Touch-Select Audio Source

Often you need to connect output from more than one source (preamplifier) such as tape recorder/player and CD (compact disc) player to audio power amplifier. This needs disconnecting/connecting wires when you want to change the source, which is quite cumbersome and irritating. Here is a circuit that helps you choose between two stereo sources by simple touch of your hand. This circuit is so compact that it can be fixed within the audio power amplifier cabinet and can use the same power supply source. The circuit uses just two CMOS ICs and a few other componenets. The ICs used are MC14551/CD4551 (quad 2-channel analogue multiplexer) and CD4011 (quad 2-input NAND gate).

When touch-plate S1 is touched (its two plates are to be bridged using a fingertip), gate N1 output (IC1, pin 3) goes high while the output of gate N2 at pin 4 goes low. This causes selection of CD outputs being connected to the power amplifier input, which is indicated by lighting of LED1. When touch-plate S2 is touched, the outputs of gates N1 and N2 toggle. That is, IC2 pin 3 is pulled ‘low’ while its pin 4 goes ‘high’. This results in selection of tape recorder outputs being connected to the input of power amplifier. This is indicated by lighting of LED2. Pin 9 is the control pin of IC2.

In the circuit, the state of multiplexer switches is shown with pin 9 ‘high’ (CD source selected). When pin 9 is pulled ‘low’, all the switches within the multiplexer change over to the alternate position to select tape player as source. Note. Although one can connect pin 7 (VEE) of IC2 to ground, but for operation with preamplifier signals going above and below ground level, one must connect it to a negative voltage (say, –1V to –1.5V) to avoid distortion.

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 fi rst 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.

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

Stepped Volume Control

Louder music, sirens or speech in response to higher ambient noise levels? This simple circuit has the answer, and it may enable your robot to be at least as noisy or loud-mouthed as the others in an arena. The circuit consists basically of a microphone, a level detector, a 4-state counter and four analogue switches connected to a resistive ladder network. Looking at the circuit diagram, the signal from electret microphone M1 is amplified by T1 whose collector voltage appears across a potentiometer. M1 gets its bias voltage through R4. Depending on the setting of P1, the 4040 counter will get a clock pulse when a certain noise level (threshold) is exceeded.

The counter state determines the configuration of the four electronic switches inside the 4066 and so the series resistance effectively seen in the audio signal line. The circuit should be powered from a 9-V regulated supply or a battery and will consume a few milliamps only. Switch S1 allows the counter to be reset, switching all 4066 switches to off, i.e., the highest attenuation will exist in the audio path as in that case none of the 1-kΩ resistors are shorted out. To calibrate the circuit, disconnect the 4040 clock input (pin 10) from the wiper of P1, and temporarily ground it through a 100 kΩ resistor. Now pulse the clock input by briefly connecting it to the +9 V line; you will see the counter outputs change state and with them, the bilateral switches in the 4066.

Author: Raj K. Gorkhali Copyright: Elektor Electronics 2007

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).

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

Input Impedance Booster II

The input resistance of a.c.-coupled op amp circuits depends almost entirely on the resistance with which the d.c. setting is determined. If CMOS op amps are used, the input resistance is normally high, currently up to 10 MΩ. If a higher value is needed, a bootstrap circuit may be used. This enables the input resistance to be boosted artificially to a very high value, indeed In the circuit shown in the diagram, resistor R1 sets the d.c. point for IC1a. The terminal of the resistor linked to pin 7 of IC1 would normally be at earth potential, so that the input impedance would be 10 MΩ.

Connecting the other terminal of the resistor to earth via IC1a and network C2-R3-R2 as far as d.c. is concerned results in the requisite d.c. setting of the op amp. As far as alternating voltages are concerned, the input signal is fed back so that only a tiny alternating current flows through R1. Therefore, Rin=R1[(R2+R3)/R3]. With resistor values as specified, Rin is about 1 GΩ. One aspect must be borne in mind: the numerical value of (R2+R3)/R3 must not exceed 0.99. This means that the value of R3 cannot be less than 100 kΩ if the value of R2 is 10 MΩ. If these conditions are not met, the circuit will become unstable.

Pan Pot

A pan pot enables a mono-phonic input signal to be positioned where desired between the stereo loudspeakers. When P1 (see diagram) is in the center position, there is no attenuation or amplification between the input and output. When the control is turned away from the center position, the signal in one channel will be amplified 3 dB more than the other. Circuit IC1 at the input is a buffer stage.

It is arranged as an inverter to ensure that the phase of the input signal is identical to that of the output signal. The input impedance is set by R1 (10 kΩ). The output of the buffer is applied to stereo amplifiers IC2 and IC3. A special arrangement here is the positioning of P1, in conjunction with R3, R4, R8, and R9, in the feedback circuits of both amplifiers. This means that any adjustment of the potentiometer will have opposite effects in the amplifiers.

Series resistors R7 and R12 serve to ensure that the outputs can handle capacitive loads. Coupling capacitors C3, C6, and C9, may be omitted if an offset voltage of 20–30 mV is of no consequence in the relevant application. Capacitors C2, C5, and C8, ensure that the op amps remain stable even at unity gain. Capacitors C1, C4, and C7, minimize any r.f. interference, resulting in a usable bandwidth of 2.5 Hz to 200 kHz.

The performance of the circuit is of sufficiently high quality to allow the pot being incorporated in good-quality control panels. Total harmonic distortion plus noise (THD+N) at a frequency of 1 kHz and a bandwidth of 22 kHz is 0.0014%. Over the band 20 Hz to 20 kHz and a bandwidth of 80 dB, this figure is still only 0.0023%. The circuit needs a power supply of ±18 V, from which it draws a current of about 16 mA.

Sounds From The Old West

This circuit shows how far integration can be taken: IC1, a Type HT82207 from Holtek does virtually everything. Only a (small) loudspeaker and the necessary selectors need to be added. The standard 18-pin Type HT82207 is an integrated sound generator, producing sounds typical of the Old West. The various sounds are selected by S1–S6 as listed below. In the quiescent state, the circuit draws a current not exceeding 1 µA.

• S1 – bugle
• S2 – neighing
• S3 – sound of hooves
• S4 – pistol shot
• S5 – crack of a rifle
• S6 – cannon fire

Oscillation Monitor

The circuit in the diagram was originally designed to monitor an oscillator, but can also be used as a general-purpose level indicator for a.c. signals. It is based on a quadruple IC containing four NAND gates. Only three of the gates are used, making the fourth free for other purposes. All the gates have a Schmitt trigger input.

When a 5 V supply is used, the Type 74HC132 is recommended; for higher voltage, a Type 4093. Note, however, that these two ICs have different pinouts. In the diagram, the differing pins of a 4093 are shown in brackets. The signal to be monitored is applied to the input of the first gate via capacitor C1. Resistor R2, in conjunction with the protection diode in the IC, guards the input to high voltages.

In the absence of a signal, resistor R1 holds the input high so that the output of the gate is low. When a signal of sufficient strength is received, the input of the gate goes low during the negative half cycle of the signal, so that the output of the gate goes high in rhythm with the input signal. However, the Schmitt trigger converts sinusoidal signals into rectangular ones, which charge capacitor C3 via diode D1. When the potential across C3 exceeds the threshold at the input of the second gate, this gate also toggles. The output of the second gate is then low, which disables the third gate, which functions as an oscillator. When the level of the input signal drops, C3 is discharged via R3.

The potential across the capacitor then no longer exceeds the threshold at the input of IC1b, whereupon IC1c is enabled and the LED flashes The LED may be connected as shown or as indicated by the dashed line. As shown, the diode remains off when there is an input signal of sufficient strength and begins to flash when the signal fails or its level drops. When the diode is linked to earth, it is on continuously when there is an input signal, and begins to flash when the input drops. When a 5 V power supply is used, R5 = 1 kΩ, and the circuit draws a current, including that of the LED, of 3 mA. The frequency of the input signal may lie between 10 Hz and 10 MHz. When a 9–12 V supply is used, the value of R5 must be altered as necessary.

Owing to the 4093 being slower than the 74HC132, the upper frequency of the input signal is then limited to 3 MHz. When the wiper of P1 is at the level of the supply voltage, the response threshold, USS, lies between 3.5 V (when Ub =5V) and 7 V (when Ub =12V). When the wiper is moved away from the positive supply line, USS (max) is 1.5 V (when Ub = 5 V). The response threshold is quite precise: a drop in the input signal level of 50–100 mV is sufficient to disable the input. When the input level is too high, a preset across the input terminals enables the level to be reduced to a value that lies in the desired range above the response threshold.

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.

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.

Turtle Beach Anounces iSeries Media Headsets.

When it comes to audio brands, there is one particular name that stands out above the rest for some people, and Turtle Beach would be one of them. The name Turtle Beach has long been synonymous with high quality audio devices, and this time around the company has announced their spanking new iSeries range of high end media headsets.

The iSeries will arrive in the form of the wireless surround sound i60 and the wireless amplified stereo i30, where these would no doubt be the first headsets that have been specially constructed for use with both Mac and iOS products. The iSeries headsets themselves will arrive with a bunch of innovations that Turtle Beach originally introduced to gaming headsets but so happen to be new for mobile users. Hopefully, with a little bit of something for everyone, folks will be satisfied.

Some of the new features introduced to gaming headsets that mobile users would fall in love with include improved chat and voice call quality as well as an unmatched level of control over audio. Regardless of whether you settle for the i60 or the i30, they will both share the same unique design which blends style, quality, comfort and durability, while boasting premium finishes, where among them include a leather headband and leather-covered memory foam, noise-isolating ear cups.

The two headsets will also be accompanied by a built-in remote which plays nice with the iPhone, iPad and the Mac. They will come with innovative dual boom-less microphones which will hopefully, deliver clear, high-quality audio during chat, voice calling and use with Siri.

Of course, the i60 will be the flagship model, where it features a control unit for use with Mac desktops and laptops that deliver fingertip audio and preset control. Turtle Beach does have some pretty high hopes that the i60 will also be the first wireless headset for the Mac platform that boasts of 7.1 DTS surround sound with adjustable speaker angles and EQ presets, a fully programmable digital signal processor (DSP), dual-band Wi-Fi for uncompressed, interference free wireless audio and dual-pairing Bluetooth 4.0 connectivity. [Via]