Audio Amplifier

Related terms:

Amplifier, Diodes, Transistors, Probability Mass Function, Random Variable

View all Topics

Learn more about Audio Amplifier

Amplifiers
Marilyn Wolf, in Embedded System Interfacing, 2019

4.9 Op Amps
Integrated audio amplifiers are designed for particular applications. The operational
amplifier or op amp, in contrast, is designed as a general-purpose component used
to build specific circuits. Op amps are used in all sorts of ways in circuit design. We
will cover more sophisticated uses of op amps in more detail in Chapter 5; here we
concentrate on their use as amplifiers.

The schematic symbol for an op amp is shown in Fig. 4.23. It has two inputs, +
and −. The one output value is formed by the difference of the two inputs. An op
amp requires a power source but we will typically omit their power connections in
our illustrations. The 741 is a classic op amp but other designs provide a range of
operating characteristics.

Fig. 4.23. Schematic symbol for an op amp.

An ideal op amp has three basic characteristics:

• Infinite input impedance.

• Infinite gain.

• Zero output impedance.

Real op amps cannot provide these superlative characteristics but reasonable circuits
can provide very good approximations of them over a wide range of frequencies.
A typical op amp circuit is built from two major blocks: a differential amplifier
compares the two input voltages; a voltage amplifier is used to provide voltage
gain for the differential amplifier's result. Op amps are mainstays for small-signal
amplification. While op amps generally do not provide the drive capability required
for large power amplifiers, they can be used as an input stage. Many power amplifiers
also use differential input stages similar to those used in op amps.

An op amp without feedback will produce output voltages at one extreme or the
other based on the polarity of V+ − V−. We generally use feedback circuits to control
the op amp.

Fig. 4.24 shows a linear amplifier built from an op amp [41]. This is an inverting
amplifier with voltage gain

Fig. 4.24. An inverting linear amplifier based on an op amp.

(4.13)

For example, R2 = 10 kΩ, R1 = 1 kΩ gives AV = 10. This formula is easy to derive. Since
the input terminals have infinite impedance, no current flows into them so:

(4.14)

We can rearrange the terms and obtain the gain formula of Eq. (4.13).

Fig. 4.25 shows a noninverting topology for an amplifier. The gain in this case is
Fig. 4.25. A noninverting linear amplifier based on an op amp.

(4.15)

In the noninverting case, R2 = 10 kΩ, R1 = 1 kΩ gives AV = 11.

Fig. 4.26 shows an op amp used to generate the difference between the two inputs.
When all four resistors have the same value, the output is VB − VA. We can generate
weighted differences by changing the relative values of the resistors.

Fig. 4.26. An op amp difference amplifier.

Real op amps have nonideal characteristics that will affect circuit design to some
degree [32]. The most important is its finite bandwidth. We evaluate bandwidth using
open loop gain. While an op amp has a very high (albeit finite) gain, that gain starts
to roll off quickly: the − 3 dB point of the μA741 op amp is about 5 Hz [60]; other
op amps may roll off at hundreds of hertz. The gain rolls off at 6 dB per decade.
The unity gain point is the frequency at which the op amp's gain is 1. The μA741, for
example, has a unity gain frequency of about 1 MHz. When we use the op amp in
a closed-loop circuit, we want to be sure that the closed loop gain at the maximum
required frequency is considerably larger than the op amp's open loop gain at that
frequency. A good rule of thumb is that the op amp should have an open loop
bandwidth of at least 10X the required closed-loop bandwidth [32].

Slew rate describes the large-signal behavior of the op amp; slew rate is also used
to evaluate other types of amplifiers. Slew rate is defined as the ratio of change in
output voltage per unit time:

(4.16)
Slew rate is limited by the amplifier's maximum output current. The μA741 has a slew
rate at unity gain of 0.5 V/μs.

The noise of an op amp will limit the dynamic range through which it can be
operated.

Common mode rejection ratio (CMRR) measures the response of the op amp to
the same voltage on both its input terminals. The magnitude of CMRR is equal to
the inverse of the ratio of the common mode output voltage to the common mode
input.

In any feedback circuit we need to be concerned about stability. Parasitics internal to

the op amp can create undesired feedback that results in instability. In some cases,
we may need to add compensation capacitors to ensure the stability of the op amp.

> Read full chapter

AC Circuits
Martin Plonus, in Electronics and Communications for Scientists and Engineers,
2001

EXAMPLE 2.14
It is desired to transfer maximum power from an audio amplifier that has a large
internal resistance to a speaker with a small internal resistance. Speakers typically
have small resistances (4, 8, 16 Ω) because the wire winding, which is attached to
the speaker cone and which moves the cone back and forth (thus creating acoustic
pressure waves), has to be light in weight to enable the speaker to respond well to
high frequencies. If the amplifier is characterized by an output impedance of 5000
Ω and the speaker by 5 Ω, find the turns ratio of an audio transformer for maximum
power transfer to the speaker. Figure 2.20a shows the mismatch if the speaker were
connected directly to the amplifier. Only a small fraction of the available power would
be developed in the speaker. We can show this by calculating the ratio of power
dissipated in the speaker to power dissipated in the amplifier: P5Ω/P5000Ω = I25/I25000
= 0.001 or 0.1%. Thus, an insignificant amount of power reaches the speaker.
FIGURE 2.20. (a) A speaker directly connected to an amplifier would usually be badly
mismatched. (b) For maximum power transfer, an iron-core (denoted by the vertical
bars) transformer is used to provide matched conditions.

We know from Section 1.6 that for maximum power transfer the load resistance
must be equal to the source resistance. This matching can be achieved by placing
an iron-core audio transformer between amplifier and speaker as shown in Fig.
2.20b. Using (2.50), we can calculate the turns ratio needed to produce a matched
condition. Thus

which gives that N1 = 33N2. Therefore, a transformer with 33 times as many turns on
the primary as on the secondary will give a matched condition in which the source
appears to be working into a load of 5000 Ω and the speaker appears to be driven
by a source of 5 Ω.

> Read full chapter

Loudspeaker systems
Leo Beranek, Tim Mellow, in Acoustics: Sound Fields, Transducers and Vibration
(Second Edition), 2019

Electromechanoacoustical circuit
The complete circuit for a loudspeaker with a transmission-line enclosure is obtained
by combining Figs. 6.4(b) and 7.36. To do this, the acoustical radiation element of
the circuit labeled “2MM1” in Fig. 6.4(b) is removed, and the circuit of Fig. 7.36
is substituted in its place. The resulting circuit with the transformer removed and
everything referred to the acoustical side is shown in Fig. 7.37.
Figure 7.37. Complete electromechanoacoustical circuit for a transmission-line
loudspeaker. The total force produced at the voice coil by the electric current is SD,
where SD is the area of the diaphragm. The volume velocity of the diaphragm is and
that of the transmission-line throat outlet is .

The quantities not listed in the previous paragraph are

is open-circuit voltage in V of the audio amplifier.

B is flux density in the air gap in T (1 T = 104 G).
l is length in m of voice-coil wire.
Rg is output electrical resistance in Ω of the audio amplifier.
RE is electrical resistance in Ω of the voice coil.
a is effective radius of the diaphragm in m.
is acoustic mass of the diaphragm and the voice coil in kg/m4.
is acoustic compliance of the diaphragm suspension in m5/N.
is acoustic resistance of the diaphragm suspension in N·s/m5.

If the outlet of the transmission line is closed off so that equals zero, then Fig. 7.37
essentially reduces to Fig. 7.6. At very low frequencies, the mass of air moving out
of the lower opening is nearly equal to that moving into the upper opening at all
instants. In other words, at very low frequencies, the volume velocities at the two
openings are nearly equal in magnitude and opposite in phase.

> Read full chapter

Cable-Shielding Experiments
Robert Lacoste, in Robert Lacoste's The Darker Side, 2010

Experimental Setup
I've covered the theory. Now I’ll focus on the process of testing inductive coupling.
I will also describe good shielding and wiring strategies to reduce noise coupling. It
took me just 30 minutes to build the small test bench illustrated in Figure 5.3.
Figure 5.3:. The reference configuration uses a single wire exposed to a strong
magnetic field with a return path through a ground plane situated 10 cm below the
wire. This configuration is our 0-dB reference by definition.

First, I needed to generate a strong magnetic field. I used an old Philips/Fluke

PM5134A lab signal generator and tuned it to frequencies from 20 to 20 kHz. To
amplify the signal, I hooked a 300-W audio amplifier and connected the speaker
output to a coil made with a 100-m roll of cabling wire. I added a 10-ohm series
power resistor to reduce the risk of blowing out the amplifier. I also used the setup
with a 20-MHz frequency—this time with a direct drive to the coil from the signal
generator because an audio amplifier doesn't have such a bandpass.

Then I used a 1.20-m × 50-cm metallic grid to build a reasonably good ground

plane and placed a wire 10 cm above the ground plane and 25 cm away from the
noise-generating coil. I connected a 47-ohm resistor between the wire end and the
ground plane, emulating a 50-ohm null signal source, and used a spectrum analyzer
to measure the level of noise received by the wire through the other end. Figures 5.4
and 5.5 show the actual installation.
Figure 5.4:. An overall view of my actual test bench. The spectrum analyzer used as
a receiver is an HP3585 model, on the left at the end of the BNC cable.

Figure 5.5:. A close-up view of the experiment. The frequency generator is at the
top and set to 1.9987 kHz. It is driving the 300-W power amplifier on the right
(the one with LEDs). The amplifier output then excites the big coil made simply
with a 100-m white wire roll. This coil generates a strong magnetic field, which is
received by a simple wire. This victim wire is positioned 10 cm above a ground plane
and connected to the ground, through a 47-ohm resistor on the right, and to the
spectrum analyzer. The boxes are empty.

Next, I switched on the power and measured the noise level at different noise
frequencies. The measured noise power levels ranged from −43 dBm at 200 Hz to
−33 dBm at 20 MHz (see Figure 5.3). Just remember that 0 dBm = 1 mW, −10 dBm
= 100 µW, −20 dBm = 10 µW, and so on. The formula is

So the received levels were ranging from 0.05 µW to 0.5 µW. We’ll use these values as
a reference to compare other wiring schemes. These powers are “0 dB.” Let's name
this reference “configuration A.”
> Read full chapter

Interfacing Digital Logic to the Real

World: A/D Conversion, D/A Conver-
sion, and Power Electronics
Ian Grout, in Digital Systems Design with FPGAs and CPLDs, 2008

8.4.1 Introduction
Devices such as FPGAs and CPLDs operate at low voltage levels (5 V and lower) and
at low current levels (µA to mA). These are considered low-power devices as they
are not designed to operate at high-voltage and/or high-current levels. However, in
many situations, low-power devices must control large electrical loads such as DC
motors, AC motors, stepper motors, and audio amplifiers. High-power electronics
[12–14] will need to interface the loads to the low-power electronics. The high-power
circuit components considered here are the:

• diode

• power transistor

• thyristor

• gate turn-off thyristor

• asymmetric thyristor

• triac

Each component type has a particular set of characteristics and use within an
electronic system. Low-power and high-power devices are interfaced either by direct
electrical connection or through an opto-isolator, as show in Figure 8.33.

Figure 8.33. Connecting the electronics

 The opto-isolator connects the electronics using an optical link rather than an
electrical link, so it electrically isolates the electronics but allows for signals to be
transferred by the optical link. This is particularly useful for situations where high
voltage levels in the high-power electronics must be electrically isolated from the
low-power electronics, and where high-power electronics and electrical load create
a substantial amount of electrical noise that could interfere with the operation of the
low-power electronics.

> Read full chapter

Volare
Lucio Di Jasio, in Programming 16-Bit PIC Microcontrollers in C (Second Edition),
2012

Producing Analog Waveforms

With help from the OC1 module we have just crossed the boundary between the
digital world, made of ones and zeros, and the analog world, where we have been
capable of generating a multitude of values between 0 and 3.3 V.

We can now play with the duty cycle, changing it from period to period to produce
waveforms of any sort and shape. Let’s start by modifying the project a little bit by
adding some code to the interrupt routine that so far has been left empty.

  OC1RS = (count < 20) 400 : 0;

  if ( ++Count >= 40)

    Count = 0;

You will need to declare count as a global integer and remember to initialize it to 0.

Save and rebuild the project to test the new code on the Explorer16 board.

Every 20 PWM period the filtered output will alternate between the value 3 V (100%)
and the value 0 V (0%), producing a square wave visible on the oscilloscope at the
frequency of 1 kHz (40 kHz/40).

A more interesting waveform could be generated by the following algorithm:

  OC1RS = Count*10;

  if ( ++Count >= 40)

    Count = 0;
 This will produce a triangular waveform (saw tooth) of approximately 3 V peak
amplitude, with a gradual ramp of the duty cycle from 0 to 100% in 40 steps (2.5%
each), followed by an abrupt fall back to 0 where it will repeat with a frequency of
1 kHz as well (Figure 15.8).

Figure 15.8. A 1 kHz triangular waveform

Neither of the two examples will qualify as a nice sound though if you try and feed
them to an audio amplifier, although they will both have a recognizable (fundamen-
tal) high-pitch tone, at about 1 kHz. Lots of harmonics will be present and will be
audible in the audio spectrum, giving the sound an unpleasant buzz.

To generate a single, clean tone what we need is a pure sinusoid. The interrupt
service routine below would serve the purpose, generating a perfect sinusoid at the
frequency of 400 Hz (in musical terms that would be close to an A).

void _ISRFAST _T3Interrupt(void)

  // compute the new sample for the next cycle

  OC1RS = 200+ (200* sin(Count *0.0628));

  if ( ++Count >= 40)

    Count = 0;

  // clear interrupt flag and exit

  _T3IF = 0;

} // T3 Interrupt

Unfortunately, as fast as the PIC24 and the math libraries of the MPLAB® C compiler
are, there is no chance for us to be able to use the sin() function, and to perform
the multiplications and additions required to obtain a new duty cycle value at the
required rate of 400 Hz. The Timer3 interrupt hits every 25 μs, too short a time for
such a complex floating-point calculation, so the interrupt service routine would
end up skipping interrupts and producing a sinusoidal output that is only half the
required frequency (one octave lower). Still, if you try and listen to it, feeding the
signal to an audio amplifier, you will be immediately able to appreciate the greatly
improved clarity of the sound.

For higher frequencies we will need to pretabulate the sinusoid values so we can
perform the fewest calculations possible (preferably working on integers only) at
run time. Here is an example that uses a table (stored in the FLASH program
memory of the PIC24) containing precomputed values. I obtained the table by using
a spreadsheet program where I used the following formula:

= offset + INT(amplitude * SIN(ROW * 6.28 / PERIOD))

for a period of 100 samples (400 Hz), offset and amplitude of 200, I obtained

= 200 + INT(200 * SIN(A1 * 6.28/100))

I filled the first column (A) of the spreadsheet with a counter and I copied the formula
over the first 100 rows of the second column (B), formatting the output for zero
decimal digits (Figure 15.9).

Figure 15.9. Spreadsheet to compute the 400 Hz sinusoid table

 Then; I cut and pasted the entire column into the source code, adding commas at
the end of each line to comply with the C syntax.

int Sample;

const int Table[100] = {

200, 212, 225, 237, 249, 261, 273, 285, 296, 307,

317, 327, 336, 345, 354, 361, 368, 375, 380, 385,

390, 393, 396, 398, 399, 399, 399, 398, 396, 393,

390, 386, 381, 375, 368, 361, 354, 345, 337, 327,

317, 307, 296, 285, 273, 262, 250, 237, 225, 212,

200, 187, 175, 162, 150, 138, 126, 115, 103, 93,

82, 72, 63, 54, 46, 38, 31, 24, 19, 14,

9, 6, 3, 1, 0, 0, 0, 2, 3, 6,

9, 13, 18, 24, 30, 37, 45, 53, 62, 72,

81, 92, 103, 114, 125, 137, 149, 161, 174, 186

};

void _ISRFAST _T3Interrupt(void)

  // load the new samples for the next cycle

  OC1RS = Table[Sample];

  if ( ++Sample >= 100)

    Sample= 0;

  // clear interrupt flag and exit

  _T3IF = 0;

} // T3 Interrupt

This time, you will be able to easily produce the 400 Hz output frequency desired
(Figure 15.10) and there will be plenty of time between the Timer3 interrupt calls to
perform other tasks as well.
Figure 15.10. A clean 400 Hz sinusoid

> Read full chapter

Introduction
C. Jutten, P. Comon, in Handbook of Blind Source Separation, 2010

1.1.3.1 A few pioneers

Although the GRETSI’85 and Snowbird’86 communications did draw some attention
from a few researchers into this problem, they actually raised great interest from L.
Kopp (Thomson-Sintra company) who engaged P. Comon for working on this prob-
lem in 1988. In this company, nobody wanted to work on this mysterious problem,
for which it was difficult to guarantee any outcome within less than a year. During
a workshop in Grenoble in September 1987, J.-F. Cardoso visited J. Hérault and C.
Jutten, who explained to him the principles of source separation and showed him a
real-time exhibition based on a source separation hardware demonstrator: a purely
analog device based on operational amplifiers, transistors and audio amplifier that
Jutten built in 1985, and which is able to separate, in real-time, two audio sources
in a mixture controlled by potentiometers [43]. Immediately and independently, J.-F.
Cardoso and P. Comon became enthusiastic about blind source separation, and the
rest of the story is known.

We cannot give here the complete list of their contributions, but we would like to fo-
cus on the earliest ones. P. Comon adapted the concept of contrast function, inspired
by Donoho’s contrast function used in blind deconvolution, and formulated the ICA
concept, first published at the HOS workshop in 1991 [22] and then in a famous
paper published in Signal Processing [23]. J.-F. Cardoso introduced many concepts: a
tensorial approach for conveniently representing and processing cumulants (togeth-
er with P. Comon) [15], performance analysis of ICA [17,13], joint diagonalization [17]
and the concept of equivariance [48,16] with the relative gradient (independently of
Amari and Cichocki’s natural gradient).

Finally, we believe that the success of the Signal Processing papers [44,26,70] has
been due to a surprising performance with respect to the algorithm’s simplicity.
Half a day was sufficient to write and test the algorithm. However, clearly, the
performance was dependent on the mixture’s hardness. J.-F. Cardoso, with Laheld,
were looking for algorithms enjoying invariant performances.

> Read full chapter

Digital Systems
Martin Plonus, in Electronics and Communications for Scientists and Engineers,
2001

9.3.3 Speech Signal

Next we consider a continuous signal such as speech. Figure 9.2a shows a typical
speech signal plotted on a log scale as a function of time. The vertical log scale
reflects the fact that the human ear perceives sound levels logarithmically. Ordinary
speech has a dynamic range of approximately 30 dB,2 which means that the ratio of
the loudest sound to the softest sound is 1000 to 1, as can be seen from Fig. 9.2a.
Combining this with the fact that human hearing is such that it takes a doubling of
power (3 dB) to give a noticeably louder sound, we can divide (quantize) the dynamic
range into 3 dB segments, which gives us 10 intervals for the dynamic range, as
shown in Fig. 9.2a. The 10 distinguishable states in the speech signal imply that the
quantity of information at any moment of time t is

FIGURE 9.2. (a) A typical speech signal showing a 30 dB dynamic range. Along the
vertical scale the signal is divided into 10 information states (bit depth), (b) Screen
of a picture tube showing 525 lines (called raster lines) which the electron beam,
sweeping from side to side, displays. All lines are unmodulated, except for one as
indicated.

(9.4)

We need to clarify resolution along the vertical scale, which is the resolution of
the analog sound's amplitude. Using 10 quantizing steps is sufficient to give
recognizable digital speech. However, a much higher resolution is needed to create
high-fidelity digital sound. Commercial audio systems use 8 and 16 bits. For exam-
ple, the resolution obtainable with 16-bit sound is 216 = 65,536 steps or levels. As
bit depth (how many steps the amplitude can be divided into) affects sound clarity,
greater bit depth allows more accurate mapping of the analog sound's amplitude.

> Read full chapter

Electrostatic loudspeakers
Leo Beranek, Tim Mellow, in Acoustics: Sound Fields, Transducers and Vibration
(Second Edition), 2019

Part XXXXIV: Lumped-element model of an electrostatic loud-

speaker

15.10 Electro-mechano-acoustical circuit

In Section 14.10 we developed an analytical (distributed-element) model of a circular
electrostatic loudspeaker. Here we will develop a simpler lumped-element model
which is valid when there is sufficient resistance (usually in the form of a dust
screen) to suppress the membrane modes. Using an analogous circuit, we will then
develop useful design formulas.

The analogous circuit of the electrostatic loudspeaker shown in Fig. 15.1 is given by

Fig. 15.16. Although this is a general circuit, we shall assume for this analysis that
the loudspeaker is circular with radius a and has no enclosure whatsoever.

Figure 15.16. Electro-mechano-acoustical analogous circuit of the electrostatic

loudspeaker shown in Fig. 15.1.
The symbols have the following meanings:

is the voltage of the generator (audio amplifier) in volts (V).

is the total input current in amperes (A).
is the static part of the input current in amperes (A).
is the motional part of the input current in amperes (A).
CE is the static capacitance between the electrodes in Farads (F).
−CE represents the negative capacitance due to electrostatic attraction in
Farads (F).
EP is the polarization supply voltage in volts (V).
d is the separation distance between the membrane and each electrode in
meters (m).
is the mechanical force driving the membrane in Newtons (N).
is the average velocity of the membrane in m/s.
is the total mechanical impedance in N·s/m.
CMD is the mechanical compliance of the membrane in m/N due to its
tension.
MMT is the total moving mass of the membrane and stator perforations in kg.
RMS is the mechanical resistance due to viscous flow losses through the stator
electrode perforations and dust screen.
SD = πa2 is the surface area of the membrane in m2, where a is the radius in
m.
is the total volume velocity in m3/s.
is the pressure in N/m2 driving the radiation load 2ZAR on both sides of the
membrane.
MAR is the acoustic radiation mass on one side of the membrane in kg/m4.
CAR is the acoustic radiation compliance on one side of the membrane in
m5/N.
RAR is the acoustic radiation resistance on one side of the membrane in
N⋅s/m5.

For simplicity, we assume that the output impedance of the amplifier and resistance
of the cables are negligible, and we also ignore any stray capacitance in the cables.
There are two transformers: the first acts as an interface between the electrical
domain and the mechanical one converting voltage to force and current to velocity
, while the second acts as an interface between the mechanical and acoustical
domains, converting force to pressure and velocity to volume velocity .

Notice how the input current divides into two: one is the static current while the
other is the motional current . The static current still flows when the membrane is
blocked (or the polarization voltage EP is turned off ), but the motional current is
dependent on the membrane velocity . Unfortunately, in most practical electrostatic
loudspeakers , so that the electrical input impedance is defined almost entirely by
CE, although it is possible to measure the motional current by “balancing out”
the static current with a capacitor [7]. The stator resistance RMS is technically an
acoustic flow resistance, but it is included on the mechanical side of Fig. 15.16 for
convenience.

> Read full chapter

Cell phone acoustics

Leo Beranek, Tim Mellow, in Acoustics: Sound Fields, Transducers and Vibration
(Second Edition), 2019

8.2 Circuit diagram for a cell phone loudspeaker/receiver

The circuit diagram for a cell phone loudspeaker system is given in Fig. 8.2. The
elements are derived from the circuit of a loudspeaker in a closed box baffle as given
in Fig. 7.6. The symbols are as follows:

Figure 8.2. Analogous circuit of a handsfree loudspeaker or receiver in a cell phone.

All circuit elements are referred to the acoustical side. In the case of a receiver, ZAE
is the impedance of the ear including any leakage because it is unlikely to be sealed.
In the case of a handsfree loudspeaker near a flat surface, ZAE can be considered to
be the radiation impedance of a piston in an infinite baffle.

is open-circuit voltage of the generator (audio amplifier) in volts (V).

Rg is generator resistance in electrical ohms (Ω).
RE is resistance of voice coil in electrical ohms (Ω).
B is steady air-gap magnetic field or flux density in Tesla (T).
l is length of wire on the voice coil winding in m.
is electric current through the voice coil winding in amperes (A).
a is radius of diaphragm in m.
SD = πa2 is area of diaphragm in m2.
·SD is force produced by the current in the coil in Pa·m2.
is volume velocity produced by the diaphragm in m3/s.
MAD is acoustic mass of the diaphragm and the voice coil in kg/m4.
CAS is total acoustic compliance of the suspensions in m5/N.
RAS is acoustic resistance of the suspensions in N·s/m5.
CAB is total acoustic compliance of the back enclosure in m5/N.
CAF is acoustic compliance of the front cavity in m5/N.
RAB is acoustic resistance of the leak path through the enclosure (needed to
relieve changes in atmospheric pressure) in N·s/m5.
MAH is acoustic mass of the sound hole(s) in kg/m4.
RAH is acoustic resistance of the dust screen in N·s/m5.
ZAE is radiation impedance (including the effect of proximity to the ear).

The radiation impedance, ZAE, is difficult to specify because the impedance for
the receiver opening is highly dependent on how the user uses the handset. Because
both of the openings are small, the radiation impedance when not too near the ear
will approximate that for a small diaphragm in the end of a tube. Possible means for
assuring a known radiation impedance in the receiver opening are to deliberately
build a degree of controlled leakage that is at least as great as the uncontrolled
leakage that would occur in normal usage. The controlled leakage path may lead
to the outside space as shown in Fig. 8.3(a) or to the internal space as shown in
Fig. 8.3(b). The more probable solution is that shown in Fig. 8.3(a), and for it a series
acoustic resistance and acoustic mass must be connected between the circles “1”
and “3” in Fig. 8.2. For the solution of Fig. 8.3(b), the mass and resistance should be
connected between the circles “1” and “2.” Obviously the controlled leakage addition
will reduce the output strength. The solution of Fig. 8.3(b) in particular will cause
loss of low frequencies because of the acoustic short circuit between the front and
back of the diaphragm, unless the space inside the handset is very large (>60 cm3).
 Figure 8.3. Cross section of leak-tolerant receiver in cell phone with (a) external leak
and (b) internal leak [1].

Acoustic low-pass filter (helmholtz resonator)

In cell phones, for both the handsfree loudspeaker and the receiver, the compli-
ance of the front cavity CAF and the mass of the sound opening MAH form a
Helmholtz resonator. This is a second-order low-pass filter. The angular resonance
frequency is

(8.1)

When listening to music, the resonance frequency is normally set at the upper limit
of the required frequency range, and the Q of the resonance is controlled by the
resistance RAH of the dust screen. To calculate the dimensions of the resonator, we
can either choose the radius a of the opening and calculate the length l according
to l = n c2a2/(4π f02V) −  a or choose the length and calculate the radius according to

(8.2)

These are general formulas for a Helmholtz resonator, such as a bottle, which are
derived from those given in Sections 4.2 and 4.3 for an acoustic mass MAH and
acoustic compliance CAF, respectively.

The quantities are defined as follows:

l is length of opening in m.
a is radius of opening in m.
is end correction factor.
f0 is resonance frequency in Hz.
V is volume of cavity in m3.
c is speed of sound = 348.8 m/s at P0 = 105 Pa and T = 22°C.

The end correction factor for the opening is given by

(8.3)

> Read full chapter

ScienceDirect is Elsevier’s leading information solution for researchers.

Copyright © 2018 Elsevier B.V. or its licensors or contributors. ScienceDirect ® is a registered trademark of Elsevier B.V. Terms and conditions apply.