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Pure Audio
The next set of circuits are pure audio, and include tone controls, equalisers and filters, as well as balanced
line driver and receivers. There is also a simple power amplifier suitable for use with headphones or low
powered speakers.
When we speak of audio, the commonly accepted range is from 20Hz to 20kHz, and at some point someone
decided that the centre frequency is 1kHz. I have never known why, since if you look at the octaves covered
by the audio range we have a sequence like this ...
20
40
80
160
320
640
1280
2560
5120
10240
20480
This is a total of 10 octaves, so the centre frequency must actually be 640Hz, and indeed this is much closer
to reality than 1kHz. If we were to divide the musical spectrum based on energy content, the centre
frequency is about an octave lower, or 320Hz (approximately - it depends to a large degree on the style of
music).
I am going to break with tradition and suggest that the 640Hz frequency is the midpoint, and 1kHz shall be
ignored for other than comparative purposes. In reality it makes little difference either way.
So that the limitations of opamps in an audio circuit can be fully understood, we shall also look at the
compensation curves typical of standard and high quality opamps, and examine the alternatives for very
wide bandwidth.
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to say the least. In fact, the 3dB frequency for the treble control changes with the pot setting. For example,
at 80% of rotation, treble boost is +3dB at just over 3.5kHz, with a maximum boost of 6.3dB at 20kHz. This
may seem to be a bad thing, having the frequency shifting about as the pot is varied, but in practice it works
very well. To obtain the results youwant, I suggest that experimentation is in order !
A final word about the tone control circuit. Note that it is a 'virtual earth' circuit, so the feedback at all times
will maintain the -ve input at zero volts. The bass and treble content of the input waveform will force the
amplifier to provide just the amount of boost or cut at any frequency to maintain the 0V condition on -in. You
will find this useful as you work towards an understanding of the complete circuit.
Order
Sections
Nominal Rolloff
First
6 dB / Octave
Second
12 dB / Octave
Third
18 dB / Octave
Fourth
24 dB / Octave
"n"
Where n > 4
6 dB / Octave / section
Response
Passes ...
Blocks ...
Low Pass
Low frequencies
High frequencies
High Pass
High frequencies
Low frequencies
Band Pass
Selected frequency
Band Stop
Selected frequency
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Essentially, there are also a few basic filter alignments, which are as follows (not all are included, and many
are applicable to filters of more than one section, or more complex arrangements):
Filter Type
Q (Typ)
Characteristics
Bessel
0.57
Butterworth
0.707
Flattest amplitude
Chebychev
0.8 - 1.3
Cauer / Elliptical
0.7 - 1.3
Circuit Type
Gain
Characteristics
Sallen-Key
Unity
Simple design
Depends on Q
Simpler design
Multiple feedback
Depends on Q
Relatively complex
State Variable
Depends on Q
Relatively complex
Biquadratic
Depends on Q
Relatively complex
Passive
< unity
Simple or complex
Cascading: Simple filters can be made with one (or none) active devices. As filter
orders become greater, it is no longer feasible, so filters are joined together in
series to obtain the desired response. This complicates the design (often
dramatically).
Cutoff Frequency: Also shown as Fo, this is the -3dB frequency of the filter,
relative to the highest peak (if any exist) in the passband.
Decade: A 10:1 (or 1:10) ratio of frequency. For example 100Hz to 1000Hz is one
decade. One decade is approximately 3.16 octaves.
Decibel: (dB) - the most common way to describe amplitude in audio. The dB scale
is logarithmic, and describes the amplitude as we hear it. A 3dB drop in gain
equates to half the power in an amplifier.
Quality Factor: Commonly known as Q, this is the inverse of the filter's damping.
For example, a Butterworth filter has a Q of 0.707, which equals a damping of
1.414. Higher Q gives more ripple for low and high pass filters, or makes a
bandpass or band stop filter more selective.
Octave: A 2:1 (or 1:2) ratio of frequency. For example, 440Hz to 880Hz is one
octave (Musical 'A' note).
Order: The order of any filter determines its rolloff frequency response. First order
filters roll of at 6dB per octave, and as the order increases, so too does the rolloff
rate. An additional 6dB / octave is gained for each additional order, starting from
first. A 3rd order filter will therefore roll off at 18dB /octave, for example.
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Note that the input of this filter must have a low impedance return to earth at the input, or the opamp will not
have any bias current or voltage, and will not work. This rule applies to low pass filters in all cases without
exception.
For the filter in Figure 13, the frequency is determined by ...
Fo = 1 / ( 2 * 10k * 100nF ) = 159 Hz
Rc = 1 / ( 2 * 159 * 100nF ) = 10k ohms
By reversing the positions of R and C, we obtain a high pass filter. The formula remains the same, and the
two filters will have a complementary response centred at the 159Hz frequency. There is no need for an
earth return at the input for a high pass section, as this is provided by the resistor.
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In reality, it may be found that the value of capacitance becomes so great that the capacitor will be more
expensive than a section of an opamp. Alternatively, the resistance may become so low that no opamp can
drive the load. In all cases, it is essential to ensure that impedances are within the acceptable range for the
opamps used. Generally, impedances of less than 1k at any frequency are not recommended. For noise
considerations, very high resistance values are also not recommended, and I suggest that 100k is a
reasonable compromise. There will be occasions where this is not practicable and higher values may be the
only sensible solution, but these should be few and far between.
If the high pass section of Figure 15 is verified by calculation, it is discovered that the cutoff frequency (Fo)
should be 7234Hz, and not 6400Hz as shown. The difference is caused because the second filter loads the
first, shifting the frequency. This is a very good reason to isolate the sections using an opamp.
Component Selection
All following filter sections will use resistance within the range of 1k to 100k. This gives two decades of
freedom, and this is more than enough to allow capacitors of sensible sizes to be used. Very low capacitance
values are to be avoided, since the capacitance of wiring (PCB tracks etc.) will modify the filter
characteristics to a possibly unacceptable degree. I generally try to keep capacitance above 1nF wherever
possible, and this may save you some grief as we progress. Likewise, any capacitance above 1uF becomes
large and relatively expensive, but within this range we also have two decades of freedom, and there is
virtually no audio filter that cannot be designed within these constraints.
Generally, it is sensible to select the capacitor value first, as these have less available values within a
decade than resistors. Capacitors have 12 values per decade (the E12 series), while resistors have up to 96
values per decade (E96 series). The latter are not usually easy to get, but the E24 series is now very
common, and has (surprise!) 24 values per decade. The E12 and E24 series are shown in Table 5.
E12
1.0
E24
1.0
1.2
1.1
1.2
1.5
1.3
1.5
1.8
1.6
1.8
2.2
2.0
2.2
2.7
2.4
2.7
3.3
3.0
3.3
3.9
3.6
3.9
4.7
4.3
4.7
5.6
5.1
5.6
6.8
6.2
6.8
8.2
7.5
8.2
9.1
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gain precisely will work out to be impossible to obtain. For the values I am using, a 14.414k resistor would be
needed, which as you can see from Table 5 is not a standard value.
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Crossover Networks
When combined, high pass and low pass filters are commonly used to create electronic crossover networks.
There are many different ways to do this, but the most common is still the second order Butterworth filter.
Although these are essentially at least as good as the passive speaker level counterpart (but not all that
much better), the results are often vastly superior. The reasons for this are discussed at great length in my
article Bi-Amplification - Not Quite Magic (but Close) and I shall not repeat them here.
In a nutshell, using an electronic crossover eliminates many of the problems that beset loudspeaker
designers when they have to design and build the crossover network. The passive crossover is influenced by
any aberration in the driver's impedance, especially at or near the crossover frequency. Since the electronic
crossover supplies the signal to a separate amplifier for each frequency band, there is no interaction and
each amp only needs to be concerned with a much smaller bandwidth.
All crossover networks are a combination of high pass and low pass filters, although this is not always
achieved in the same way. Some crossovers use an opamp as a subtracting amplifier, so rather than using a
separate filter, the bass (for example) is subtracted from the main signal to provide the midrange and high
frequency. Alternatively, the mid+high output from the filter is subtracted from the main signal to separate the
bass, and this configuration is shown below. These crossovers are phase coherent (both outputs are always
in phase at any frequency), but are asymmetrical. A typical design is shown in Figure 19, and the response
clearly shows that the high pass rolloff is 12dB/octave, but the low pass is only 6dB/octave.
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The band pass filter (as a single filter) is not normally very useful in audio reproduction. The frequency range
passed is too narrow to be useful for anything other than equalisation circuits, or for test and analysis
equipment. Having said this, there are many uses for bandpass filters that are very common in
the production of music - the synthesiser, guitar wah-wah pedals, Vocoder, etc.
The range is far too great to cover in any real detail, but we can at least look at the fundamental principles,
as these are common to all of the applications. The two most common parameters quoted for bandpass
filters are frequency (fo) and Q (quality factor). The latter may be inverted and referred to as damping.
A simple bandpass filter consisting of two reactive elements has an ultimate rolloff of 6dB/octave. When one
looks at resonance, the slope appears to be much greater than this, which is fair and reasonable, since it is.
Eventually, the high slope due to resonance effects cannot be maintained, and the final slope is at
6dB/octave, as shown in Figure 20. But wait! This is an article about opamp designs, and there is no opamp
there, just a stupid inductor and capacitor. True, but we need to be able to understand the concept of
resonance before delving into the opamp version of the circuit.
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With the component values as shown in Figure 21, the voltage at the output will be about 0.5 mV at
resonance, but there will be 1V across both the inductor and capacitor. At resonance, the reactance of the
capacitor and inductor are equal and opposite, so the circuit will appear to be almost a short circuit - input
current is limited only by the wiring resistances and source impedance.
The maximum possible Q of an LC filter is dictated by the series resistance of the inductor. The Q may be
reduced by adding resistance, but cannot be increased. The circuit Q is determined by the reactance of the
cap or coil at resonance and the series resistance (for parallel resonance) or shunt resistance for series
resonant circuits.
References
I have used various references while compiling this article, with most coming from my own accumulated
knowledge. Some of this accumulated knowledge is directly due to the following publications:
National Semiconductor Linear Applications (I and II), published by National Semiconductor
National Semiconductor Audio Handbook, published by National Semiconductor
IC Op-Amp Cookbook - Walter G Jung (1974), published by Howard W Sams & Co., Inc. ISBN 0-672-20969-
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1
Active Filter Cookbook - Don Lancaster (1979), published by Howard W Sams & Co., Inc. ISBN 0-67221168-8
Data sheets from National Semiconductor, Texas Instruments, Burr-Brown, Analog Devices, Philips and
many others.
Recommended Reading
Opamps For Everyone - by Ron Mancini, Editor in Chief, Texas Instruments, Sep 2001