Audition High Pass Filter



  1. High Pass Filter- Explained. A high pass filter is a filter which passes high-frequency signals and blocks, or impedes, low-frequency signals. In other words, high-frequency signals go through much easier and low-frequency signals have a much harder getting through, which is why it's a high pass filter.
  2. The basic High Pass Filter is built by a series connection of capacitor and resistor. While the input signal is applied to the capacitor, the output is drawn across the resistor. High Pass Filter Circuit.

There was an era where while making a telephone call over distance places, one had to put his mouth very close to the transmitter, speak very slowly and very loudly so that message can be heard clearly by the person at the other end. Today, we can even make video calls over worldwide with high-quality resolutions. The secret of such a tremendous development of technology lies in Electrical filtertheory and Transmission line theory. Electrical filters are circuits that pass only selected band of frequencies while attenuating other unwanted frequencies. One of such filters is High pass filter.

In original, FFR filter and high pass filter has the functionality to get the lower and upper limit of audio frequency input by the user manually. In this research, an effort will be done to make FFR filter and high pass filter adaptive in nature by automatically selecting. A shelving filter, also referred to as a shelf filter, shelf EQ, shelving EQ etc. Allows you to boost or attenuate either the high end or the low end of the frequency spectrum.A shelving filter which boosts or attenuates the high end of the frequency spectrum is known as a ‘high shelf’.

What is a High Pass Filter?

The definition of high pass filter is a filter which passes only those signals whose frequencies are higher than cutoff frequencies thereby attenuating signals of lower frequencies. The value of cutoff frequency depends on the design of the filter.


High Pass Filter Circuit

The basic High Pass Filter is built by a series connection of capacitor and resistor. Download pctel port devices driver. While the input signal is applied to the capacitor, the output is drawn across the resistor.

In this circuit arrangement, the capacitor has high reactance at lower frequencies so it acts as an open circuit to the low-frequency input signals until cutoff frequency ‘fc’ is reached. Filter attenuates all the signals below the cutoff frequency level. At frequencies above cut off frequency level reactance of the capacitor becomes low and it acts as a short circuit to these frequencies thereby allowing them to pass directly to the output.

Passive RC High Pass Filter

The above shown High Pass filter is also known as Passive RC High Pass filter as the circuit is built using only passive elements. There is no need of applying external power for working of the filter. Here capacitor is the reactive element and output is drawn across the resistor.

High Pass Filter Characteristics

When we talk about cutoff frequency we refer to the point in the frequency response of the filter where the gain is equal to 50% the peak gain of the signal .i.e. 3dB of the peak gain. In High Pass Filter gain increases with an increase in frequencies.


This cutoff frequency fc depends on R and C values of the circuit. Here Time constant τ = RC, the cutoff frequency is inverse proportional to the time constant.

Filter

Cutoff frequency = 1/2πRC

Circuit gain is given by AV = Vout/Vin

.i.e. AV = (Vout)/(V in) = R/√(R2 + Xc2) = R/Z

At low frequency f: Xc→∞, Vout = 0

At high-frequency f: Xc →0, Vout = Vin

High Pass Filter Frequency Response or High Pass Filter Bode Plot

In high pass filter, all frequencies lying below the cutoff frequency ‘fc’ are attenuated. At this cut off frequency point we get -3dB gain and at this point reactance of the capacitor and resistor values will be same .i.e. R = Xc. Gain is calculated as

Gain (dB) = 20 log (Vout/Vin)

The slope of high pass filter curve is +20 d B/ decade .i.e. after passing cutoff frequency level the output response of the circuit increases from 0 to Vin at a rate of +20 dB per decade which is 6 dB increase per octave.

The region from the initial point to cutoff frequency point is known as stop band as no frequencies are allowed to pass. The region from above the cutoff frequency point. i.e. -3 dB point is known as the passband. At cutoff frequency, point output voltage amplitude will be 70.7% of the input voltage.

Here bandwidth of the filter denotes the value of frequency from which signals are allowed to pass. For example, if the bandwidth of the high pass filter is given as 50 kHz it means that only frequencies from 50 kHz to infinity are allowed to pass.

The phase angle of the output signal is +450 at the cut off frequency. The formula to calculate the phase shift of high pass filter is

∅=arctan ⁡(1/2πfRC)

In practical application, the output response of filter does not extend to infinity. The electrical characteristic of the filter elements applies the limitation to the filter response. By proper selection of filter components, we can adjust the range of frequencies to be attenuated, the range to be passed etc…

High Pass Filter using Op-Amp

In this high pass filter along with passive filter elements, we add Op-amp to the circuit. Instead of getting an infinite output response, here the output response is limited by open loop characteristics of the Op-amp. Hence this filter acts as a band-pass filter with a cut off frequency which is defined by the bandwidth and gain characteristics of Op-amp.

The open loop voltage gain of Op-amp acts as a limitation to the bandwidth of the amplifier. The gain of the amplifier reduces to 0 dB with the increase in input frequency. The response of the circuit is similar to passive high pass filter but here gain of the Op-amp amplifies the amplitude of the output signal.

The gain of the filter using non inverting Op-amp is given by:

AV = Vout/Vin = (Af (f/fc))/√(1+ (f/fc)^2 )

where Af is passband gain of the filter = 1+( R2)/R1

f is the frequency of the input signal in Hz

Pass

fc is the cut off frequency

When low tolerance resistors and capacitors are used these High Pass Active filters provide good accuracy and performance.

Active High Pass Filter

High Pass Filter using Op-amp is also known as an active high pass filter because along with passive elements capacitor and resistor an active element Op-amp is used in the circuit. Using this active element we can control the cutoff frequency and output response range of the filter.

Second Order High Pass Filter

The filter circuits we saw till now are all considered as first order high pass filters. In second order high pass filter, an additional block of an RC network is added to the first order high pass filter at the input path.

Second Order High Pass Filter

The frequency response of second order high pass filter is similar to the first order high pass filter. But in second order high pass filter stop band will be twice that of first order filter at 40dB/Decade. Higher order filters can be formed by cascading first and second order filters. Although there is no limit to the order, the size of the filter increases along with their order and accuracy degrades. If in higher order filter R1 =R2= R3 etc… and C1=C2=C3= etc…then cutoff frequency will be same regardless of the order of the filter.

The cutoff frequency of second order High Pass Active filter can be given as

fc= 1/(2π√(R3 R4 C1 C2 ))

High Pass Filter Transfer Function

As the impedance of the capacitor changes frequently, electronic filters have a frequency-dependent response.

The complex impedance of a capacitor is given as Zc=1/sC

Where,s = σ + jω, ω is the angular frequency in radians per second

The transfer function of a circuit can be found using standard circuit analysis techniques such as Ohm’s law, Kirchhoff’s Laws, Superposition etc. The basic form of a Transfer function is given by the equation

H(s) = (am s^m+a(m-1) s^(m-1)+⋯+a0)/(bn s^n+b(n-1) s^(n-1)+⋯+b0 )

The order of the filter is known by the degree of the denominator. Poles and Zeros of the circuit are extracted by solving roots of the equation. The function may have real or complex roots. The way these roots are plotted on s plane, where σ is denoted by the horizontal axis and ω is denoted by the vertical axis, reveals lots of information about the circuit. For high pass filter, a zero is located at the origin.

Adobe audition high pass filter

H( jω) = Vout/Vin = (-Z2 (jω))/(Z1 (jω))

= – R2/(R1+ 1/jωC)

= -R2/R1 ( 1/(1+ 1/(jωR1 C))

Here H(∞) = R2/R1 , gain when ω →∞

High Pass Filter Audition

τ = R1 C and ωc= 1/(τ ) .i.e. ωc= 1/(R1C ) is the cut-off frequency

Thus the transfer function of high pass filter is given by H( jω) = – H(∞)( 1/(1+ 1/jωτ))

= – H(∞)( 1/(1- (jωc)/ω))

When input frequency is low then Z1 (jω) is large, therefore the output response is low.

H( jω) = (- H(∞))/√(1+(ωc/ω)^2 ) = 0 when ω=0; H(∞)/√2 when ω=ω_c;

and H(∞) when ω=∞. Here negative sign indicates phase shift.

When R1= R2, s = jω and H(0) = 1

So, the transfer function of High Pass Filter H(jω) = jω/(jω+ ω_c )

Butter worth High Pass Filter

Besides being rejecting the unwanted frequencies, an ideal filter should also have uniform sensitivity for wanted frequencies. Such an ideal filter is impractical. But Stephen Butter worth in his paper “On the theory of filter amplifiers” showed that this type of filter can be achieved by increasing the number of filter elements of right magnitudes.

Butter worth filter is designed in such a way that it gives flat frequency response in the passband of the filter and decreases towards zero in the stop band. A basic prototype of Butter worth filter is the low pass design but by modifications high pass and band pass filters can be designed.

As we have seen above for a first order high pass filter unit gain is H(jω) = jω/(jω+ ω_c)

For n such filters in series H (jω) = (jω/(jω+ ω_c ))^n which upon solving equals to

‘n’ controls the order of transition between pass band and stop band. Therefore higher the order, rapid the transition so, at n = ∞ Butter worth filter becomes an ideal High Pass Filter.

During the implementation of this filter for simplicity we consider ωc = 1 and solve the transfer function

for s = jω .i.e. H(s) = s/(s+ωc ) = s/(s+1) for order 1:

H(s) = s^2/(s^2+∆ωs+(ωc^2 ) for order 2

Therefore the transfer function of the cascade in High Pass Filter is

Applications of High Pass Filter

The high pass filter applications mainly include the following.

  • These filters are used in speakers for amplification.
  • High pass filter is used to remove unwanted sounds near to the lower end of the audible range.
  • To prevent the amplification of DC current that could harm the amplifier, high pass filters are used for AC-coupling.
  • High Pass filter in Image Processing: High pass filters are used in image processing for sharpening the details. By applying these filters over an image we can exaggerate every tiny part of details in an image. But overdoing can damage the image as these filters amplify the noise in the image.

There are still lots of developments to be made in the design of these filters to achieve stable and ideal results. These simple devices play a significant role in variouscontrol systems, automatic systems, Image and audio processing. Which of the application of High pass filter have you come across?

Noise reduction & restoration

De noise:

simple just move the slider to 0-100% to reduce amount of sound.
Enable output Noise only: listen only the noise level useful to cut the noise.
Then apply gain to get actual value from reduced lower sound.
Pros: simple just like noise reduction technique in other softwares like audacity.

Adaptive noise reduction: (more parameters)

standard noise reduction only for post processing, not for live processing, so adaptive noise reduction must for live podacasts or streaming games.
Note: it’s realtime processing so need little bit extra cpu power otherwise delay due to heavy processing.
Reduce Noise By: 6-30db better, 20db by default,
Just it substract the noise. (you can monitor it while adjusting)
Noiseness: it’s indicates amount of noise in original audio.(so we can reduce accordingly)
Fine tune noise floor: manually adjusting it above or below the automatically calculated noise floor.
Signal threshold: manually adjusting desried audio level above or below the automatically calculated threshold.
Special decay rate: how quickly drops to 60db
Baseband preservation:
Saving frequency of vocals helps to boost audio quality. (Need to know about equalization)
FFT SIZE: length of frequency band. Or how many individual frequency bands are analyzed.
Processing focus:
High frequency focus
Mid frequency focus
  • Techniques for restoring audio
  • Sound Remover effect
  • Automatic Click Remover effect
  • Click/Pop Eliminator effect
  • DeReverb effect
  • Hiss Reduction effect (Waveform Editor only)

Fliter and Equalizer

FFT Filter Effect:

Fast Fourier Transform an algorithm quickly analyzes frequency and amplitude of sound wave.
High pass filter(HP): Passes the high frequency and cuts the lower frequencies.
Low pass filter(LP): passes the lower frequency and cuts the higher frequencies.
Narrow band pass filter/ telephony : 350-4000hz here only the vocals of human to avoid background noise. Telephone uses this spectrum.
Notch filter: to remove precise frequency low 60hz, monitor it just up& down , find a noise in a specific range, then use notch filter and drag drop to remove entire sound.
Tradeoff between frequency and time accuracy.
Lower reduced transient artifacts

Parametric Equalizer in Adobe audition

Q/width: controls width of the effected frequency,
High q value effects low range frequencies (ex 100hz)
Q value which is a ratio of width to center frequency

Amplitude & compression AU

Noise gate effect: stops the noise below noise threshold (ex threshold,-30db then it only passes above threshold db like -29 )
Expander: just like compressor, but it compress the lower noise ex: threshold -30 db compress ratio 1:2
Then 30db becomes 15db if ratio 1:10 30-3db= 27db.
compressor: compress the above limit threshold .
Then compressor threshold limit is -10
attack time: wait before compressing after reaching threshold
Ratio = 1:2 , before total db level to -1db. Now it’s -5db.
Because it’s compressed remaining 10db into 1:2 ratio means half then 10 becomes 5.
Gain: after compressing the noise level reduces we can apply some gain to -6,-3 or -1 db.
Normalisation: boosting lower level audio signals upto peak level audio signal.(make sure to completely reduce background noise)
Ex: audio track has -15 db , and -10db – 6-db -3db.
When we apply normalisation . All audio singals gained to upto -3db it’s a peak in audio track.
Before applying normalisation make sure to cut excessive wave lengh. So the audio track is innsame wavelength.
Presets: -1db more than -1db or 0 distortion or in bearble sound.
-6 mostly use for voiceovers
-10db some people use it for quite voice but not recommended.
Equalization: plays vital role to remove unnecessary noise based on frequency.

attack & release time

Determines how many milliseconds it takes for the output signal to reach the specified level
(to avoid unnecessary processing at short sudden changes )
Determines how many milliseconds the current output level is maintained
To see good attack and release times for different types of audio content, choose various options from the Presets menu.
Fast attack & release time for drums, music instruments.
You can know by just doing post processing audio. To see how much time the audio at peak level.(it’s a art)
Hold time:

Dynamic processing

Level meter & gain reduction meter:
This works by compressing lower & higher audio signals. To maintain a level.
(Noise reduction done by expander, loudness reduced by compressor, also added bandwidth/frequency specific processing)

Dynamic effects:

Compressor: reducing peak voice with certain ratio
Expander: reducing lower threshold with certain ratio
Limiter: totally reducing noise level for a specific threshold.

Hard limiter effect:

Maximum amplitude:
Input boost: before cutting the audio, we pre amplify the audio to avoid audio clipping.
Look ahead time/ attack time: time to wait
Determines how quickly starts
Release time: time to release back to normal.
Determines how quickly compression stops after reaching normal.

Multiband compressor effect

just like parametric Equalizer it’s also a good & advanced effect to process audio frequently.
Main: compressing at band or frequency specific.
3 bands
Mid: 2k
5bands (not a scientific values)
Low mid: 100-500
High mid 2k to 10k
(select bands by either manually entering frequencies or presets and edit later)
Vocal frequencies at 80hz to 10k
Voice Fundamentals falls beween 80-300hz
High pass filter in adobe audition
bypass: ignore the band
Input level meter:
gain: 4db (boost or cut +4db or -4db)
attack: 10ms

speech volume leveler effect

compression effect that optimizes dialogue, evening out levels and removing background noise.
Target volume level (db):
Low settings: amplify speech slightly without boosting the noise floor.
High settings: amplifys entire signal as it drops to close the noise floor
Boost low signals: shorter & low volume passages (for most speech content, skip this for smooth audio)
Minimizes background noise while amplifying and leveling speech content.
ex: -45db to -60db
higher: upper -85db greater background noise but lower amplitude & leveling.
lower: (down -30db) higher amplitude & leveling but background noise also boosts.
Advanced settings:
compressor : maintains strong signal by gain if processed signal falls .

Adobe audition ideal effects for live streaming & podcast

virtual audio cable or (voice meter banana ,fl studio)
sound card to audition >> virtual cable inout device>>obs.

High Pass Filter In Adobe Audition

adobe audition effects:
Noisegate if through vst plugins (optional).
enable high pass filter: to remove background noises
Use presets: like vocal enhancer or cusomize and save as preset.
Tip: vocal frequency 100hz – 10k even lower 400-4000hz
Denoise simple: just reducing amount of noise by 0-100%
still not reduced , windows>>sound settings>> input device level (dercrease) input device boost(reduce)
if mixer& interface: reduce gain.
Adaptive noise.

Adobe Audition High Pass Filter

automatically adjusts between ranges dynamicai, unlike static denoise.
Multiband compressor:
if you unfamiliar with multiband just use tube modelled compressor
we its reduces above threshold in a certain ratio.
More than -1db unbearable sound.

Adobe Audition High Pass Filter

speech leveler (option)
it has gain & compression & noise suppression techniques so use it after equalizer (to get best results)
Avoid over processing.(just focus on audible sound at beginning)

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