Mini Audio Signal Generator

A small audio test
generator is very useful for quickly tracing a signal through an audio
unit. Its main purpose is speed rather than refinement. A single
sine-wave signal of about 1 kHz is normally all that is needed:
distortion is not terribly important. It is, however, important that the
unit does not draw too high a current. The generator described meets
these modest requirements. It uses standard components, produces a
signal of 899 Hz at an output level of 1 V r.m.s. and draws a current of
only 20 µA. In theory, the low current drain would give a 9 V battery a
life of 25,000 hours. The circuit is a traditional Wien bridge
oscillator based on a Type TLC271 op amp. The frequency determining
bridge is formed by C1, C2 and R1–R4. The two inputs of the op amp are
held at half the supply voltage by dividers R3-R4 and R5-R6
respectively.

Circuit diagram:

Mini Audio Signal Generator Circuit

Mini Audio Signal Generator Circuit Diagram

Resistors R5 and R6 also form part of the feedback loop. The
amplification is set to about ´3 with P1. Diodes D1 and D2 are peak
limiters. Since the limiting is based on the non-linearity of the
diodes, there is a certain amount of distortion. At the nominal output
voltage of 1 V r.m.s., the distortion is about 10%. This is, however, of
no consequence in fast tests. Nevertheless, if 10% is considered too
high, it may be improved appreciably by linking pin 8 of IC1 to ground.
This increases the current drain of the circuit to 640 µA, but the
distortion is down to 0.7%, provided the circuit is adjusted properly.
If a distortion meter or similar is not available, simply adjust the
output to 1 V r.m.s. Since the distortion of the unit is not measured in
hundredths of a per cent, C1 and C2 may be ceramic types without much
detriment.

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Logic Probe With Sound

This logic probe can be selected to operate on TTL or CMOS
logic levels, depending on switch S1. A string of resistors associated
with switch S1 sets the threshold levels for a window comparator
comprising IC1a and IC1b. Depending on whether the level applied to the
probe is high or low, the window comparator turns on LED1 (high) or LED2
(low). The 1.2M and 680k resistors set the probe signal to a midrange
value when the probe is open-circuit, thereby preventing either LED from being lit.

Circuit diagram:

Logic Probe With Sound

Logic Probe With Sound

If a pulse signal is present, the output of IC1a will toggle the clock
input of flipflop IC2a. This drives LED3 which either lights for each
pulse or continuously, depending on the setting of switch S2. Finally,
the outputs of IC1a & IC1b are connected by diodes D5 & D6 to
the base of transistor Q1 which is connected to the Reset input of
flipflop IC2b. This has a piezo sounder (not buzzer) connected between
its Q and Q-bar outputs so that it produces a sound which echoes the
input pulse signal.

Author: Tom Hughes
Copyright: Silicon Chip Electronics

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Test Beeper For Your Stereo

The test beeper
generates a sinusoidal signal with a frequency of 1,000 Hz, a common
test frequency for audio amplifiers. It consists of a classical
Wien-Bridge oscillator (also known as a Wien-Robinson oscillator). The
network that determines the frequency consists here of a series
connection of a resistor and capacitor (R1/C1) and a parallel connection
(R2/C2), where the values of the resistors and capacitors are equal to
each other. This network behaves, at the oscillator frequency (1 kHz in
this case), as two pure resistors. The opamp (IC1) ensures that the
attenuation of the network (3 times) is compensated for.

In principle a gain of 3 times should have been sufficient to
sustain the oscillation, but that is in theory. Because of tolerances in
the values, the amplification needs to be (automatically) adjusted.
Instead of an intelligent amplitude controller we chose for a somewhat
simpler solution. With P1, R3 and R4 you can adjust the gain to the
point that oscillation takes place. The range of P1 (±10%) is large
enough the cover the tolerance range. To sustain the oscillation, a gain
of slightly more than 3 times is required, which would, however, cause
the amplifier to clip (the ‘round-trip’ signal becomes increasingly
larger, after all).

Circuit

Circuit diagram

To prevent this from happening, a resistor in series with two
anti-parallel diodes (D1 and D2) are connected in parallel with the
feedback (P1 and R3). If the voltage increases to the point that the
threshold voltage of the diodes is exceeded, then these will slowly
start to conduct. The consequence of this is that the total resistance
of the feedback is reduced and with that also the amplitude of the
signal. So D1 and D2 provide a stabilizing function. The distortion of
this simple oscillator, after adjustment of P1 and an output voltage of
100 mV (P2 to maximum) is around 0,1%. You can adjust the amplitude of
the output signal with P2 as required for the application. The circuit
is powered from a 9-V battery. Because of the low current consumption of
only 2 mA the circuit will provide many hours of service.

Author: Ton Giesberts – Copyright: Elektor Electronics 2007

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SDR Soundcard Tester

The key to using a
soundcard successfully in digital signal processing or digital radio
applications lies principally in the characteristics of the soundcard
itself. This applies in particular to SDR
(software defi ned radio) programs that turn your PC into a top-class
AM/SSB/CW receiver, assuming your soundcard cooperates. If you want to
experiment with SDR and avoid a lot of
frustration, it is worth checking first whether the PC soundcard you plan
to use is suitable. There are three essential elements to success:

  • the soundcard must have a stereo line-level input;
  • the card must be equipped with an input anti-aliasing filter; and
  • the sample rate must be at least 48 kHz and the card must be able to cope with signals up to 24 kHz.

Many laptops have only a mono microphone input, sometimes also
rather limited in bandwidth. In this case it may be possible to use an
external USB soundcard. Most desktop PCs these
days have an internal integrated soundcard, although some of these do
not feature an anti-aliasing fi lter. Attempts to disable the integrated
soundcard and replace it with a better one often meet with failure;
again, an external USB soundcard is a possible solution.

SDR Soundcard Tester Circuit

SDR Soundcard Tester Circuit Diagram

To avoid guesswork, the best way to proceed is to test the soundcard
using this very small circuit. This will help to diagnose any problems
and will help determine whether the card is suitable for use with an SDR
program. Figure 1 shows a simple square-wave generator built around an
NE555 timer IC. At the output is a 15 kHz signal rich in higher
harmonics. Using this we can determine whether or not the soundcard can
process the harmonics at 30 kHz, 45 kHz and so on. An anti-aliasing
filter at the soundcard input should attenuate all signals above 24 kHz.
The frequency of the test generator is, within limits, dependent on its
supply voltage.

Using an adjustable power supply, a frequency range from 10 kHz to
20 kHz can therefore be covered. There are two RC networks at the output
of the test circuit, a high-pass filter and a low-pass filter, acting
as simple phase shifters. At the basic frequency of 15 kHz these provide
a total phase difference of 90 degrees, corresponding exactly to the
typical situation at the output of an SDR receiver circuit using an I-Q mixer: signals at the same frequency but differing in phase. To test the soundcard we need an SDR program running on the PC as well as the circuit of Figure 1. Suitable software includes SDradio (available for download from http://digilander.libero.it/i2phd/sdradio/).

When things are running correctly, the screen should display just
two signals: the wanted signal at 15 kHz and a weaker image at –15 kHz
(Figure 2). Suppression of the image may not be particularly good as the
test circuit does not have very high phase and amplitude accuracy. If,
however, the signals have the same level, there is a problem in the
processing of the two channels: it is probable that the soundcard only
has a monophonic input. If there is no anti-aliasing filter at the input
of the soundcard the spectrum will show a large number of extra lines
(Figure 3): it is easy to work out which harmonic corresponds to which
alias frequency.

The results obtained using an I-Q receiver were grim: frequencies
all the way out to 100 kHz were wrapped into the audible range,
resulting in bubbling, hissing and whistling. In theory it would be
possible to add an anti-aliasing filter to the output of the receiver to
allow use with soundcards that are not equipped with such a filter. In
practice, however, it is not easy to achieve the required sharp cutoff
and symmetry between the two channels. A typical soundcard has a low
pass filter set at 24 kHz which by 27 kHz is already attenuating the
signal by some 60 dB. This is only practical using digital fi lters; an
adjustable analogue circuit to achieve this performance would be so
complex that the simplicity benefits of SDR receiver technology would entirely evaporate.

Author: Burkhard Kainka – Copyright: Elektor Electronics 2007

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DCF77 Preamplifier

A popular project among
microcontroller aficionados is to build a radio-controlled clock. Tiny
receiver boards are available, with a pre-adjusted ferrite antenna, that
receive and demodulate the DCF77 time signal broadcast from Mainf
lingen in Germany. DCF77 has a range of about 1,000 miles. All the
microcontroller need do is decode the signal and output the results on a
display. The reception quality achieved by these ready-made boards
tends to be proportional to their price.

In areas of marginal reception a higher quality receiver is needed,
and a small selective preamplifier stage will usually improve the
situation further. The original ferrite antenna is desoldered from the
receiver module and connected to the input of the preamplifier. This
input consists of a source follower (T1) which has very little damping
effect on the resonant circuit. A bipolar transistor (T2) provides a
gain of around 5 dB. The output signal is coupled to the antenna input
of the DCF77 module via a transformer.

DCF77 Preamplifier Circuit

DCF77 Preamplifier Circuit Diagram

The secondary of the transformer, in conjunction with capacitors C4
and C5, forms a resonant circuit which must be adjusted so that it is
centered on the carrier frequency. An oscilloscope is needed for this
adjustment, and a signal generator, set to generate a 77.5 kHz sine
wave, is also very useful. This signal is fed, at an amplitude of a few
milli-volts, into the antenna input. With the oscilloscope connected
across C4 and C5 to monitor the signal on the output resonant circuit,
trimmer C5 is adjusted until maximum amplitude is observed.

It is essential that the transformer used is suitable for
constructing a resonant circuit at the carrier frequency. Our proto-type
used a FT50-77 core from Amidon on which we made two 57-turn windings.
It is also possible to trim the resonant frequency of the circuit by
using a transformer whose core can be adjusted in and out. In this case,
of course, the trimmer capacitor can be dispensed with.

Rainer Reusch
Elektor Electronics 2008

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Crossover For Subwoofer

The crossover network is
intended for use when an existing audio installation is to be extended
by the addition of a subwoofer. Often, this additional loudspeaker is
one that has been lying around for some time. If its frequency response
extends down far enough, all is well and good, but a filter is then
needed to cut off any frequencies above, say, 150 Hz. Often, a subwoofer
network is an active filter, but here this would necessitate an
additional power supply. The present network is a passive one, designed
so that the speaker signal of the existing system can be used as the
input signal.

Crossover Circuit For Subwoofer

Crossover Circuit Diagram For Subwoofer

Since the bass information is present in both (stereo) loudspeakers,
the signal for the sub woofer can simply be tapped from one of them.
The network is a 1st order low-pass filter with variable input (P1) and
presettable cut-off frequency (P2). The signal from the loudspeaker is
applied to terminal ‘LSP’. Voltage divider R1-R2-P1 is designed for use
with the output signal of an average output amplifier of around d 50 W.
The crossover frequency of the network may be varied between 50 Hz and
160 Hz with P2. The values of R3, P2, and C1, are calculated on the
assumption that the subwoofer amplifier to be connected to K1 has a
standard input resistance of 47 kΩ.

If this figure is lower, the value of C1 will need to be increased
slightly. It is advisable to open the volume of the subwoofer amplifier
fully and adjust the sound level with P1. This ensures that the input of
the subwoofer amplifier cannot be overloaded or damaged. Make sure that
the ground of the loudspeaker signal line is linked to the ground of
the subwoofer amplifier. If phase reversal is required, this is best
done by reversing the wires to the subwoofer. If notwithstanding the
above additional protection is desired at the input of the subwoofer
amplifier, this is best effected by ‘overload protection ’ elsewhere in
this site.

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Passive RIAA Preamplifier

There are two types of
preamplifiers for magnetic phono cartridges. An example of the most
common type is the one described in the March 2002 issue of SILICON CHIP. It has the RIAA
equalisation network in the feedback loop. The second type was
previously used in valve circuits which typically had no feedback loop
and used passive RC networks to provide the phono equalisation. This
experimental preamp was put together using inexpensive FETs to compare the performance of these two types of preamp. The first stage, consisting of Q1 and Q2, is a simple FET audio amplifier, where the FETs are connected in parallel to reduce noise. This is followed by a passive RIAA network consisting of 240kO and 15kO resistors and the associated 0.1OF .022OF and .0047OF capacitors.

Circuit diagram:

Passive RIAA Preamplifier Circuit

Passive RIAA Preamplifier Circuit Diagram

Some of the gain loss in the passive network is then made up by FET
Q3. It also has a 51kO drain resistor and is buffered by bipolar
transistor Q4 which is connected as an emitter-follower stage. All
resistors are 1% tolerance metal film type while the capacitors for
equalisation are MKT polyester types. Ideally, the Idss of all FETs
should be matched for both channels. Resistors R3 and R8 should be
adjusted so that the drain voltage in each stage is between 13V and 14V,
to give symmetrical signal clipping. The power supply can be three 9V
batteries connected in series. Current consumption is only 3mA for the
stereo circuit.

Author: Sam Yoshioka – Copyright: Silicon Chip Electronics

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Audio Level Adapter

The problem that this
circuit is designed to solve appeared when the author was installing a
new radio in his Audi A3. The new radio had four outputs for
loudspeakers and a line-level output for a subwoofer. However, the A3 as
delivered from the factory already has an amplifier for the rear
loudspeakers, as well as the pre-installed subwoofer, in the boot space.
The original Audi radio therefore has only line-level outputs for the
rear loudspeakers. So, to replace the original radio without making
other changes to the installed amplification system, he needed to
convert the outputs of the new radio corresponding to the rear
loudspeakers into line level outputs.

Most of the commercially-available adapters to do this job contain
small transformers for galvanic isolation. These introduce phase shifts
and create a certain amount of distortion, which the author was keen to
minimize. The result is this simple adapter circuit that does not employ
a transformer. The outputs of most radios available today have a
differential (bridge-type) push-pull output stage. There is thus no
ground output, just two outputs per channel with a 180 ° phase
difference between them. If the outputs are each connected to a common
point via a 100 Ω resistor, that point becomes a virtual ground.

Circuit

Circuit diagram

The ground is relatively stable as (in the stereo case) it has an
impedance of 25 Ω. Each output driver is seeing a 200 Ω load: if the
amplifier is rated for 50 W output into a 4 Ω load this means that each
resistor will dissipate less than 0.5 W. Hence 1 W rated resistors will
be more than adequate, especially in view of the fact that typical music
has a crest factor of at least five. Even a small DC offset from the
virtual ground is not a problem, as most modern amplifiers feature
differential inputs or at least allow the ground connection of an input
to float. To reduce the signals to line level, each has to be connected
to a potential divider: a multi-turn preset potentiometer is ideal.

The author used a linear 10 kΩ trimmer to reduce the output voltage
level from up to about 12 Vpp to around 2 V to 3 V. This latter level is
suitable for the input to a power amplifier. An appropriate trimmer
setting can be found by ear, adjusting the volume of the rear speakers
for the desired balance. There is no need for a printed circuit board
for this project. The 1 W resistors can be soldered directly to the
connections of the multi-turn presets, and so the whole thing can be
assembled ‘in the air’ and shrouded in heat-shrink tubing. The circuit
can then be tucked away in the space behind the radio itself.

Author: Jörg Ehrig (Germany) – Copyright: Elektor Electronics 2011

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USB Powered Audio Power Amplifier

This circuit of
multimedia speakers for PCs has single-chip-based design, low-voltage
power supply, compatibility with USB power,
easy heat-sinking, low cost, high flexibility and wide temperature
tolerance. At the heart of the circuit is IC TDA2822M. This IC is, in
fact, mono-lithic type in 8-lead mini DIP package. It is intended for use as a dual audio power amplifier in battery-powered sound players.

Specifications of TDA2822M are low quiescent current, low crossover
distortion, supply voltage down to 1.8 volts and minimum output power of
around 450 mW/channel with 4-ohm loudspeaker at 5V DC supply input. An
ideal power amplifier can be simply defined as a circuit that can
deliver audio power into external loads without generating significant
signal distortion and without consuming excessive quiescent current.

This circuit is powered by 5V DC supply available from the USB
port of the PC. When power switch S1 is flipped to ‘on’ position, 5V
power supply is extended to the circuit and power-indicator red LED1
lights up instantly. Resistor R1 is a current surge limiter and
capacitors C1 and C4 act as buffers. Working of the circuit is simple.
Audio signals from the PC audio socket/headphone socket are fed to the
amplifier circuit through components R2 and C2 (left channel), and R3
and C3 (right channel).

Circuit Diagram

USB Powered Audio Power Amplifier Circuit

USB Powered Audio Power Amplifier Circuit Diagram

Potmeter VR1 works as the volume controller for left (L) channel and
potmeter VR2 works for right ® channel. Pin 7 of TDA2822M receives the
left-channel sound signals and pin 6 receives the right-channel signals
through VR1 and VR2, respectively. Ampl i f ied signals for driving the
left and right loudspeakers are available at pins 1 and 3 of IC1,
respectively. Components R5 and C8, and R6 and C10 form the traditional
zobel network.

Assemble the circuit on a medium-size, general-purpose PCB
and enclose in a suitable cabinet. It is advisable to use a socket for
IC TDA2822M. The external connections should be made using suitably
screened wires for better result.

Author: T.K. Hareendran – Copyright: EFY Mag

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TDA2052 Active Audio System

This active audio audio
system use three TDA2052 chips and 5 speakers ( one woofer, two tweeters
and two midranges ). For this TDA2052 active audio system we need dual
20 volts power supply and five volts supply for the stand by function.To
the input of the every audio IC chip is placed an audio filter for
filtering the audio signal for used speakers ( low pass for woofer ,
high pass for midranges and tweeters)

The subwoofer plays the 20 to 300 Hz frequency range, while the
remaining 300 Hz to 20KHz are sent to two separate channels with stereo
effect.If one of the amplifier is affected by clipping distortion the
others amplifiers are not affected .

TDA2052 Active Audio System

TDA2052 Active Audio System

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