USB FM Transmitter

Here's a simple VHF FM transmitter that could be used to play audio files from an MP3 player or computer on a standard VHF FM radio. The circuit use no coils that have to be wound. This FM transmitter can be used to listen to your own music throughout your home. When this FM transmitter used in the car, there is no need for a separate input to the car stereo to play back the music files from your MP3 player.

To keep the circuit simple as well as compact, it was decided to use a chip made by Maxim Integrated Products, the MAX2606 [1]. This IC from the MAX2605-MAX2609 series has been specifically designed for low-noise RF applications with a fixed frequency. The VCO (Voltage Controlled Oscillator) in this IC uses a Colpitts oscillator circuit. The variable-capacitance (varicap) diode and feedback capacitors
for the tuning have also been integrated on this chip, so that you only need an external inductor to fix the central oscillator frequency.

It is possible to fine-tune the frequency by varying the voltage to the varicap. Not much is demanded of the inductor, a type with a relatively low Q factor (35 to 40) is sufficient according to Maxim. The supply voltage to the IC should be between 2.7 and 5.5 V, the current consumption is between 2 and 4 mA. With values like these it seemed a good idea to supply the circuit with power from a USB port.

A common-mode choke is connected in series with the USB connections in order to avoid interference between the circuit and the PC supply. There is not much else to the circuit. The stereo signal connected to K1 is combined via R1 and R2 and is then passed via volume control P1 to the Tune input of IC1, where it causes the carrier wave to be frequency modulated. Filter R6/C7 is used to restrict the bandwidth of the audio signal. The setting of the frequency (across the whole VHF FM broadcast band) is done with P2, which is connected to the 5 V supply voltage.

The PCB designed uses resistors and capacitors with 0805 SMD packaging. The size of the board is only 41.2 x 17.9 mm, which is practically dongle-sized. For the aerial an almost straight copper track has been placed at the edge of the board. In practice we achieved a range of about 6 metres (18 feet) with this. There is also room for a 5-way SIL header on the board. Here we find the inputs to the 3.5 mm jack plug, the input to P1 and the supply voltage. The latter permits the circuit to be powered independently from the mains supply, via for example three AA batteries or a Lithium button cell. Inductor L1 in the prototype is a type made by Murata that has a fairly high Q factor: minimum 60 at 100 MHz.

Take care when you solder filter choke L2, since the connections on both sides are very close together. The supply voltage is connected to this, so make sure that you don’t short out the USB supply! Use a resistance meter to check that there is no short between the two supply connectors before connecting the circuit to a USB port on a computer or to the batteries.

P1 has the opposite effect to what you would expect (clockwise reduces the volume), because this made the board layout much easier. The deviation and audio bandwidth varies with the setting of P1. The maximum sensitivity of the audio input is fairly large. With P1 set to its maximum level, a stereo input of 10 mVrms is sufficient for the sound on the radio to remain clear. This also depends on the setting of the VCO. With a higher tuning voltage the input signal may be almost twice as large (see VCO tuning curve in the data sheet). Above that level some audible distortion becomes apparent. If the attenuation can’t be easily set by P1, you can increase the values of R1 and R2 without any problems.

Measurements with an RF analyzer showed that the third harmonic had a strong presence in the transmitted spectrum (about 10 dB below the fundamental frequency). This should really have been much lower. With a low-impedance source connected to both inputs the bandwidth varies from 13.1 kHz (P1 at maximum) to 57 kHz (with the wiper of P1 set to 1/10).

In this circuit the pre-emphasis of the input is missing. Radios in Europe have a built-in de-emphasis network of 50 μs (75 μs in the US). The sound from the radio will therefore sound noticeably muffled. To correct this, and also to stop a stereo receiver from mistakenly reacting to a 19 kHz component in the audio signal, an enhancement circuit Is published elsewhere in this issue (Pre-emphasis for FM Transmitter, also with a PCB). Author: Mathieu Coustans, Elektor Magazine, 2009

MP3 FM Transmitter Parts List
Resistors (all SMD 0805)
R1,R2 = 22kΩ
R3 = 4kΩ7
R4,R5 = 1kΩ
R6 = 270Ω
P1 = 10kΩ preset, SMD (TS53YJ103MR10 Vishay Sfernice, Farnell # 1557933)
P2 = 100kΩ preset, SMD(TS53YJ104MR10 Vishay Sfernice, Farnell # 1557934)
Capacitors (all SMD 0805)
C1,C2,C5 = 4μF7 10V
C3,C8 = 100nF
C4,C7 = 2nF2
C6 = 470nF
Inductors
L1 = 390nF, SMD 1206 (LQH31HNR39K03L Murata, Farnell # 1515418)
L2 = 2200Ω @ 100MHz, SMD, common-mode choke, 1206 type(DLW31SN222SQ2L Murata, Farnell #1515599)
Semiconductors
IC1 = MAX2606EUT+, SMD SOT23-6 (Maxim Integrated Products)
Miscellaneous
K1 = 3.5mm stereo audio jack SMD (SJ1-3513-SMT
CUI Inc, DIGI-Key # CP1-3513SJCT-ND)
K2 = 5-pin header (only required in combination with 090305-I pre-emphasis circuit)
K3 = USB connector type A, SMD (2410 07 Lumberg, Farnell # 1308875)

Notice. The use of a VHF FM transmitter, even a low power device like the one described here, is subject to radio regulations and may not be legal in all countries.

Source : http://fmtvguide.blogspot.com/2009/07/mp3-fm-transmitter-circuit.htm

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Guitar and Bass Sustain Unit

We have all heard that wonderful sound of a guitar, where the note just hangs there seemingly forever (or at least until next Thursday).  Sustain can be obtained by turning the amp up full, but the rest of the band will just kill you - they need to be able to hear themselves too!  This little project is best used in the effects loop of a guitar amp (if it has one - not all do).  It can be used direct from the guitar, but the effect is not as good, since it is designed for relatively high levels (around 1 Volt).

The circuit is very simple to build, and mine is on a piece of Veroboard.  Because it can easily be built as a pedal or even into a guitar amp (such as that described in Project 27), I do expect to make PCBs available in the not too distant future depending on demand, and these may have a few additional functions as well. 

Here is not a lot to it, but the LED and LDR (Light Dependent Resistor) are critical - they must be completely enclosed in a light proof enclosure of some kind.  Vactrol make some very nice little LDR opto-isolators, but unfortunately they are not easy to get, and are fairly expensive.  The next best thing is a couple of pieces of black heatshrink tubing.  The LED and LDR must be as close to each other as possible, and a flat topped LED is recommended if you can get one. 

Note the rather unusual earth (ground) connection.  This is not a mistake in the drawing.  U2A is used to buffer the 1/2 supply voltage created by R3 and R4, and instead of using the 12V supply negative as earth, the output of U2A is used instead.  This gives a balanced supply from a single voltage source.  Note that the AC/DC adapter (plug pack or wall wart - select the term you are most comfortable with :-) must not be used to power other equipment as well, since this may cause problems.  If you wish, a conventional +/-15V supply may be used instead, and U2A will not be used.  Note that if a +/-15V supply is used, you must increase the value of R13 to about 3.3k to limit LED current to around 10mA.

All resistors are 1/4 or 1/2 Watt, and may be 1% or 5%.  R1 and R2 should be metal film for lowest noise.  Although the TL072 is suggested for the audio path, other opamps may be used as well.  Likewise, the LM1458 can also be substituted if you like.  Caps are 16V types, but higher voltage units can be used if desired.  D7 is a power on indicator, and D6 is there to prevent damage to the circuit if the polarity of the applied 12V DC is incorrect.  Be warned that the AC/ DC adapter will be damaged if the polarity is wrong, and it is left connected for any length of time.

VR1 is a simple volume control, and is used to set the output level.  VR2 is the limiting threshold control - as it is adjusted to a higher setting, the volume will decrease.  You may wish to wire the pot "backwards", so that maximum output is obtained when VR2 is set to the fully clockwise position.

U1A is the gain control stage.  Maximum gain as shown is unity, but R2 can be increased if you find that the gain is too low.  When the signal level is high enough for D1 - D4 to conduct, the LED illuminates, and reduces the gain of the input stage.  Any further increase of input voltage will just cause the LED to glow brighter, which reduces the gain.  In this way, a constant output level is maintained, since as the input signal reduces, so does the LED brightness and the stage gain increases again.

The connections shown will be fine for most purposes, but some LDRs may give distortion at low frequencies.  A 100uF cap in parallel with the LED will probably help if this is a problem.  LDRs typically have a slow "release" time.  After illumination, they take some time to return to their full dark resistance.  This characteristic is exploited here, to allow a very simple circuit with an almost perfect attack and release time for musical instrument use.

http://sound.westhost.com/

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Off Line Telephone Tester Circuit Diagram

Here is a circuit of an off-line telephone tester which does not require any telephone line for testing a telephone instrument. The circuit is so simple that it can be easily assembled even by a novice having very little knowledge of electronics. A telephone line may be considered to be a source of some 50 volts DC with a source impedance of about 1 kilo-ohm. During ringing, in place of DC, an AC voltage of 70 to 80 volts (at 17 to 25 Hz) is present across the telephone line. When the subscriber lifts the handset, the same is sensed by the telephone exchange and the ringing AC voltage is disconnected and DC is reconnected to the line. Lifting of the handset from the telephone cradle results in shunting of the line’s two wires by low impedance of the telephone instrument. As a result, 50V DC level drops to about 12 volts across the telephone instrument. During conversation, the audio gets superimposed on this DC voltage. Since any DC supply can be used for testing a telephone instrument, the same is derived here from AC mains using step-down transformer X1. Middle point of the transformer’s secondary has been used as common for the two full-wave rectifiers—one comprising diodes D1 and D2 together with smoothing capacitor C1 and the other formed by diodes D3 and D4 along with filter capacitor C2. The former supplies about 12 volts for the telephone instrument through primary of transformer X2 which thus simulates a source impedance, and a choke which blocks AC audio signals present in the secondary of transformer X2. The AF signal available in secondary of X2 is sufficiently strong to directly drive a 32-ohm headset which is connected to the circuit through headphone socket SK1 via rotary switch S2. During ringing, a pulsating DC voltage from transformer X1 via rectifier diode D5, push-to-on switch S3, and contact ‘B’ of rotary switch S2 is applied across secondary of transformer X2. The boosted voltage available across primary of transformer X2 is sufficient to drive the ringer in the telephone instrument. Please avoid pressing of switch S3 for more than a few seconds at a time to prevent damage to the circuit due to high voltage across primary of transformer X2. The circuit also incorporates a music IC (UM66) whose output is connected to secondary of transformer X2 via switch S2 after suitably boosting its output with the help of darlington transistor pair T1 and T2. This output can be used to test the audio section of any telephone instrument. After having assembled the circuit satisfactorily, the following procedure may be followed for testing a telephone instrument:
1. Connect the telephone to the terminals marked ‘To Telephone Under Test’and switch on mains (switch S1).
2. To test the ringer portion, flip switch S2 to position ‘B’ and press S3 for a moment. You should hear the ring in case the ringer circuit of the telephone under test is working. Please ensure that handset is on cradle during this test.
3. For testing the audio section, flip switch S1 to position ‘C’ and connect a headphone to socket SK1. Pick the telephone handset and speak into its microphone. If audio section is working satisfactorily, you should be able to hear your speach via the headphone. If you dial a number, you should be able to hear the pulse clicks or pulse tone in the headphone, depending on whether the telephone under test is functioning in pulse or tone mode. If the telephone under test has a built-in musical hold facility, on pressing the ‘hold’ button you should be able to hear the music. Now flip switch S2 to position ‘A’. You should be able to hear music generated by IC1 through earpiece of the handset of the telephone under test, indicating propor functioning of the AF amplifier section. The circuit can be assembled on a small piece of veroboard. Try to mount the two transformers on opposite sides of the board, displaced by 90 degrees. Always keep handy multi-type modular plugs for testing various types of telephones. Mount all switches, sockets and LEDs on the front of testing panel

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Audio Visual Ringer Circuit Diagram

Many a times one needs an ex- tra telephone ringer in an ad- joining room to know if there is an incoming call. For example, if the telephone is installed in the drawing room you may need an extra ringer in the bedroom. All that needs to be done is to connect the given circuit in parallel with the existing telephone lines using twin flexible wires. This circuit does not require any external power source for its operation. The section comprising resistor R1 and diodes D5 and LED1 provides a visual indication of the ring. Remaining part of the circuit is the audio ringer based on IC1 (BA8204 or ML8204). This integrated circuit, specially designed for telec- om application as bell sound generator, requires very few external parts. It is readily available in 8-pin mini DIP pack.
Resistor R3 is used for bell sensitivity adjustment. The bell frequency is controlled by resistor R5 and capacitor C4, and the repeat frequency is controlled by resistor R4 and capacitor C3. A little experimentation with the various values of the resistors and capacitors may be carried out to obtain desired pleasing tone. Working of the circuit is quite simple. The bell signal, approximately 75V AC, passes through capacitor C1 and resistor R2 and appears across the diode bridge comprising diodes D1 to D4. The rectified DC output is smoothed by capacitor C2. The dual-tone ring signal is output from pin 8 of IC1 and its volume is adjusted by volume control VR1. Thereafter, it is impressed on the piezo-ceramic sound generator

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Telephone Headgear Circuit Diagram

Acompact, inexpensive and low component count telecom head- set can be constructed using two readily available transistors and a few other electronic components. This circuit is very useful for hands-free operation of EPABX and pager communication. Since the circuit draws very little current, it is ideal for parallel operation with electronic telephone set. Working of the circuit is simple and straightforward. Resistor R1 and an ordinary neon glow- lamp forms a complete visual ringer circuit. This simple arrangement does not require a DC blocking capacitor because, under idle conditions, the telephone line voltage is insufficient to ionise the neon gas and thus the lamp does not light. Only when the ring signal is being received, it flashes at the ringing rate to indicate an incoming call. The bridge rectifier using diodes D1 through D4 acts as a polarity guard which protects the electronic circuit from any changes in the telephone line polarity. Zener diode D5 at the output of this bridge rectifier is used for additional circuit protection. Section comprising transistor T1, resistors R2, R3 and zener diode D6 forms a constant voltage regulator that provides a low voltage output of about 5 volts. Dial tone and speech signals from exchange are coupled to the receiving sound amplifier stage built around transistors T2 and related parts, i.e. resistors R7, R6 and capacitor C5. Amplified signals from collector of transistor T2 are connected to dynamic receiver RT-200 (used as earpiece) via capacitor C7. A condenser microphone, connected as shown in the circuit, is used as transmitter. Audio signals developed across the microphone are coupled to the base of transistor T1 via capacitor C3. Resistor R4 determines the DC bias required for the microphone. After amplification by transistor T1, the audio signals are coupled to the telephone lines via the diode bridge. The whole circuit can be wired on a very small PCB and housed in a medium size headphone, as shown in the illustration. For better results at low line currents, value of resistor R2 may be reduced after testing

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Smart Phone Light Circuit Diagram

The circuit shown here is used to switch on a lamp when the tele- phone rings, if the ambient light is insufficient. The circuit uses only two ICs and it can be implemented very easily. A light dependent resistance (LDR), with about 5 kilo-ohms resistance in the ambient light and greather than 100 kilo-ohms in darkness, is at the heart of the circuit. The circuit is fully isolated from the phone lines and it draws current only when the phone rings. The circuit provides automatic switching on of a lamp during darkness when the phone is kept in a place such as the bedroom. The lamp can be battery powered to provide light during power failure or load shedding. This avoids delay in attending to a call. The light switches off automatically after a programmable time period and it needs no attention at all. If required, the lamp lighting period can be extended by simply pressing a pushbutton switch (S1). The first part of the circuit functions as a ring detector. When telephone is on-hook, around 48V DC is present across the TIP and RING terminals. The diode in the opto-coupler is ‘off’ during this condition and it draws practically no current from he telephone lines. The opto-coupler also isolates the circuit from the telephone lines. Transistor in the opto-coupler is normally ‘off’ and a voltage of +5V is present at the ring indicator line. When telephone rings, an AC voltage of around 70-80V AC, which is present across the telephone lines, is used to turn on the diode inside the opto-coupler (IC2) which in turn switches on transistor inside the opto-coupler. The voltage at its collector passes through 0-volt level during ringing to trigger IC3 74LS123(A) monostable flip-flop. The other opto-coupler (IC1) is used to detect the ambient light condition. When there is sufficient light, LDR has a low resistance of about 5 kilo-ohms and the transistor inside the opto-coupler is in ‘on’ state. When there is insufficient light available, the resistance of LDR increases to a few mega-ohms and the transistor switches to ‘off’ state. Thus the DC voltage present at the collector of transistor inside the opto-coupler is normally 0V and it jumps to 5V when there is no light or insufficient light. The 74LS123 retriggerable monostable multivibrator is used to generate a programmable pulse-width. The first monostable 74LS123(A) generates a pulse from the trigger input available during ringing, provided its pin 2 input (marked B) is logic high (i.e. during darkness). It remains high for the programmed duration and switches back to 0V at the end of the pulse period. This high-to-low transition (trailing edge) is used to trigger the second monostable flip-flop 74LS123(B) in the same package. Output of the second monostable is used to control a relay. The lamp being controlled via the N/O contacts of the relay gets switched ‘on.’ The ‘on’ period can be extended by simply pressing pushbutton switch S1. If nobody attends the phone, the light turns off automatically after the specific time period equal to the pulse-width of the second flip-flop. The light sensitivity of LDR can be changed by changing resistance R2 connected at collector of the transistor in light monitor circuit. Similarly, switch-on period of the lamp can be controlled by changing capacitor C3’s value in the second 74123(B) monostable circuit

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Telephone Number Display Circuit Diagram

The given circuit, when connected in parallel to a telephone, dis- plays the number dialled from the telephone set using the DTMF mode. This circuit can also show the number dialled from the phone of the called party. This is particularly helpful for receiving any number over the phone lines. The DTMF signal—generated by the phone on dialling a number—is decoded by DTMF decoder CM8870P1 (IC1), which converts the received DTMF signal into its equivalent BCD number that corresponds to the dialled number. This binary number is stored sequentially in 10 latches each time a number is dialled from the phone. The first number is stored in IC5A (1/2 of CD4508) while the second number is stored in IC5B and so on. The binary output from IC1 for digit ‘0’ as decoded by IC1 is 10102 (=1010), and this cannot be displayed by the seven-segment decoder, IC10. Therefore the binary output of IC1 is passed through a logic-circuit which converts an input of ‘10102’ into ‘00002’ without affecting the inputs ‘1’ through ‘9’. This is accomplished by gates N13 through N15 (IC11) and N1 (IC12). The storing of numbers in respective latches is done by IC2 (4017). The data valid output from pin 15 of IC1 is used to clock IC2. The ten outputs of IC2 are sequentially connected to the store and clear inputs of all the latches, except the last one, where the clear input is tied to ground. When an output pin of IC2 is high, the corresponding latch is cleared of previous data and kept ready for storing new data. Then, on clocking IC2, the same pin becomes low and the data present at the inputs of that latch at that instant gets stored and the next latch is cleared and kept ready. The similar input and output pins of all latches are connected together to form two separate input and output buses. There is only one 7-segment decoder/driver IC10 for all the ten displays. This not only reduces size and cost but reduces power requirement too. The output from a latch is available only when its disable pins (3 and 15) are brought low. This is done by IC3, IC12 and IC13. IC3 is clocked by an astable multivibrator IC4 (555). IC3 also drives the displays by switching corresponding transistors. When a latch is enabled, its corresponding display is turned on and the content of that latch, after decoding by IC10, gets displayed in the corresponding display. For instance, contents of IC5A are displayed on display ‘DIS1,’ that of IC5B on ‘DIS2’ and so on. The system should be connected to the telephone lines via a DPDT switch (not shown) for manual switching, otherwise any circuit capable of sensing handset’s off-hook condition and thereby switching relays, etc. can be used for automatic switching. The power-supply switch can also be replaced then. Though this circuit is capable of showing a maximum of ten digits, one can reduce the display digits as required. For doing this, connect the reset pin of IC2, say, for a 7-digit display, with S6 output at pin 5. The present circuit can be built on a veroboard and housed in a suitable box. The displays are common-cathode type. To make the system compact, small, 7-segment displays can be used but with some extra cost. Also, different colour displays can be used for the first three or four digits to separate the exchange code/STD code, etc. The circuit can be suitably adopted for calling-line display

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