough to cover the basics of HPFs used in audio.
Recap On Active & Passive High-Pass Filters
Here’s a short table to recap what we’ve discussed in this section.
|Active Audio HPF||Passive Audio HPF|
|Requires power||Does not require power|
|Includes active and passive components (including op-amps)||Only includes passive components (resistors, capacitors, etc.)|
|Offers amplification above unity gain (boosts in addition to cuts)||Cannot offer amplification above unity gain (cuts only)|
|Low output impedance (load-independent performance)||Higher output impedance (load-dependent performance)|
|Higher manufacturing cost||Lower manufacturing cost|
Additional points could be made that aren’t as applicable to audio HPFs (as they are to very high frequency signals) but are worth mentioning anyway:
- Passive HPFs can include inductors.
- Active HPFs cannot handle as high of signal amplitudes as passive filters due to the op-amp.
- Active HPFs have limited bandwidth due to the op-amp.
Mixing With High-Pass Filters
Now that we understand what high-pass filters are and how they work, let’s have a look at how they’re used in the context of audio mixing.
High-pass filters are used for mixing in the following ways:
- Filtering Out Low-End Rumble/Cutting Problem Frequencies
- Reduce Competition Between Instruments In The Low-End
- Accentuating Fundamentals Of Percussion With Resonance
Filtering Out Low-End Rumble/Cutting Problem Frequencies
High-pass filters are invaluable tools when it comes to removing unneeded and oftentimes detrimental low-end information from tracks in a mix.
Low-end rumble can come from traffic, HVAC, construction, banging and other vibrations in the studio. Other low-end issues can arise from electromagnetic noise and interference in the audio system.
Feedback can also be an issue in the low end as can resonances within the acoustic environment.
This noise is non-musical and can ruin an otherwise great mix.
With high-pass filters, we can effectively filter out the non-musical low-end noise, thereby improving the headroom of the overall mix and clearing out the low-end frequency band(s) for sources/instruments that actually have important information (kick drums, bass guitar, tubas, etc.)
Reduce Competition Between Instruments In The Low-End
This ties in nicely with the previous point. By high-pass filtering instruments that have nothing to offer in terms of musical/important low-end information, we clean up a frequency band for those instruments that do offer low-end information.
Accentuating Fundamentals Of Percussion With Resonance
It may seem counter-intuitive that a cutting filter could accentuate frequencies.
For this mixing technique, we’ll need a high-pass filter than offers some amount of resonance or ripple near its cutoff frequency. If we have this resonance, we can effectively boost the fundamental of a signal (kick drum, snare drum, tom drum, etc.) while removing all the non-musical information below the fundamental.
Automating a high-pass filter can be used to great effect in creating sonic interest in a sound source.
If you’re into synthesizers, you’re likely aware of how automating or otherwise modulating the high-pass filter can generate cool results.
Envelope filter effects pedals can also modulate a high-pass filter to achieve their sonic effect, especially when there’s a resonance peak near the cutoff.
When it comes to standalone high-pass filters, we can extend these effects to any sound source by automating the high-pass filter (particularly the cutoff frequency parameter).
We can also use automation to effectively increase or decrease the perceived depth of the track and also to reduce competition in the low-end as other tracks are introduced to (or taken out of) the arrangement.
For more information on envelope filters, check out my article What Are Envelope Filter Effects Pedals & How Do They Work?
Other Uses Of High-Pass Filters In Audio
High-pass filters are used in more than just mixing when it comes to audio technology. In this section, we’ll have a look at other applications for high-pass filters.
Other uses of high-pass filters include:
- Pre-Emphasis Filters
- Loudspeaker Crossovers
- Feedback Control
- Inclusion In Band-Pass Filters
- Inclusion In Band-Stop Filters
- Microphone Switches
Pre-emphasis filters are used in systems where pre-emphasis and de-emphasis are needed for improved signal transfer. These applications are, most notably, FM radio and vinyl recording/playback.
Pre-emphasis filters are generally high-pass, low shelf cut or high shelf boost filters. They are used in order to improve signal-to-noise ratio in the high-end (with FM radio) or improve storage (vinyl is notoriously poor at storing low-end information in its grooves).
A de-emphasis filter is then needed at playback to undo the effect of the pre-emphasis filter, bringing the signal back to its original frequency response.
To help visualize, here is an image of a pre-emphasis filter (in pink) and de-emphasis filter (in blue) for FM radio (time constant of 75 µs and a cutoff frequency of 2,122 Hz):
Similarly, the RIAA equalization standard is a pre/de-emphasis EQ for the recording and playback of phonograph/vinyl records. It is represented by the image below with the blue line representing the playback (de-emphasis) EQ and the pink line representing the recording (pre-emphasis) EQ:
Individual loudspeaker drivers (the speaker units themselves) are notoriously bad at reproducing the entire audible range of human hearing. This is why loudspeakers and studio monitors will often come with at last two speakers (generally a mid-range woofer and a tweeter, if not more).
In larger systems, there is generally a subwoofer (or multiple) that are able to effectively reproduce the low-end information.
In order for each driver to work optimally, each drive should only be sent the frequency band in which it’s designed to reproduce as a transducer. Crossovers do just that, splitting the incoming audio signal into different bands in order to properly drive the individual speaker drivers.
So then, to split the audio signal sent to the main speakers versus the subwoofer, a high-pass filter can be used.
Similarly, to split the audio signal sent to the tweeters versus the other speakers, a high-pass filter can be used.
Feedback is always a possibility when microphones and loudspeakers are used in the same system.
The low-end resonances of an acoustic environment can be a liability when it comes to feedback.
Low-end frequencies are more likely to cause feedback as they vibrate the solid structures of the environment. A subwoofer, for example, can vibrate the stage and if the microphones on the stage are not properly isolated, they can pickup this low-end vibration and cause a low-end feedback loop.
One way to combat this is to simply high-pass the microphone signals that aren’t positioned to pickup low-end frequencies.
For more information on reducing microphone feedback, check out my article 12 Methods To Prevent & Eliminate Microphone/Audio Feedback.
Inclusion In Band-Pass Filters
What is a band-pass filter in audio? A band-pass filter “passes” a band of frequencies (a defined range above a low cutoff and below a high cutoff) while progressively attenuating frequencies below the low cutoff and above the high cutoff.
Bandpass filters can be thought of as high-pass and low-pass filters in series/cascade. The high-pass filter cutoff frequency (fH) will be lower than the low-pass filter cutoff frequency (fL).
Here’s a visual representation of a bandpass filter frequency plot:
And here’s a simplified schematic representing an analog bandpass filter with a first-order high-pass and first-order low-pass filter:
Inclusion In Band-Stop Filters
What is a band-stop filter in audio? A band-stop filter (aka a notch filter or band-reject filter) works by removing frequencies in a specified band within the overall frequency spectrum. It allows frequencies below the low cutoff point to pass along with frequencies above the high cutoff point.
Bandstop filters can be thought of as high-pass and low-pass filters in parallel. The high-pass filter cutoff frequency (fH) will be greater than the low-pass filter cutoff frequency (fL).
Here’s a visual representation of a bandstop filter frequency plot:
And here’s a simplified schematic representing an analog bandstop filter with a first-order high-pass and first-order low-pass filter:
Some microphones have built-in high-pass filter switches that can be toggled on and off.
By engaging the HPF at the microphone when the application calls for it, we can record a cleaner direct signal and rely less upon other processes in the signal chain.
Microphone high-pass filters are often used for the following reasons:
- To rid of low-end rumble and noise in the signal
- To help reduce the proximity effect within directional microphones
- To help reduce plosives in directional microphones
- To remove unnecessary low-end so that a microphone signal may better fit within a mix
The AKG C 414 XLII (link to check the price on Amazon) is an example of a microphone that has 3 different high-pass filter switches (along with an option to disengage any filtering). These high-pass filter options are defined as:
- HPF @ 40 Hz with −12 dB/octave roll-off
- HPF @ 80 Hz with −12 dB/octave roll-off
- HPF @ 160 Hz with −6 dB/octave roll-off
AKG is featured in the following My New Microphone articles:
• Top Best Microphone Brands You Should Know And Use
• Top Best Headphone Brands In The World
What is a low-pass filter in audio EQ? A low-pass filter (LPF) “passes” the low-frequencies below their cutoff while progressively attenuating frequencies above their cutoff. In other words, low-pass filters remove high-frequency content from an audio signal above a defined cut-off point.
Related article: Audio EQ: What Is A Low-Pass Filter & How Do LPFs Work?
What is a shelving EQ? Shelving eq utilizes high and/or low shelf filters to affect all frequencies above or below a certain cutoff frequency, respectively. Shelving can be used to either boost/amplify or cut/attenuate and affects all frequencies equally beyond a certain point.