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الکترونیک نقشه مدار شماتیک پروژه مدار چاپی دیتاشیت اموزش

                            

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+ نوشته شده در  86/09/11ساعت   توسط احد دانشمند 

  

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جمهوری اسلامی ايران

333

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6

1.69%

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1

0.28%

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1

0.28%

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1

0.28%

امارات

1

0.28%

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1

0.28%

رومانی

1

0.28%

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1

0.28%

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3

0.33%

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3

0.33%

ترکيه 

  1  

0.93%

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1

0.93%

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1

31%

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1

0.93%

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8

2.25%

 

 

 
 
 
  
 

 

+ نوشته شده در  86/09/11ساعت   توسط احد دانشمند 

FM Telephone Bug


 
 

Schematic

This is the schematic of the phone transmitter

Parts

Part
Total Qty.
Description
Substitutions
R1 1 180 Ohm 1/4 W Resistor
R2 1 12K 1/4 W Resistor
C1 1 330pF Capacitor
C2 1 12pF Capacitor
C3 1 471pF Capacitor
C4 1 22pF Capacitor
Q1 1 2SA933 Transistor
D1, D2, D3, D4 4 1SS119 Silicon Diode
D5 1 Red LED
S1 1 SPDT Switch
L1 1 Tuning Coil
MISC 1 Wire, Circuit Board

Notes

  1. L1 is 7 turns of 22 AWG wire wound on a 9/64 drill bit. You may need to experiment with the number of turns.

  2. By stretching and compressing the coils of L1, you can change the frequency of the transmitter. The min frequency is about 88 Mhz, while the max frequency is around 94 Mhz.

  3. The green wire from the phone line goes to IN1. The red wire from the phone line goes to IN2. The green wire from OUT1 goes to the phone(s), as well as the red wire from OUT2.

  4. The antenna is a piece of thin (22 AWG) wire about 5 inches long.

  5. All capacitors are rated for 250V or greater.

  6. The transmitter is powered by the phone line and is on only when the phone is in use. S1 can be used to turn the transmitter off if it is not needed.

  7. If you have problems with the LED burning out, then add a 300 ohm 1/4W resistor in series with it.

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Telephone Recorder

 

                        ***   بزودی تمامی مدارات با ترجمه فارسی ارائه میشود  ***


 
This nifty little circuit lets you record your phone conversations automatically. The device connects to the phone line, your tape recorder's microphone input, and the recorder's remote control jack. It senses the voltage in the phone line and begins recording when the line drops to 5 volts or less.

Schematic

This is the schematic of the Telephone Recorder

Parts

Part
Total Qty.
Description
Substitutions
R1 1 270K 1/4 W Resistor
R2 1 1.5K 1/4 W Resistor
R3 1 68K 1/4 W Resistor
R4 1 33K 1/4 W Resistor
C1 1 0.22uF 150 Volt Capacitor
Q1, Q2 2 2N4954 NPN Transistor
D1 1 1N645 Diode
MISC 1 Wire, Plugs To Match Jacks On Recorder, Board, Phone Plug

Notes

  1. The circuit can be placed anywhere on the phone line, even inside a phone.

  2. Some countries or states require you to notify anyone you are talking to that the conversation is being recorded. Most recoders do this with a beep-beep. Also, you may have to get permission from the phone company before you connect anything to their lines.
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Telephone Hold Button


 
Although a hold feature is standard on most new phones, a lot of us still use the origional bell phones. Those of us that require a hold feature will find this circuit very useful. It is easy to build, and is compact enough to be installed inside the phone with no real problem. It is also powered by the phone line itself, eliminating the need for batteries.

Schematic

                                          This is the schematic of the Phone Hold Button

Parts

Part
Total Qty.
Description
Substitutions
R1 1 1.5K 1/4 W Resistor
R2 1 1K 1/4W Resistor
D1 1 1N4002 Silicon Diode 1N4003, 1N4004, 1N4005, 1N4006, 1N4007
SCR1 1 C106Y SCR
LED1 1 Red LED Green LED, Yellow LED
S1 1 Normally Open Push-Button Switch
MISC 1 Wire, Board, Case (If Used)

Notes

  1. To place a call on hold, simply hold down the button while hanging up the phone. To take a call off hold, just pick up the phone, or any extension.

  2. Even though this is a simple circuit, you may have to check with your phone company before use.
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Simple Phone Tap


 
This circuit is extremely simple, therefore there is less chance of any problems. It can be placed anywhere on the phone line and it will record any conversation on any phone on that line.

Please note: I have received several emails saying that this circuit will not work and that it may hold your line off hook and to me it looks like it will (it would put quite a load on the phone line). For some, it has worked fine. Build at your own risk.

Schematic

This the Schematic of the Simple Phone Tap

Parts

Part
Total Qty.
Description
Substitutions
R1 1 470 Ohm 1/4 Watt Resistor
R2 1 1K 1/4 Watt Resistor
R3 1 100K 1/4 Watt Resistor
R4 1 6K 1/4 Watt Resistor
C1, C2 2 0.01 uF 100V Ceramic Capacitor
K1 1 24VDC Reed Relay
MISC 1 Wire, Headphone Plugs, Phone Plug Or Alligator Clips

Notes

  1. To use the circuit, simply connect it in series to the phone. Then plug in the recorder. It will start recording when the phone is picked up.

  2. The values of R1 and R2 may have to be adjusted, depending on the characteristics of the phone line it is connected to (they vary). If the R1+R2 combination is increased sufficiently (try 2.47K) then the circuit can be connected in parallel with the line, thus covering all attached devices. ().

  3. T.A. Betts offers this suggestion for getting this problematic circuit working: "If you drop the resistors and add a 2.2uF capacitor the line will not stay off hook. It blocks the DC from the moniter position."

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Ringing Phone Light Flasher

 
I have received several emails asking how to connect up some lights so that when the phone rings, they flash. This is very useful in a situation were there is lots of noise and it is impossable to hear the phone, such as a workshop. Here is such a device.

Schematic

Schematic for phone flasher

Parts

Part
Total Qty.
Description
Substitutions
C1 1 0.47uF 250V Capacitor
R1, R2 2 10K 1/4 W Resistor
R3 1 1K 1/4W Resistor
D1, D2 2 20V 1/4W Zener Diode
D3 1 1N4148 Diode
Q1 1 2N3904 NPN Transistor 2N2222
U1 1 4N27 Opto Isolator
RELAY 1 Solid State Or Regular Relay (See Notes)
MISC 1 Case, Wire, Board

Notes

  1. You may need to use a lower voltage zener for D1 and D2.

  2. You can use a regular relay instead of a solid state relay, but the arcing accross the contacts may destroy it pretty quickly.

  3. Be very sure that you have not accidentally connected 120V to the phone line when building and installing this circuit.
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Phone Busy Indicator


 
Have you ever been using the modem or fax and someone else picks up the phone, breaking the connection? Well, this simple circuit should put an end to that. It signals that the phone is in use by lighting a red LED. When the phone is not in use, a green LED is lit. It needs no external power and can be connected anywhere on the phone line, even mounted inside the phone.

Schematic

This is a schematic of the Phone Busy indicator

Parts

Part
Total Qty.
Description
Substitutions
R1 1 3.3K 1/4 W Resistor
R2 1 33K 1/4 W Resistor
R3 1 56K 1/4 W Resistor
R4 1 22K 1/4 W Resistor
R5 1 4.7K 1/4 W Resistor
Q1, Q2 2 2N3392 NPN Transistor
BR1 1 1.5 Amp 250 PIV Bridge Rectifier
LED1 1 Red LED
LED2 1 Green LED
MISC 1 Wire, Case, Phone Cord

Notes

  1. This is a very simple circuit and is easily made on a perf board and mounted inside the phone.

  2. LED1 and LED2 flash on and off while the phone is ringing.

  3. Do not worry about mixing up the Tip and Ring connections.

  4. The ring voltage on a phone line is anywhere from 90 to 130 volts. Make sure no one calls while you are making the line connections or you'll know it. :-)

  5. In some countries or states you will have to ask the phone company before you connect this to the line. It might even require an inspection.

  6. If the circuit causes distortion on the phone line, connect a 680 ohm resistor in between one of the incoming line wires and the bridge rectifier.

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Cut Phone Line Detector


 
A while ago I got an email asking for the schematic of a circuit to detect cut phone lines. It didn't take me long to find this circuit in . When the circuit detects that a phone line has been cut, it activates a MOSFET which can be used to drive a relay, motor, etc. It can also be connected to a security system.

Schematic

This is the schematic of the cut phone line detector

Parts

Part
Total Qty.
Description
Substitutions
R1, R2, R3 3 22 Meg 1/4 W Resistor
R4 1 2.2 Meg 1/4 W Resistor
C1 1 0.47uF 250V Mylar Capicitor
Q1 1 2N3904 Transistor 2N2222
Q2 1 2N3906 Transistor
Q3 1 IRF510 Power MOSFET
D1 1 1N914 Diode
Load 1 See "Notes"
MISC 1 Wire, Phone Connectors, Circiut Board

Notes

  1. The "Load" can be a relay, lamp, motor, etc. The circuit can also be connected to a security system to sound an alarm in case the phone line is cut.

  2. If the circuit is connected to a security system or other circuit, both circuits must be electrically isolated from each other using an opto-isolator, relay, etc. This also means that the Cut Phone Line Detector must be powered by a seperate 9V supply.
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                              مدارات الکترونیک دانشمند

Guitar and Bass Sustain Unit


Introduction

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.


Description

The complete schematic is shown in Figure 1.  There 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.

Figure 1
Figure 1 - Guitar and Bass Compressor

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.

I have also used mine on music, and it gives a very good account of itself - so much so that I would recommend this unit as a simple compressor for almost any application.


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             مدارات الکترونیک دانشمند

Musical Instrument (Expandable) Graphic Equaliser


Introduction

This equaliser is designed as a preamp suitable for musical instruments - guitar, bass and keyboard in particular. Unlike most conventional graphic equalisers, each slider ranges from fully off to fully on, and not the more conventional +/-12dB or so that is normally available.

As a result, there is no flat setting (other than all off!). This graphic is designed to be used to create a sound, and is not suitable for hi-fi. It may be used as an add-on unit to existing instrument amp preamps, tone controls, etc. The flexibility is extraordinary, allowing a hollow "single frequency" type sound, right through to almost any tonal variant imaginable.

This is the first of several projects based on the multiple-feedback bandpass filter described in Project 63, it can be made with as many (or as few) filter sections as you want.

Because of the repetetive nature of the filter units, I will be designing a PCB for them at some time in the future (depending on demand). One board will carry two or 4 filters, and the boards will be quite small so they can be packed into a case easily. The remainder of the circuitry can easily be constructed on Veroboard or similar.


Description

The input circuit is completely conventional, and uses 1/2 of a dual opamp as the initial gain stage. This is followed by the volume control, second gain stage and buffer. The output of the buffer is fed to the inputs of the filter stages, each of which has a slider for its specific frequency. The outputs of the sliders are summed using another opamp, and a distortion effect is included in the final output stage. This can be left out altogether if distortion is not desired.

If used for guitar, the frequencies needed only have to range from 80Hz to about 7kHz, but to make the unit more versatile I suggest that the lowest frequency should be 31Hz, and the highest around 12kHz. This can be extended if you want.

Decisions!
Now you have to decide on the frequency resolution. 1/3 octave would be really nice, but the number of sliders can be a nightmare. At the very least, you will need octave band, and the suggested frequencies are ...

31   63   125   250   500   1k0   2k0   4k0   8k0   16k

Should you decide on 1/2 octave band frequencies, 20 sliders will cover the range suggested (plus a bit) - these might be ...

31   44   63   87   125   175   250   350   500   700   1k0   1k4   2k0   2k8   4k0   5k6   8k0   11k   16k   20k

The 20kHz filter can be (should be?) left off for instrument use, so that means only 19 slide pots will ne needed. Lastly, 1/3 octave band needs 30 sliders to cover the full frequency range, but the 25Hz and 20kHz bands will not be needed. This still requires 28 slide pots, but the flexibility is greater than you will ever get with conventional tone controls ...

31 40 50 63 80 100 125 160 200 250 315 400 500 630 800 1k0 1k2 1k6 2k0 2k5 3k2 4k0 5k0 6k3 8k0 10k 12k 16k

There is no reason at all that the unit has to be 1/2 octave or 1/3 octave all the way. The midrange can be 1/3 octave for finest control, but go to 1/2 octave at the extremes. Especially for guitar and bass, I would prefer 1/3 octave up to 1kHz, then 1/2 octave from 1kHz to 8kHz. The final slider would be a 1 octave band filter at 16kHz. The sequence now looks like this ...

31 40 50 63 80 100 125 160 200 250 315 400 500 630 800 1k0 1k4 2k0 2k8 4k0 5k6 8k0 16k

This gives 23 filters and slide pots, a reasonable compromise that should give excellent results. To ensure reasonable continuity, the filters at 1kHz and 8kHz will need to be a compromise. 1/3 octave filters need a Q of 4, and 1/2 octave filters use a Q of 3, so the 1kHz filter will actually have a Q of 3, and the 8kHz filter will be best with a Q of 2. This might look daunting, but the MFB Filter design program will make short work of determining the component values. Unfortunately, this is only available for users of Microsoft Windows. Note that you will also need the Visual Basic 4 (VB4) runtime library, which can be obtained from Annoyances.org (easy) or the Microsoft support Website (less easy).

If you want to use the frequencies shown above, the table at the end of this page shows the values for each filter.

The Circuit
Figure 1 shows the schematic of the input section, and is virtually identical to the guitar preamp presented in Project 27. The two input jacks allow rudimentary mixing of two sources, but are mainly designed to provide a high gain and a low gain input to help prevent input stage overload. The "Hi" input connects the signal directly to the opamp input, and the "Lo" introduces a 6dB loss to allow for high output pickups. The buffer stage has an effective load of about 810 ohms - a difficult load for an opamp to drive. I suggest that an NE5532 opamp is used for U1, as it is one of the few that can drive such a load without difficulty. Although a TL072 can be used, this should be for testing or as a last resort. Pinouts are the same for both types, but the NE5532 is more critical of supply bypassing, and the addition of 100nF ceramic caps from each supply to ground is strongly recommended (as shown). These should be as close to the IC package as possible.

Figure 1
Figure 1 - Instrument Equaliser Input Stage & Buffer

The filters and slider pots (with their mixing resistors) are shown in Figure 2. To see the actual filter circuit, refer to Project 63, it is far too cumbersome to draw each of these in full! Even so, only six of the 23 filters are shown. There is one filter module and one slider for each frequency. For guitar especially, you might want to provide more gain for the higher frequencies (typically from about 2kHz to 8kHz). No problem. Since the mixing resistors are nominally 100k, starting from the 1k4 slider, drop the value to 82k, then use 47k resistors for the remaining bands. This gives a 6dB increase in top-end boost which should be sufficient (you can have more, but this will increase the noise level).

Figure 2
Figure 2 - Filter Bank (Part), Slide Pots and Mixing Resistors

The filters do not need really quiet opamps, and considering the number this would be prohibitively expensive. The opamps do need to be at least to the standard of the TL072 or filter performance will suffer. The suggested frequency ranges will give good performance at all frequencies, since the Q (and hence the demands on the opamps) is reduced as the frequency increases.

Finally, the mixer and output stage are shown in Figure 3. The mixer is a conventional "virtual earth" type, and minimises interaction between the slide pots. The distortion stage uses the diodes (all 1N4148 types) as a clipping circuit, and in conjunction with VR24 (Master Volume) allows the amount of distortion to be adjusted from zero to 'heavy metal' (aka 'grunge'). It may be necessary to use more diodes than the 4 shown. An additional 4 diodes will raise the maximum output level to about 1,5V RMS before clipping starts. The final opamp is a buffer, and contributes no gain.

Figure 3
Figure 3 - Mixer and Distortion Circuits

A word of warning. Don't expect this preamp to be especially quiet, because it won't be. Use of a low noise opamp for the mixer helps, but as with all guitar amps, some noise is inevitable. This is made worse by all the filter circuits, but each only adds noise in its own band, so the cumulative noise is not as great as it might be. Using the distortion control will increase noise, and this can be dramatic at full distortion. In reality, this is not much different from a conventional guitar preamp that is turned up LOUD to get the same distortion. The more gain you have, the greater the noise (ye cannae change the laws of physics!).

Using the equaliser is simplicity itself. Just slide sliders up and down to get the sound you want. There is no "correct" way to use this unit - it is designed to enable you to get sounds. As described above, you can get more of any given frequency by reducing the value of the mixing resistor, but there is a limit to how much noise is tolerable.

The total gain of the unit (with all sliders at maximum) is about 15 times for the input stage, and a further 7.6 for the mixer (using all 100k resistors). This gives a total gain of 113 (or 41dB). Actual gain will be different, depending on the slider setting, and can be increased (or reduced) by changing the value of R33 (lower the value for less gain and vice versa) or R7 (lower value gives more gain). If you change the gain structure, be careful that the input gain is not made too high, or you will get distortion with high output pickups.

To power the circuit, any power supply capable of +/-15V (+/-12V at a pinch) will do, provided that it is capable of 100mA or so.


Filter Component Values

The table shows the values I calculated for each filter. Component references are based on the diagram in Project 63, which is reproduced here for convenience (pin connections are shown for a single opamp). For this application, omit C3, R4 and short the non-inverting opamp input to ground.

Figure 4
Figure 4 - Multiple Feedback Bandpass Filter

Freq R1 R2 R3 C1, C2 Freq R1 R2 R3 C1, C2
31 82k 2k7 160k 220nF 500 27k 820 56k 47nF
40 82k 2k7 160k 180nF 630 27k 820 56k 39nF
50 82k 2k7 160k 150nF 800 27k 820 56k 27nF+2n7
63 82k 2k7 160k 120nF 1k0 8k2 510 18k 47nF+4n7
80 82k 2k7 160k 100nF 1k4 8k2 510 18k 39nF
100 82k 2k7 160k 82nF 2k0 8k2 510 18k 27nF
125 82k 2k7 160k 56nF+5n6 2k8 8k2 510 18k 18nF+1n5
160 82k 2k7 160k 47nF 4k0 8k2 510 18k 12nF+1n8
200 82k 2k7 160k 39nF 5k6 8k2 750 18k 8n2
250 82k 2k7 160k 27nF+4n7 8k0 8k2 1k2 18k 4n7
315 82k 2k7 160k 22nF+2n7 16k 8k2 1k2 18k 2n2
400 82k 2k7 160k 18nF+1n5

I have tried to keep the values reasonably sensible. This is not easy with 1/3 octave band equalisers, but all in all the results are quite acceptable (not too many different values). Note that the Q of the filters is changed as the frequency increases - feel free to use the calculator to reverse calculate the values to see the actual gain, Q and frequency error. None of these will be significant in use.


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                مدارات الکترونیک دانشمند

Ultra Simple Bass Guitar Compressor


Introduction

Using a compressor (or to be more correct, a peak limiter) on bass guitar is one sure way to get more apparent volume, without distortion.  A good bass compressor will often have a relatively slow attack, so that you get a very solid "chunky" start to each note, with lovely sustain and equal volume for all notes (speaker box allowing, of course).

The project described here is one that you can just build and have working straight away (although it is entirely possible that you will want to experiment a bit), and requires only a small handful of parts.  In its simplest form, there are no active electronics at all - and this is exactly what is described.

In case you were wondering, it can also be used with guitar, and can give excellent sustain, although this circuit is too slow to give perfect results.


Description

The way a compressor / limiter works is quite simple.  Once the preset threshold has been reached, the gain of the amplifier is reduced to maintain the output at the preset level.  As the signal decays, the gain is allowed to increase again to compensate, until the amp is at full gain and the signal then dies out naturally.

The unit described here uses a light dependent resistor (LDR) and a small light globe, of the type commonly referred to as a "grain of wheat".  These are very small, and having a small filament, they react quite quickly to an applied signal.  LDRs have a very high resistance when dark, and this falls as more light is received.  Typical LDRs will have a dark resistance of several megohms, and a minimum resistance of about 200 ohms or so.  The distortion introduced is very slight (typically less than 0.5%), especially at low levels.

Now, if the lamp were to be placed across the speaker output of your amp, and its light shines on an LDR, as the light gets brighter, the LDR will have less resistance.  The LDR is arranged in the circuit to form a voltage divider, so that as the resistance decreases, the input level is reduced, and a simple limiter is operational.

The problem with this approach is that the lamp will start to glow brightly enough to reduce the input signal with only a few volts of speaker output, so the level will be very low - with only a few watts of speaker drive.  This is fixed by using a wirewound pot across the speaker terminals, so that the amount of output signal getting to the lamp can be varied.  In this way, the output level is set by the pot, and the amount of compression is set by the amplifier's volume control.  Figure 1 shows the complete circuit.

Figure 1
Figure 1 - Simple Bass Guitar Compressor

The pot will need to be rated at 3W, and with a 500 Ohm pot, it can be used with amplifier powers up to a bit over 150W into 8 Ohms (or 300W into 4 Ohms).  It is very important that the input section is properly shielded, otherwise the amplifier may oscillate, and the lamp and LDR must not be placed too close together for the same reason.

Ideally, you will use a small piece of clear Perspex rod, with a hole drilled into one end to take the lamp.  The LDR is then glued to the other end using a transparent adhesive (model glue is ideal).  Figure 2 shows the suggested method of assembly, which will ensure that you don't have problems with oscillation from the amp.  Don't glue the lamp in place, as you will probably have to replace it at some time or another.  These little lamps normally will last a long time in this sort of circuit, but they will eventually fail, so keep a spare in the box.  You might want to connect the lamp using a small screw-down terminal block, so that a replacement can be made without having to use a soldering iron.

When the light pipe is completed, wrap the LDR end with aluminium foil, and tightly twist a bare wire around the foil to make good contact.  Tape the assembly firmly so that nothing comes undone.  This acts as a shield, and is connected to the earth (ground) connection on the input jack.  Make sure that the foil does not short circuit the LDR leads, or you will get no signal at all.  Note that one of the LDR leads will be connected to ground anyway - it does not matter which one.

Figure 2
Figure 2 - Assembly Of The Compressor

The complete unit should be housed in a metal box that is completely light proof.  Any ambient light that penetrates the box will affect the LDR, and will either introduce hum or cause greatly reduced performance (or both).  The die-cast aluminium boxes available from many retail electronics suppliers are ideal, as they are very robust, and provide excellent shielding.

Make sure that the speaker connectors are of the insulated type, because some amplifiers do not use earth referenced outputs.  Failure to ensure that these connectors are properly insulated may damage the amp or cause the amp to oscillate.  Also make sure that the speaker leads are kept well away from the input connectors.  If necessary, a shield may be made from thin metal and used to separate the two halves of the circuit.

Using The Compressor
Plug the bass directly into the input, and another lead from the output to the input on the amp.  Plug a spare speaker lead into the speaker input (or run a lead from the amp to the Speaker In jack, and another from the Speaker Out to the loudspeaker.  The speaker sockets are completely interchangeable, so you can use either for In or Out.  The Input and Output jacks are NOT interchangeable, although the circuit will still seem to work (just not as well, and the tone will go all funny as the LDR loads down the pickups).

Set the 500 Ohm pot fully off, and play a note or three.  Once you are satisfied that all is well, turn the pot to maximum, and increase the volume a little.  When you play a note, there should be a solid attack, and then the level should quickly stabilise, but at a greatly reduced volume.  You will find that you can get really good sustain, and you simply play about with the compressor pot and the amp's volume control to get the sound you want at the volume you need.  The apparent loudness will increase (often by a large margin) because the amp can be consistently driven harder, but will not distort.  You will get some distortion during the attack period, but (surprise) this can often be used as a sound in itself, and is not unpleasant because of the short duration.

So, there it is.  Not much electronics, but more of an exercise in construction.  It works surprisingly well, and I think you will have lots of fun with it.  The attack time is a wee bit long for guitar, but the sound is not unpleasant, and you can also increase the gain (a lot!) and get amazing sustain with minimal distortion.


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Spring Reverb Unit For Guitar or Keyboards
 Updated 17 November 2006


Introduction

Well, its not really just for guitar or keyboards, you can use it for anything that you want. Spring reverb units are most commonly used in guitar amps, having been replaced by digital effects in most other areas. This cannot really be classed as a "real" project, because the circuitry is somewhat experimental, and may change quite dramatically depending on the type of spring reverb unit you can actually get your hands on.

The one I have is an Accutronics (they are still going, so check out their web site - see below), but you might already have one, or can get something different, so you will have to experiment.

Most of the possibilities are discussed here, so with a small amount of mucking about you should be able to create a reverb unit tailored to your exact needs. For additional information, see the end of this page.

Since the P113 headphone amp is very easily modified for constant current drive, this is recommended. The circuit diagram in Figire 3 is the original, but it does have limitations. The main limit is the allowable voltage swing, but this is overcome by using the P113 board with appropriate modifications (all described in the construction article).


Description

The basic spring reverb chamber is a simple affair (see Figure 1), with an input and output transducer, and one or more (usually three or four) springs lightly stretched between them. Each spring should have different characteristics, to ensure that the unit does not simply create "boinging" noises. Stay well clear of single spring units, they are usually cheap Taiwanese affairs and can often found in really cheap guitar amps. They sound awful, and nothing you do will ever change this. This is not to say that the Taiwanese don't make decent spring reverb units too, I just haven't seen one yet.

Fig 1
Figure 1 - Basic Spring Reverb Unit

Many reverb units appear to have only two springs, but you will see that there are joins in the middle. This is where two springs are joined, and each spring should be very slightly different. Ultimately it doesn't matter how many springs they really have, a spring reverb always sounds like what it is. This is not a criticism, merely a description of the sound.

Of the units around, most of the newer ones have a low impedance (about 8 Ohms) input transducer, and are well suited to being driven with a small power amp IC. The one I have has a relatively high input impedance (173 Ohms DC resistance, and according to the specs, about 1700 Ohms impedance), but the principles are still pretty much the same.

Another common type of reverb tank (common terminology, BTW), is the folded spring type. These have the springs arranged in a Z pattern and sound quite good. They have been used by some very well known guitar amp manufacturers.


Transducer Drive

In all cases you will need a small power amp to be able to drive the unit properly, but you must be very careful, because overdrive causes the small pole piece to become magnetically saturated, leading to gross distortion that increases with decreasing frequency. One solution to this is to use a series resistor to reduce the drive and give a higher output impedance from the amp. This usually improves frequency response, especially at higher frequencies, but tends to disappear the bottom end. This is not always a bad thing, since in reality low frequency reverberation in a typical room or auditorium is rare, and generally sounds awful when it does exist.

Another possibility is to use an amplifier with a high output impedance, but this is not necessary because of the very low power handling of the input transducer. As a result, I suggest the series resistor method, as this is the easiest to implement, and helps to protect the transducer from gross overloads. The basic scheme is shown in Figure 2, and has the added advantage that modification to the reverb unit is not needed (many (most?) have the earth of both input and output transducers connected to the chassis - to use a current amp, this would need to be changed).

Using current drive is explained (see additional info, below) and I have used it and it works well. The problem is that it makes the reverb very "toppy", with very little bass at all. While this might suit some players, I prefer a modified current drive, where the output impedance is defined (rather than "infinite") because you can tailor the sound to your liking much more easily. This is a little tricky with the small power amp ICs though. If you want more information on this, send me an e-mail - if I get enough interest I will work something out.

Fig 2
Figure 2 - Reverb Input Transducer Drive Amp

I have seen quite a few reverb drive amps used in other circuits, including just an opamp. Opamps do not have sufficient current capability to drive the input transducer properly, and even some of the small power amp ICs are a pain. The circuit shown has good drive, low current drain and works well. Most of the circuits I have seen also do not make any attempt to obtain current drive, and use the low impedance output from the drive amp. This is not the best way to drive these transducers, and the method shown works much better.

The resistor marked S.O.T. (Select On Test) will need to be selected to provide the best reverb sound, with the minimum voltage loss. I suggest a starting value of about 47 Ohms (depends on the input transducer's impedance), and experiment from there. The positive voltage needs to be not less than 15V (18V is the rated maximum) to be able to get good drive levels with a high enough value of series resistor. In a pinch you might be able to get away with 9V, but you will not have much drive level and will need more gain at the output. This increases noise and the possibility of feedback.


Reverb Preamp

The output transducer will have an output of (typically) about 10mV, and a gain of 10 (20dB) is usually enough to match the output of the guitar. It is an easy matter to increase this if you want to. The circuit shown uses 1/2 of a NE5532 low noise opamp (a TL072 can also be used, but with a noise penalty) - this is quite adequate for what we need here.

With the values shown, the gain is variable from unity up to a maximum of about 40dB (22 times), which should be enough for anyone. ("640k of RAM should be enough for anyone" - Bill Gates :-) ).


Complete Circuit

The complete circuit is shown in Figure 3, with a reverb mute switch and level controls. The drive control (VR1) can be a trimpot (or even fixed), since once you have determined the maximum level this will not need to be changed. There is no gain control for the guitar input, as the circuit has unity gain, so amp settings are unaffected by using the reverb.

The capacitor marked S.O.T. will need to be selected to give the sound you want. High values (above 100nF) will give quite a lot of bottom end, which tends to sound boomy and very indistinct. You will probably find that a value somewhere between 1.5nF and 10nF will sound the best - try 4.7nF as a starting point. Like the guitar amp itself, a reverb unit has its own sound, and it is only reasonable that you should be able to change it to suit your own taste.

Fig 3
Figure 3 - Complete Circuit

The power for the opamp is, Pin 4 -ve, Pin 8 +ve. Note that the opamp requires a dual supply - +/- 15V is fine, or for battery operation (not really recommended) you could get away with ±9V.

The unit could also be installed inside the amp head, and wired into the circuit. I will have to leave it to you to determine the gain needed for the various stages, since it is currently designed for "typical" guitar levels. Make sure that you provide proper isolation between the input and output of the reverb tank. I have seen circuits where this was not done, and the whole reverb circuit goes into feedback. Isolation is provided in this circuit by the virtual earth mixer (pin 2 of U1 is at 0 Volts at AC and DC).

Most reverb units use RCA sockets for input and output, and be careful with mounting. The springs will clang most alarmingly if moved about while playing, and acoustic feedback can also be a problem, especially if the low frequency gain is too high.

Fig 4
Figure 4 - Modified Version, Using P113 Headphone Amp PCB

The version shown in Figure 4 uses the drive amp configured for high output impedance. Maximum input level depends on the gain of the drive amp, which is controlled by R3L. The optimum value depends on the impedance of the reverb tank's input impedance. As shown, it is optimum for a (nominal) 160 Ohm coil. Note that the input transducer must not connect to the tank chassis. Reverb units are available with isolated inputs for just this purpose.

The recovery amp has a gain of 40 as shown (32dB), but this can be changed by using a different value for R3R (lower value, higher gain).



Additional Information

Torres Engineering - Supply and information on spring reverb tanks
Roy's "Accutron" Page - Some more useful info.

Almost all the reverb tanks that you will see are Accutronics (aka Sound Enhancements). They are made by:


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Guitar Tremolo Unit
 

Tremolo is one of those simple effects that has just lasted forever (well, almost). The circuit shown here has wide range, and a very controlled and musical modulation characteristic, and should keep the guitarists happy for minutes at a time.

The project is simple to build, and can even be housed in a pedal if desired.  If the pedal option is used, don't try to run it from batteries, as they will not last very long due to the LED current. It really needs a +/- 15V supply as shown in the circuit to operate properly.


Tremolo Unit Description

The unit is simple to build, and does not need really low noise opamps, since they only act as a modulator oscillator. I used 1458 dual types in the prototype, and they are more than good enough. The transistors can be any low noise NPN type, and they are simply buffers, ensuring a high input impedance and low output impedance.

If the unit is to built into an amplifier, it may well be possible to leave out the input transistor, since a low impedance drive circuit is probably already available from an existing opamp. It may also be possible to leave out the second transistor if a high impedance input is available at the insertion point. This is somewhat unlikely, since the most common place to have the modulator is before the tone controls.

Figure 1
Figure 1 - Tremolo Unit Circuit

The opamp power supply pins are:  Pin 4,  -ve and Pin 8, +ve.  This is the same on virtually all dual opamps. The value of C2 might need to be changed (in some cases it can be omitted) if the load impedance is less than about 20k Ohms.

The oscillator is a simple opamp feedback type, and produces a triangle wave from the capacitor (C3). This is amplified and buffered, and fed to the LED in the opto-coupler. If you are unable to obtain this device (made by Vactrol), use a high quality Light Dependent Resistor (LDR) with a LED in a light-proof encapsulation - heat-shrink tubing is good, but you will probably need two layers to ensure it is completely sealed against light getting in. Use a high output LED, and make sure that the LED and LDR are properly aligned for maximum sensitivity.

The second LED is used as a panel indicator, and can be any colour you choose. When switched off, the LEDs will both be off, and the panel LED flashes at the selected rate. It might be necessary to reduce the value of R10 to ensure that there is enough drive to the LEDs to get the full modulation.

The frequency range is from about 2.5Hz to 14Hz with the values as shown, but this can be changed to suit your needs. This is generally a good range, and will be more than wide enough for most users.

Amplitude modulation can be varied from none at all, to full modulation with the signal varying from fully on to fully off. The frequency range that can be covered with full modulation is dependent on the speed of the LDR. Most of the commonly available ones are fast enough to give a good modulation depth at even the highest frequency.


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100W Guitar Amplifier

Updated 04 Jan 2002

Introduction

Note:  This project is superseded by a new version, which has several useful additions.  PCBs are available (but only for the new amp).

Guitar amplifiers are always an interesting challenge. The tone controls, gain and overload characteristics are very individual, and the ideal combination varies from one guitarist to the next, and from one guitar to the next. There is no amp that satisfies everyone's requirements, and this offering is not expected to be an exception.

One major difference however, is that if you build it yourself, you can modify things to suit your own needs, experimentation is the key to this circuit, which is presented in basic form, with every expectation that builders will modify just about everything.

The amp is rated at 100W into a 4 Ohms load, as this is typical of a "combo" type amp with two 8 Ohm speakers in parallel. Alternatively, you can run the amp into a "quad" box (4 x 8 Ohm speakers in series parallel - see Figure 5) and will get about 60 Watts. For the really adventurous, 2 quad boxes and the amp head will provide 100W, but will be much louder than the twin. This is a common combination for guitarists, but it does make it hard for the sound guy to bring everything else up to the same level.


The Pre-Amplifier

The preamp circuit is shown in Figure 1, and has a few interesting characteristics that separate it from the "normal" - assuming that there is such a thing. This is a very basic design (this is deliberate), and is easy to build on Veroboard or similar. The gain structure is designed to provide a huge amount of gain, which is ideal for those guitarists who like to get that fully distorted "fat" sound.

However, with a couple of simple changes, the preamp can be tamed to suit just about any style of playing. Likewise, the tone controls can be modified to suit anything from an electrified violin to a bass guitar - you might even find that for anything other than bass, they have a suitable range to cover most possibilities, and even a few bassists will find that they can get the punchy sound they want, without the low-end "waffle" that many bass players dislike.

Figure 1
Figure 1 - Guitar Pre-Amplifier

Notes:
  • IC pinouts are industry standard for dual opamps - pin 4 is -ve supply, and pin 8 is +ve supply.
  • Opamp supply pins must be bypassed to earth with 100nF caps (preferably ceramic) as close as possible to the opamp itself.
  • Diodes are 1N4148, 1N914 or similar.
  • Pots should be linear for tone controls, and log for volume and master.

From Figure 1, you can see that the preamp uses a dual opamp as its only amplification. As shown, with a typical guitar input, it is possible to get a very fat overdrive sound, by winding up the volume, and then setting the master for a suitable level. The overall frequency response is deliberately limited to prevent extreme low-end waffle, and to cut the extreme highs to help reduce noise - not that it helps all that much, because with all that gain, noise is always going to be a problem.

Note:
The schematic has been modified slightly to improve the tone control performance (04 Jan 2002). A new schematic is now on line - the differences are relatively minor, but make the component values for the tone controls a bit cheaper (smaller value caps, and higher value pots). The power amp has been heavily revised, and the new version is also available.

If a really quiet amp is desired, you should substitute a 5532 dual opamp. These are more expensive (and harder to get), but will offer a substantial noise reduction. If you don't need all the gain that is available, simply increase the value of the first 4k7 resistor - for even less noise and gain, increase the second 4k7 as well.

If the bright switch is too bright (too much treble), increase the 1k resistor to tame it down again. Reduce the value to get more bite. The tone control arrangement shown will give zero output if all controls are set to minimum - this is unlikely to be a common requirement in use, but be aware of it when testing.

The diode network at the output is designed to allow the preamp to generate a "soft" clipping characteristic when the volume is turned up. Because of the diode clipping, the power amp needs to have an input sensitivity of about 750mV for full output, otherwise it will not be possible to get full power even with the Master gain control at the maximum setting.

Make sure that the input connectors are isolated from the chassis. The earth isolation components in the power supply help to prevent hum (especially when the amp is connected to other mains powered equipment).

UPDATES: I have had quite a few enquiries about the input connection setup. This is almost an industry standard, and is quite the opposite of what you might think it means. The same basic idea is used on Fender amps, as well as many others. The Hi input is used for normal (relatively low output) guitar pickups, and is "Hi" gain. Lo is 6dB less gain, and is intended for high output pickups so the first amplifier stage does not distort. The switching jack on the Hi input means that when a guitar is connected to the Lo input, it forms a voltage divider because the other input is shorted to earth. I hope this clears up any confusion (it will probably create more!).

I have also had several enquiries about the tone controls, one being that they don't do anything. If the preamp does not work properly, it is because it has been wired incorrectly - period! I know the circuit works, and it works very well, so please don't send e-mails claiming that it does not do what is claimed. For some reason, this project generates more e-mail than just about any other, and in all cases where I have had complaints, wiring errors have eventually been found.

The golden rule here is to check the wiring, then keep on checking it until you find the error, since I can assure you that if it does not work there is at least one mistake, and probably more.


Power Amplifier

The power amp is based on the 60 Watt amp previously published (Project 03), but it has increased gain to match the preamp. It has also been modified to give a bit of extra punch - not to the standard of a valve amp, but somewhat better than the average transistor amplifier. Other modifications include the short circuit protection - the two little groups of components next to the bias diodes.

Should the output be shorted, much more than the normal 7V peak will appear across the 1 Ohm resistors. This will turn on the appropriate transistor, cutting the base drive to the output stage. The effect is not particularly nice, but will save the output from instant destruction in the event of a short. Given the nature of stage work, a short circuit is something that will happen, it is only a matter of time. The circuit is designed not to operate under any normal conditions, but will limit the output current to about 8.5 Amps.

Figure 2
Figure 2 - Power Amplifier

At the input end, there is provision for an an auxiliary output, and an input. The latter is switched by the jack, so you can use the "Out" and "In" connections for an external effects unit. Alternatively, the input jack can be used to connect an external preamp to the power amp, disconnecting the preamp.

NOTE CAREFULLY There is a lot to be said for using more powerful transistors for the output stage. MJ15003/4 transistors are very high power, and will run cooler because of the TO-3 casing (lower thermal resistance). Beware of counterfeits though! There are many other high power transistors that can be used, and the amp is quite tolerant of substitutes (as long as their ratings are at least equal to the devices shown).

The speaker and line out connections allow up to two 8 Ohm speaker cabinets (giving 4 Ohms), and a line level output for connection to a direct injection (DI) box. The level is about 1.3V (or +5dBm) at full undistorted output - change the 560 Ohm resistor to modify the level if desired.

The two 1 Ohm resistors must be rated at 10 Watts (they will still get quite hot, so mount them well away from other components). These can be mounted to the heatsink with small brackets if you want to keep them a bit cooler - remember to ensure that the heatsink can handle the extra heat input, as these two will add about 10 Watts of additional heat energy. The four 0.1 Ohm resistors should be 5W types. The amp is otherwise quite conventional. Use the parallel arrangement as shown, anything less will cause the transistors to be operated outside their safe operating area, which will result in the eventual failure of the output stage.

Make sure that the two bias diodes are mounted well clear of anything that gets hot - including the heatsink. These diodes are the two in series. All diodes should be 1N4001 (or 1N400? - anything in the 1N400x range is fine). A heatsink is not needed for any of the driver transistors.

The life of a guitar amp is a hard one, and I suggest that you use the largest heatsink you can afford, since it is very common to have elevated temperatures on stage (mainly due to all the lighting), and this reduces the safety margin that normally applies for domestic equipment. The heatsink should be rated at 0.5 degree C/Watt to allow for worst case long term operation at up to 40 degrees C (this is not uncommon on stage).

Make sure that the speaker connectors are isolated from the chassis, to keep the integrity of the earth isolation components in the power supply.


Power Supply

WARNING - Do not attempt construction of the power supply if you do not know how to wire mains equipment.

The power supply is again nice and simple, and does not even use traditional regulators for the preamp. A pair of zeners is sufficient to get the voltage we need, because the current is only quite low. The power transformer should be a toroidal for best performance, but a convention tranny will do if you cannot get the toroidal.

Figure 3
Figure 3 - Power Supply

The transformer rating should be 150VA minimum - there is no maximum, but the larger sizes start to get seriously expensive. Anything over 250VA is overkill, and will provide no benefit. The slow-blow fuse is needed if a toroidal transformer is used, because these have a much higher "inrush" current at power-on than a conventional transformer. Note that the 5 Amp rating is for operation from 220 to 240 Volt mains - you will need an 8 or 10 Amp fuse here for operation at 115 Volts.

Use good quality electrolytics, since they will also be subjected to the higher than normal temperatures of stage work. The bridge rectifier should be a 35 Amp chassis mount type (mounted on the chassis with thermal compound). Use 1 Watt zener diodes, and make sure that the zener supply resistors (680 Ohm, also 1 Watt) are kept away from other components, as they will get quite warm in operation.

The earth isolation components are designed to prevent hum from interconnected equipment, and provide safety for the guitarist (did I just hear 3,000 drummers saying "Why ??"). The 10 Ohm resistor stops any earth loop problems (the major cause of hum), and the 100nF capacitor bypasses radio frequencies. The bridge rectifier should be rated at at least 5A, and is designed to conduct fault currents. Should a major fault occur (such as the transformer breaking down between primary and secondary), the internal diodes will become short circuited (due to the overload). This type of fault is extremely rare, but it is better to be prepared than not.

Another alternative is to use a pair of high current diodes in parallel (but facing in opposite directions). This will work well, but will probably cost as much (or even more) than the bridge.

Fuses should be as specified - do not be tempted to use a higher rating (e.g. aluminium foil, a nail, or anything else that is not a fuse). Don't laugh, I have seen all of the above used in desperation. The result is that far more damage is done to the equipment than should have been the case, and there is always the added risk of electrocution, fire, or both.

Electrical Safety
Once mains wiring is completed, use heatshrink tubing to ensure that all connections are insulated. Exposed mains wiring is hazardous to your health, and can reduce life expectancy to a matter of a few seconds !

Also, make sure that the mains lead is securely fastened, in a manner acceptable to local regulations. Ensure that the earth lead is longer than the active and neutral, and has some slack. This guarantees that it will be the last lead to break should the mains lead become detached from its restraint. The mains earth connection should use a separate bolt (do not use a component mounting bolt or screw), and must be very secure. Use washers, a lock washer and two nuts (the second is a locknut) to stop vibration from loosening the connection.


Speaker Boxes

The two suggested boxes are shown (in basic form only - you will need to work out the woodworking details yourself). The first (Figure 4) is a standard 2 speaker cabinet, and I strongly recommend using the open-back box, as this is the preferred option for most guitarists. Two 8 Ohm speakers are wired in parallel (giving 4 Ohms), and it is expected that with 12" speakers (300mm) this combination will be quite loud enough. Try to get speakers that are rated at at least 100W each - this safety margin is a requirement for guitar, since the amp will be overdriven for much of the time and this produces up to double the rated output of the amp.

The details of finish, handles (and the actual dimensions) of the boxes I shall leave to the builder, but I will make a few comments:

  • Tops and bottoms are shown as being inside the side panels. This does not really matter, since all corners should be reinforced with 25mm square (1") timber. All joints should be glued and screwed. Pre-drill the screw holes to prevent the end grain of the MDF from splitting.
  • Use a router if available to round off all the edges and corners, and use corner protectors.
  • Vinyl is still the most robust covering for stage gear, but carpet can be used if you prefer.
  • Use strong handles, as the boxes will be quite heavy when completed. Side "pocket" handles are best for the quad, but a strap handle can be used for the twin.
  • The baffle of the twin, and the top section of the quad are angled. This projects the sound towards the guitarist, and is better than propping the front edge on a brick or similar.
  • The baffle is shown recessed. This is to allow for a grille frame, which should fit neatly inside the recess and be fastened with Velcro or grille mounting clips.
  • Speakers should not be held in place with wood screws - use bolts, washers and nuts, or "T-nuts". Wood screws will eventually loosen, and the speakers will rattle.
For those who don't know what a tee nut is, the drawing should give you the general idea. They are readily available from specialist fastener suppliers. If you can't get hold of them, use metal thread screws with nuts and washers, and a thread locking fluid. "Nylock" nuts can also be used - they are the ones with a nylon collar inside the nut.

Generally, one thing to avoid is vented boxes - they just don't sound right for guitar. Naturally, if you like the sound of vented boxes, then go for it - guitar amps are probably one of the most personal amps in the world, and there is no right or wrong combination, as long as you get the sound you want.

Figure 4
Figure 4 - Suggested Twin Speaker Box And Wiring

The second example (Figure 5) is the classic "quad" box, and uses 4 x 8 Ohm speakers in series/parallel. This gives an impedance of 8 Ohms, so two quad boxes can be used if you really want the amp to be that loud. You might be able to get 4 Ohm speakers, in which case the series/parallel connection will give you a 4 Ohm box, so only one is needed. I suggest that the quad box also be open-backed, but this is not essential. One of the most popular guitar amps around uses closed back quads, and they sound pretty good to me.

Figure 5
Figure 5 - Suggested Quad Speaker Box And Wiring

For the speaker boxes, I recommend MDF (Medium Density Fibreboard). This is a much better material to work with than chipboard, and is also stronger. Chipboard has been used (and still is) by many manufacturers because of its one redeeming feature - it is cheap. MDF will cost quite a bit more, but the end result is worth the expense - a better finish, and a stronger box. Don't be tempted to use anything thinner than 19mm (3/4"), or the cabinet will resonate too much, and will also lack strength.

Many manufacturers use a thin (typically about 6mm) fibre board at the back of open backed cabinets to provide some protection for the drivers, and a lead storage area. Don't ! Make the rear protection panel(s) from 19mm MDF too, since this will prevent the unwanted resonances from the thin material typically used.

Speakers should also be fairly efficient if possible (> 90dB W/m), since a 3dB reduction in efficiency will result in the same SPL (Sound Pressure Level) output as an amp with half the power and 3dB more efficient speakers. Check out the local dealers for musical instrument speakers - do not use hi-fi speakers, or you will surely be disappointed - they are not designed for musical instrument applications, and usually sound awful.

Also avoid loudspeakers with aluminium dome dust caps - they sound utterly disgusting when a guitar amp is overdriven, with a hard top-end that radiates at frequencies that are discordant. Any harmonic above the seventh is discordant (out of tune), and an overdriven guitar amp is one of the few instrument combinations that can create such high harmonics. As a result, most guitar speakers are designed to roll off the top end above about 7kHz or so to avoid this problem. An aluminium dome does the opposite, and radiates wildly at the upper frequencies. This is both unpredictable and unpleasant.

Anecdote:  Some years ago, I was asked by a well known Australian guitarist if I could fly to Melbourne (from Sydney - about 1000 km) to solve this awful problem in the studio. It didn't matter how they miked the guitar amp, it still sounded terrible on the recording. It turned out that the aluminium dust cap was radiating so strongly at somewhere between 5kHz and 12kHz that it destroyed the sound, giving a most unappetising metallic edge to the music. The remedy was to carefully cut away the dust cap, and glue a piece of thin felt in its place. About an hour later (after the glue had dried), the result was that the recording engineer and guitarist alike were stunned at the difference - the sound was as smooth as silk (well, you know what I mean) and all the nastiness was gone.

Most of the established guitar amp manufacturers use speakers specially made for them by one of a few specialist loudspeaker builders, and they are normally hard to get. Try music shops (or repair shops) to see if they have speakers that might be suitable. The second-hand market might be another good place to look - you might even be able to get a complete speaker box for a reasonable price, which saves having to do the woodwork !


Effects

As shown, the amp has no effects at all, but does have an effects send and receive (via the two input jacks). Internal tremolo and reverb can be added, and suitable circuits are available on the project pages. These are designed as "stand alone" effects, but can be integrated easily, using the effects loop already provided.


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100W Guitar Amplifier Mk II
New Version Created 27 Jan 2002

           برای فیبر مدار چاپی به اینجا درخواست و مراجعه کنیدIntroduction  

Guitar amplifiers are always an interesting challenge. The tone controls, gain and overload characteristics are very individual, and the ideal combination varies from one guitarist to the next, and from one guitar to the next. There is no amp that satisfies everyone's requirements, and this offering is not expected to be an exception.

One major difference however, is that if you build it yourself, you can modify things to suit your own needs, the ability to experiment is the key to this circuit, which is although presented in complete form, there is every expectation that builders will make modifications to suit themselves.

The amp is rated at 100W into a 4 Ohms load, as this is typical of a "combo" type amp with two 8 Ohm speakers in parallel. Alternatively, you can run the amp into a "quad" box (4 x 8 Ohm speakers in series parallel - see Figure 5 in Project 27b, the original article) and will get about 60 Watts. For the really adventurous, 2 quad boxes and the amp head will provide 100W, but will be much louder than the twin. This is a common combination for guitarists, but it does make it hard for the sound guy to bring everything else up to the same level.

Note: This is a fully revised version of the original 100W guitar amp, and although there are a great many similarities, there are some substantial differences - so much so that a new version was warranted. This is (in part) because PCBs are now available for both the power and preamps. The update was sufficiently substantial to warrant retaining the original version, which is still available as Project 27b.

Typical of the comments I get regularly about the P27 power and preamp combo is this e-mail from Tony ...

I'm delighted with the P27B/27 combination. It gives me the clear, punchy, uncluttered sound I've been looking for.

I've grown tired of whistles, bells and other embellishments that some anonymous guitar amp designer somewhere is telling me I've got to have. I've now got the sound I was hoping for. Love the Twin Reverb treble boost. Takes me back to 1960!!

Without your module/boards and advice I'd have been playing about with breadboards for hours unsuccessfully searching for THE sound. Thanks.

This is just one of many, many e-mails I've received, but manages to sum up most of the comments in a couple of short sentences. This has been a popular project from the beginning, and is a solid and reliable performer that does not sacrifice sound or performance.


The Pre-Amplifier

The preamp circuit is shown in Figure 1, and has a few interesting characteristics that separate it from the "normal" - assuming that there is such a thing. This is simple but elegant design, that provides excellent tonal range. The gain structure is designed to provide a huge amount of gain, which is ideal for those guitarists who like to get that fully distorted "fat" sound.

However, with a couple of simple changes, the preamp can be tamed to suit just about any style of playing. Likewise, the tone controls as shown have sufficient range to cover almost anything from an electrified violin to a bass guitar - The response can be limited if you wish (by experimenting with the tone control capacitor values), but I suggest that you try it "as is" before making any changes. (See below for more info.)

Figure 1
Figure 1 - Guitar Pre-Amplifier

From Figure 1, you can see that the preamp uses a dual opamp as its only amplification. The lone transistor is an emitter follower, and maintains a low output impedance after the master volume control. As shown, with a typical guitar input, it is possible to get a very fat overdrive sound by winding up the volume, and then setting the master for a suitable level. The overall frequency response is deliberately limited to prevent extreme low-end waffle, and to cut the extreme highs to help reduce noise and to limit the response to the normal requirements for guitar. If you use the TL072 opamp as shown, you may find that noise is a problem - especially at high gain with lots of treble boost. I strongly suggest that you use an OPA2134 - a premium audio opamp from Texas Instruments (Burr-Brown division), you will then find this quite possibly the quietest guitar amp you have ever heard (or not heard :-). At any gain setting, there is more pickup noise from my guitar than circuit noise - and for the prototype I used carbon resistors!

opamp Notes:
1 - IC pinouts are industry standard for dual opamps - pin 4 is -ve supply, and pin 8 is +ve supply.
2 - Opamp supply pins must be bypassed to earth with 100nF caps (preferably ceramic) as close as possible to the opamp itself.
3 - Diodes are 1N4148, 1N914 or similar.
4 - Pots should be linear for tone controls, and log for volume and master.

The power supply section (bottom left corner) connects directly to the main +/-35V power amp supply. Use 1 Watt zener diodes (D5 and D6), and make sure that the zener supply resistors (R18 and R19, 680 ohm 1 Watt) are kept away from other components, as they will get quite warm in operation. Again, the preamp PCB accommodates the supply on the board.

The pin connections shown (either large dots or "port" symbols) are the pins from the PCB. Normally, all pots would be PCB types, and mounted directly to the board. For a DIY project, that would limit the layout to that imposed by the board, so all connections use wiring. It may look a bit hard, but is quite simple and looks fine when the unit is completed. Cable ties keep the wiring neat, and only a single connection to the GND point should be used (several are provided, so choose one that suits your layout. VCC is +35V from the main supply, and VEE is the -35V supply.

If you don't need all the gain that is available, simply increase the value of R6 (the first 4k7 resistor) - for even less noise and gain, increase R11 (the second 4k7) as well. For more gain, decrease R11 - I suggest a minimum of 2k2 here.

If the bright switch is too bright (too much treble), increase the 1k resistor (R5) to tame it down again. Reduce the value to get more bite. The tone control arrangement shown will give zero output if all controls are set to minimum - this is unlikely to be a common requirement in use, but be aware of it when testing.

The diode network at the output is designed to allow the preamp to generate a "soft" clipping characteristic when the volume is turned up. Because of the diode clipping, the power amp needs to have an input sensitivity of about 750mV for full output, otherwise it will not be possible to get full power even with the Master gain control at the maximum setting.

Make sure that the input connectors are isolated from the chassis. The earth isolation components in the power supply help to prevent hum (especially when the amp is connected to other mains powered equipment).

If problems are encountered with this circuit, then you have made a wiring mistake ... period. A golden rule here is to check the wiring, then keep on checking it until you find the error, since I can assure you that if it does not work properly there is at least one mistake, and probably more.

The input, effects and output connections are shown in Figure 1B.

  • Input - these are quite the opposite of what you might think. The same basic idea is used on Fender amps, as well as nearly all others that have dual inputs for a channel. The Hi input is used for normal (relatively low output) guitar pickups, and is "Hi" gain. "Lo" in this design has about 14 dB less gain, and is intended for high output pickups so the first amplifier stage does not distort. The switching jack on the Hi input means that when a guitar is connected to the Lo input, it forms a voltage divider because the other input is shorted to earth.

  • Effects - Preamp out and power amp in connections allow you to insert effects, such as compression (for really cool sustain, that keeps notes just hanging there), reverb, digital effects units, etc. The preamp out is wired so that the preamp signal can be extracted without disconnecting the power amp, so can be used as a direct feed to the mixer if desired. This is especially useful for bass. The preamp output can also be used to slave another power amplifier (as if you need even more - you do for bass, but not guitar).

  • Output - A pair of output connectors is always handy, so that you can use two speaker boxes (don't go below 4 ohms though), or one can be used for a speaker level DI box. Because of the high impedance output stage, headphones cannot (and must not!) be connected to the speaker outputs. The 'phones will be damaged at the very least, but (and much, much worse) you could easily cause instant permanent hearing loss.

Figure 1b
Figure 1B - Internal Wiring

The connections shown are very similar (ok, virtually identical :-) to those used in my prototype. Noise is extremely low, and probably could have been lower if I had made the amp a little bigger. All connectors must be fully insulated types, so there is no connection to chassis. This is very important !

You will see from the above diagram that I did not include the "loop breaker" circuit shown in the power supply diagram. For my needs, it is not required, for your needs, I shall let you decide. If you choose to use it, then the earth (chassis) connection marked * (next to the input connectors) must be left off.

A few important points ...

  • The main zero volt point is the connection between the filter caps. This is the reference for all zero volt returns, including the 0.1 ohm speaker feedback resistor. Do not connect the feedback resistor directly to the amp's GND point, or you will create distortion and possible instability.
  • The supply for the amp and preamp must be taken directly from the filter caps - the diagram above is literal - that means that you follow the path of the wiring as shown.
  • Although mentioned above, you might well ask why the pots don't mount directly to the PCB to save wiring. Simple really. Had I done it that way, you would have to use the same type pots as I designed for, and the panel layout would have to be the same too, with exactly the same spacings. I figured that this would be too limiting, so wiring it is. The wiring actually doesn't take long and is quite simple to do, so is not a problem.
  • I did not include the "Bright" switch in Figure 1B for clarity. I expect that it will cause few problems.

Bass Guitar, Electric Piano

As shown, the preamp is just as usable for bass or electric piano as for rhythm or lead guitar. A couple of changes that you may consider are ...

  • Delete the clipping diodes (unless fuzz bass/piano is something you want, of course). If these are removed, then the output should be taken directly from the Master output pin (M-OUT in Figure 1), so leave out / change the following ...
    • Delete R14, and D1-D4
    • Delete Q1 and associated components (C14, C15, R15, R16, R17)
    • Delete VR5
    • Change R13 from 4.7k to 100 ohms

You may also want to experiment with the tone control caps - I shall leave it to the builder to decide what to change, based on listening tests. C3 and C8 may be increased to 4.7uF to provide an extended bass response. If the gain is too high, simply increase R11 (10k would be a good starting point and will halve the gain).


Power Amplifier

The power amp (like the previous version) is loosely based on the 60 Watt amp previously published (Project 03), but it has increased gain to match the preamp. Other modifications include the short circuit protection - the two little groups of components next to the bias diodes (D2 and D3). This new version is not massively different from the original, but has adjustable bias, and is designed to provide a "constant current" (i.e. high impedance) output to the speakers - this is achieved using R23 and R26. Note that with this arrangement, the gain will change depending on the load impedance, with lower impedances giving lower power amp gain. This is not a problem, so may safely be ignored.

Should the output be shorted, the constant current output characteristic will provide an initial level of protection, but is not completely foolproof. The short circuit protection will limit the output current to a relatively safe level, but a sustained short will cause the output transistors to fail if the amp is driven hard. The protection is designed not to operate under normal conditions, but will limit the peak output current to about 8.5 Amps. Under these conditions, the internal fuses (or the output transistors) will probably blow if the short is not detected in time.

Figure 2
Figure 2 - Power Amplifier

Figure 2 shows the power amp PCB components - except for R26 which does not mount on the board. See Figure 1B to see where this should be physically mounted. The bias current is adjustable, and should be set for about 25mA quiescent current (more on this later). The recommendation for power transistors has been changed to higher power devices. This will give improved reliability under sustained heavy usage.

NOTE CAREFULLY As shown, the power transistors will have an easy time driving any load down to 4 ohms. If you don't use the PCB (or are happy to mount power transistors off the board), you can use TO3 transistors for the output stage. MJ15003/4 transistors are very high power, and will run cooler because of the TO-3 casing (lower thermal resistance). Beware of counterfeits though! There are many other high power transistors that can be used, and the amp is quite tolerant of substitutes (as long as their ratings are at least equal to the devices shown). The PCB can accommodate Toshiba or Motorola 150W flat-pack power transistors with relative ease - if you wanted to go that way. TIP3055/2966 or MJE3055/2955 can also be used for light or ordinary duty.

At the input end (as shown in Figure 1B), there is provision for an auxiliary output, and an input. The latter is switched by the jack, so you can use the "Out" and "In" connections for an external effects unit. Alternatively, the input jack can be used to connect an external preamp to the power amp, disconnecting the preamp.

The speaker connections allow up to two 8 Ohm speaker cabinets (giving 4 Ohms). Do not use less than 4 ohm loads on this amplifier - it is not designed for it, and will not give reliable service!

All the low value (i.e. 0.1 and 0.22 ohm) resistors must be rated at 5W. The two 0.22 ohm resistors will get quite warm, so mount them away from other components. Needless to say, I recommend using the PCB, as this has been designed for optimum performance, and the amp gives a very good account of itself. So good in fact, that it can also be used as a hi-fi amp, and it sounds excellent. If you were to use the amp for hi-fi, the bias current should be increased to 50mA. Ideally, you would use better (faster / more linear) output transistors as well, but even with those specified the amp performs very well indeed. This is largely because they are run at relatively low power, and the severe non-linearity effects one would expect with only two transistors do not occur because of the parallel output stage.

Make sure that the bias transistor is attached to one of the drivers (the PCB is laid out to make this easy to do). A small quantity of heatsink compound and a cable tie will do the job well. The diodes are there to protect the amp from catastrophic failure should the bias servo be incorrectly wired (or set for maximum current). All diodes should be 1N4001 (or 1N400? - anything in the 1N400x range is fine). A heatsink is not needed for any of the driver transistors.

The life of a guitar amp is a hard one, and I suggest that you use the largest heatsink you can afford, since it is very common to have elevated temperatures on stage (mainly due to all the lighting), and this reduces the safety margin that normally applies for domestic equipment. The heatsink should be rated at 0.5° C/Watt to allow for worst case long term operation at up to 40°C (this is not uncommon on stage).

Make sure that the speaker connectors are isolated from the chassis, to keep the integrity of the earth isolation components in the power supply, and to ensure that the high impedance output is maintained.


Power Supply

WARNING - Do not attempt construction of the power supply if you do not know how to wire mains equipment.

The power supply is again nice and simple, and does not even use traditional regulators for the preamp (details are on the preamp schematic in Figure 1). The power transformer should be a toroidal for best performance, but a convention tranny will do just fine if you cannot get the toroidal.

NOTE Do not use a higher voltage than shown - the amplifier is designed for a maximum loaded supply voltage of +/-35V, and this must not be exceeded. Normal tolerance for mains variations is +/-10%, and this is allowed for. The transformer must be rated for a nominal 25-0-25 volt output, and no more. Less is Ok if the full 100W is not needed.

Figure 3
Figure 3 - Power Supply

The transformer rating should be 150VA (3A) minimum - there is no maximum, but the larger sizes start to get seriously expensive. Anything over 250VA is overkill, and will provide no benefit. The slow-blow fuse is needed if a toroidal transformer is used, because these have a much higher "inrush" current at power-on than a conventional transformer. Note that the 2 Amp rating is for operation from 220 to 240 Volt mains and as shown is suitable for a 200VA transformer - you will need an 4 or 5 Amp fuse here for operation at 115 Volts. Smaller transformers can use a smaller fuse - I am using a 2A slow blow fuse in my prototype (160VA transformer at 240V mains input), which seems to be fine - it allows for a maximum load of 480VA which will never be achieved except under fault conditions.

Use good quality electrolytics (50V rating, preferably 105°C types), since they will also be subjected to the higher than normal temperatures of stage work. The bridge rectifier should be a 35 Amp chassis mount type (mounted on the chassis with thermal compound).

The earth isolation components are designed to prevent hum from interconnected equipment, and provide safety for the guitarist (did I just hear 3,000 drummers asking "Why ??"). The 10 Ohm resistor stops any earth loop problems (the major cause of hum), and the 100nF capacitor bypasses radio frequencies. The bridge rectifier should be rated at least 5A, and is designed to conduct fault currents. Should a major fault occur (such as the transformer breaking down between primary and secondary), the internal diodes will become short circuited (due to the overload). This type of fault is extremely rare, but it is better to be prepared than not.

Another alternative is to use a pair of high current diodes in parallel (but facing in opposite directions). This will work well, but will probably cost as much (or even more) than the bridge.

All fuses should be as specified - do not be tempted to use a higher rating (e.g. aluminium foil, a nail, or anything else that is not a fuse). Don't laugh, I have seen all of the above used in desperation. The result is that far more damage is done to the equipment than should have been the case, and there is always the added risk of electrocution, fire, or both.

Electrical Safety
Once mains wiring is completed, use heatshrink tubing to ensure that all connections are insulated. Exposed mains wiring is hazardous to your health, and can reduce life expectancy to a matter of a few seconds !

Also, make sure that the mains lead is securely fastened, in a manner acceptable to local regulations. Ensure that the earth lead is longer than the active and neutral, and has some slack. This guarantees that it will be the last lead to break should the mains lead become detached from its restraint. Better still, use an IEC mains connector and a standard IEC mains lead. These are available with integral filters, and in some cases a fuse as well. A detachable mains lead is always more convenient than a fixed type (until your "roadie" loses the lead, of course. You will never do such a thing yourself :-)

The mains earth connection should use a separate bolt (do not use a component mounting bolt or screw), and must be very secure. Use washers, a lock washer and two nuts (the second is a locknut) to stop vibration from loosening the connection.


Testing

If you do not have a dual output bench power supply
Before power is first applied, temporarily install 22 Ohm 5W wirewound "safety" resistors in place of the fuses. Do not connect the load at this time! When power is applied, check that the DC voltage at the output is less than 1V, and measure each supply rail. They may be slightly different, but both should be no less than about 20V. If widely different from the above, check all transistors for heating - if any device is hot, turn off the power immediately, then correct the mistake.

If you do have a suitable bench supply
This is much easier! Do not connect a load at this time. Slowly advance the voltage until you have about +/-20V, watching the supply current. If current suddenly starts to climb rapidly, and voltage stops increasing then something is wrong, otherwise continue with testing. (Note: as the supply voltage is increased, the output voltage will fluctuate initially, then drop to near 0V at a supply voltage of about +/-15V or so. This is normal.)

Once all is well, connect a speaker load and signal source (still with the safety resistors installed), and check that suitable noises (such as music or tone) issue forth - keep the volume low, or the amp will distort badly with the resistors still there if you try to get too much power out of it.

If the amp has passed these tests, remove the safety resistors and re-install the fuses. Disconnect the speaker load, and turn the amp back on. Verify that the DC voltage at the speaker terminal does not exceed 100mV, and perform another "heat test" on all transistors and resistors.

When you are satisfied that all is well, set the bias current. Connect a multimeter between the collectors of Q10 and Q11 - you are measuring the voltage drop across the two 0.22 ohm resistors (R20 and R21). The desired quiescent current is 25mA, so the voltage you measure across the resistors should be set to 11mV +/-2mV. The setting is not overly critical, but at lower currents, there is less dissipation in the output transistors. Current is approximately 2.2mA / mV, so 10mV (for example) will be 22mA.

After the current is set, allow the amp to warm up, and readjust the bias when the temperature stabilises. This may need to be re-checked a couple of times, as the temperature and quiescent current are slightly interdependent. When you are happy with the bias setting, you may seal the trimpot with a dab of nail polish.

NOTE Note: If R22 gets hot or burns out, the amplifier is oscillating! This is invariably because of poor layout, inadequate (or no) shielding between preamp and power amp, or use of unshielded leads for the amplifier input. Please see the photos of my completed amp to see how it should be laid out.

Please see Project 27B for the box designs and other useful info.  to see photos of the new amp
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48V Phantom Feed Supply for Microphones
Page Updated 13 June 2003


PCBs are available for this project - click the image for details

Introduction

For reasons that I find somewhat puzzling, there are very few decent 48V phantom power supply schematics available on the Net. Those that I have seen are either very crude, or require the use of a special transformer that (naturally) is all but unobtainable (or both). In contrast, a 15-0-15V transformer can be obtained almost anywhere, but alas, does not have enough voltage after rectification and filtering.

Fortunately, this is not a problem, as a voltage doubler supply will give more than enough voltage, and is easy to build. The design featured here uses just that, and allows the use of a readily available transformer and other low-cost parts, to give a supply with extremely good performance. As shown, there is no short circuit protection, but a phantom supply is unlikely to be shorted anyway, so this is not a limitation.

Of the designs that are available on the Net, there is the supply shown in the P30 mixer (see Figure 3), which is similar to the design shown here, but is not as refined.

Another version of phantom supply (reasonably common) uses an oscillator and voltage multiplier to provide the 48V (or thereabouts) supply, but these are not suitable (IMO) due to very poor regulation and an inherent inability to provide enough current. The maximum current that can be drawn from a standard phantom circuit (using 6.81k resistors) is 14mA into a short circuit. Since all phantom powered systems need some operating voltage (a typical value being around 10V), the maximum practical current drawn by each powered circuit is around 11mA. Naturally, some will draw less than this. The voltage multiplier supply will be struggling tosupply even this meagre current, and will have high battery drain as well.

There are also quite a few suggestions that you can use as little as 18V for phantom powering (using a pair of 9V batteries), but unless the microphone or DI box (for example) is specified to operate at such a low voltage, then I would not recommend it - headroom will be reduced dramatically, and distortion will be a problem at even relatively low levels. Some equipment will not work at all. The lowest recommended phantom supply voltage is 30V, and I consider even that to be too low for most things.

Output current of this design is rated at 100mA at 48V, although you will be able to get more - 200mA is not unreasonable, and even then, output ripple can be expected to be well below 1mV. Simulation gives a figure of 10uV peak to peak at 200mA, but this is likely to be rather optimistic.

It is probable that few people will ever need the maximum suggested output current, since 200mA is capable of supplying up to 20 phantom powered microphones at once.


Description

The circuit is shown in Figure 1, and as described avove, uses a voltage doubler rectifier. From there, a pair of resistors provide additional smoothing to the secondary filter caps. R3 is used to balance the voltage across C3 and C4, and must not be omitted.


Figure 1 - 48V Power Supply

The regulator was a very common topology prior to the introduction of 3-terminal regulator ICs, and is used here so that high voltage regulators are not needed. These are much harder to get than the standard versions, and still require additional circuitry because 48V versions are not made. Although the circuit looks complex, it is very easy to build (especially if the PCB is used).

The zener diode is the reference voltage, and 1/2 the output voltage is compared to the zener voltage by Q3 (the error amplifier). If output voltage increases, Q3 is turned on harder, removing base drive from Q1 (and hence Q2), reducing the output voltage to the preset value.

As can be seen, there are no adjustments, and this means that the 48V may be a little higher (or lower) than rated. This is not a problem however, and all phantom feed microphones will handle the variation without any problems at all.

Load regulation is far better than you might expect, with typically 100mV variation between full load (100mA) and no load. At 200mA load, the voltage falls by less than 150mV compared to the no-load voltage. Line (input) regulation is also quite good, with less than 200mV output change with +20% and -20% input voltage, with a load of 100mA.

The next problem is how to actually send the phantom power to the microphone and not the mixer input circuits. The latter will not be impressed with 48V DC applied, and will most likely voice their displeasure by failing instantly. The standard value of 6.81k (0.1% tolerance) for phantom feed circuits can be reduced to 6.8k (a standard E12 series resistor value), and I suggest that using a multimeter to match the resistors to at least 0.1% is the easiest and cheapest alternative. Each pair of resistors should be matched to within 10 ohms (or less if possible) of each other for best results. This is better than 0.1%, and ensures that common mode performance is not compromised.

In case you were wondering about my claim that 10 ohms is better than 0.1%, a worst case pair of 0.1% 6.81k resistors could have a difference of 13.62 ohms - one resistor at the maximum positive tolerance, and the other at maximum negative tolerance. Fairly obviously, the closer the match the better, and the multimeter used does not have to be absolutely accurate, since you are measuring for a difference rather than an absolute resistance value. If your multimeter refuses to measure to the number of digits needed, see the appendix for an alternative method you can use.

Figure 2 shows the basic phantom powering scheme. Only one channel is shown - subsequent channels are identical, up to a typical maximum of 10 (20 at a pinch) for a single supply module.

Although shown using bipolar electrolytic caps, some constructors will no doubt want to use something 'better', but polyester or similar caps at those values will be very large! Assuming a mic circuit input impedance of 1.2k (fairly typical), the two 22uF caps as shown will give a -3dB frequency of 12Hz - this is needed to get flat response to 20Hz. Naturally, if the lowest frequency you need is higher, then lower capacitance is acceptable. Likewise, if the mic preamp input impedance is higher than 1.2k, less capacitance can also be used.

It is worth noting that many mixers use polarised electrolytics at the inputs of phantom power circuits. While this is quite ok while phantom power is applied, the caps will be unbiased when phantom power is not being used. This is not a good idea (IMO), and can lead to audible distortion or colouration. For a "cost no object" design, use 10uF/50V (or higher) polyester caps, enclosed in their own shielding can. These can be wired into the PCB without too much difficulty. Keep wires tightly twisted for maximum noise rejection.


Figure 2 - Phantom Powering Circuit

Zener diodes must be used as shown to limit the maximum voltage applied to the mic input circuits. The worst possible scenario is when a mic lead is connected to a mic while phantom power is on. The cable is effectively a capacitor, and the sudden discharge of the cable and coupling caps can create a high current through the zeners, which must be capable of withstanding the surge without failure. Fortunately, 1W zeners are rugged enough to take it, and this is almost an industry standard circuit. Maximum surge current for 1W 10V zeners is typically around 450mA, and this is unlikely to be exceeded in practice. 10V zeners are specified because it is virtually impossible for any microphone to exceed that level, and the mic preamp will clip well before you achieve the 7V RMS input voltage limit imposed by the zeners. In addition, 10V zeners have a higher current capacity than higher voltages for an additional safety margin. The 10 ohm series resistors will have little or no influence on input level or noise, and help to limit the peak zener current.

Some mixers boast a "silent" phantom switch to alleviate the typical loud BANG through the mixer when phantom power is turned on or off. The phantom distribution PCB (2 channels) has this feature, but it is not shown here.


Construction

Somewhat naturally, I suggest the PCBs be used, as this makes construction very easy. The PCBs both measure only 64mm x 38mm (2.5" x 1.5"), so will be easy to retrofit into all but the most compact mixer. Where there is just no space at all, an external box can house the regulator and distribution board(s).

If you don't wish to buy the PCBs, you may use Veroboard, but unless great care is taken with the earthing arrangements, noise will almost certainly be worse than quoted. All electrolytic caps should be rated at 50V or higher (NOTE: C5 in Figure 1 must be rated at 63V!). A standard 50V ceramic is recommended for C7, which is used to ensure that the regulator does not oscillate. Not shown (or needed) are film caps in parallel with the electrolytics - if you wanted to, these may be added, but with the filtering shown high frequency noise should be non-existent.

The power transformer does not need to be anything too fancy, but I suggest a separate box for it to minimise hum and noise. Typically, a 20VA 15-0-15V (or a multitap transformer with a 30V connection) will be more than enough. These are available readily in Australia, but I can't speak for the rest of the world. If the worst comes to the worst, you can use a pair of single winding 15V transformers, with the windings in series to give 30V. I recommend a conventional 'EI' transformer if possible, as these have less capacitance between primary and secondary, and will allow less HF mains noise through.

It is extermely important that the transformer is not used to power other supplies or equipment. The centre-tap must not be connected to anything, and needs to be insulated to prevent contact. The supply circuit uses the full 30V AC in a floating configuration, and connection to another supply or rectifier will cause a short on the winding.

Q1 on the supply must be fitted with a heatsink, and worst case power dissipation will be around 5W. This may not seem like a great deal, but a 10°C/W heatsink (typical of a large PCB mount type) will get to 50°C above ambient temperature (i.e. too hot!) at 200mA output.

At 100mA load, this is reduced to about 3W, which is a little more manageable. Even so, there is no such thing as a heatsink that is too big, so I suggest that you use the largest one that you can. Forced air (fan) cooling will not be necessary.

The PCB is laid out such that the power transistor can be attached directly to the chassis if desired (using insulating washers and heatsink compound, of course), and this alleviates the need for a separate heatsink. For only one or two phantom powered mics, a heatsink is not essential, but a small one is cheap insurance. In this case, R1 and R2 in Figure 1 may be increased in value, and this will provide even better filtering. 100 ohms will be more than satisfactory for a two microphone system.


Appendix - Resistor Matching

The age old methods are sometimes the best. A Wheatstone Bridge used to be the standard method of accurately measuring resistance, inductance and capacitance, but new digital instruments have taken over. This is a shame (I think), because the old methods actually taught you something as you used them - not the case with any digital instrument.

A Wheatstone Bridge is easy to set up, and while it is not especially accurate in absolute terms, it can be made extremely sensitive to variations between components. Using a power supply, 9V battery (or even a small AC voltage from a transformer), the circuit shown in Figure 3 will resolve a difference of 1 ohm easily with a suitable test setup. A difference of 10 ohms will give a 4.38mV output signal from a 12V input.


Figure 3 - Wheatstone Bridge

Because the bridge circuit does not care if you use AC or DC, an amplifier can be used to detect the null, and it can be made almost unbelievably sensitive. In use, wire up the circuit as shown, and use a meter or amplifier and headphones (or speaker) to monitor the difference signal. Be warned - the amp input signal will be 6V RMS (for a 12V input) with the DUT (Device Under Test) removed! Install a test resistor, and adjust VR1 for a complete null (no signal). That is now your reference resistor. Connect the remaining resistors into the circuit, and aim for an output signal of less than 4mV - do not re-adjust VR1.

The signal input (applied unsurprisingly to Sig1 and Sig2) must be floating if you use an amp for monitoring, since one side of the amp input will be earthed. A 50 or 60Hz 12V transformer is fine for this. Press the TEST button only when the DUT is securely attached, or the output will be very loud indeed. You can connected a pair of diodes across the Monitor terminals - in parallel and opposite polarity. This will keep the maximum level down to something more sensible.

The Wheatstone Bridge is such a useful device that you can soon expect to see a complete article describing its use for comparing resistors, capacitors and inductors - actually, you can match any passive components and even complete networks by this means. Wheatstone bridges are also used for precision temperature measurement, strain gauges (use to test mechanical movement in structures) and has many other uses.


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Universal Preamp / Mixer - Part 2


     PCBs are available for this project - Click the PCB image for details

Introduction

The standard version of the P94 has been quite popular over the years, and is a very versatile unit. Based on a number of queries since it was first published, this article shows just how versatile it is. See Project 94 for the original article, which has further circuit descriptions, tone control frequency response curves, etc.

The PCB has two sections - an input stage and a virtual earth mixer. In this version, the roles of the two are simply reversed, with the input stage now forming a master tone control stage, and the mixer is a multi-channel input.


Description

The circuit is very simple, and the standard P94 PCB is used. A single board can be used to mix from 2 to 10 inputs (only four are shown), and the switches let you switch off any channel that's not being used. This keeps noise to a minimum. Because a virtual earth mixer has a noise gain that is determined by the number of inputs, using switches to disable unused inputs keeps the noise as low as possible.

The circuit does function as a true mixer, with the ability to mix as many of the input sources as needed. There is no interaction between the individual level pots, so adjusting one will have no effect on the level of any of the other inputs.

The input stage (actually the second section on the PCB) has a maximum gain of 2 with the level control at maximum. This can be increased or decreased by changing the value of R115/215. A higher resistance gives higher gain and vice versa.

Figure 1
Figure 1 - Input Level, Switching and Mixer

The only change from the original schematic is the value of R115/215, and replacement of R111/211 with a link. The individual channel inputs are wired externally, which is repetitive but simple. Use of a PCB for this is ill advised because it would restrict the layout to that of the board.

The output of the mixer stage goes to a master volume control, which controls all channels simultaneously, while maintaining the relationship between mixed sources. A balance pot can be added if desired (see Project 01 for some ideas on this). Note that I have specified log pots (audio taper) for all inputs and the master for simplicity, and to maintain a respectable input impedance. As shown, the worst case input impedance of each channel input is about 20k.

Figure 2
Figure 2 - Master Volume and Tone Controls

The master section is unchanged from the original P94 article. The input stage (based around U1) has a gain of 2 (6dB), allowing a total maximum gain of 4 times (12dB) with all controls at maximum. This can be increased in both the input mixer and the master section. To increase master gain, decrease R104/204. The gain of this stage is determined by ...

Av = ( R105 / R104 ) + 1     For example, using 10k and 4.7k respectively
Av = ( 10 / 4.7 ) + 1 = 2.13 + 1 = 3.13 = 10dB (close enough)

Note that because of the reversal of the circuit (the input stage is now the output stage), you must include external 100 ohm series resistors as shown at each output (Mix A and Mix B). If these are omitted, the opamp will oscillate if a shielded lead is attached (standard interconnect leads are always shielded).

Only the Left channel is shown. The right channel is identical, with resistor and capacitor designations as indicated on the circuit diagram.

In all other respects, refer to the Project 94 article. This shows tone control response and construction / test information, plus other data you will find useful to understand the circuit, and how it works. The reversal of the stages is easily followed because all PCB termination points have been shown as they appear.


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Universal Preamp / Mixer


     PCBs are available for this project - Click the PCB image for details

Introduction

I have had a great many enquiries about small mixers, and this project should suit the needs of anyone who needs a very basic mixing unit. It has an input buffer, tone controls, and a 4 input mixing amplifier. I have called it "universal" since the same PCB can be used for many applications requiring basic amplifier modules. It can be used as a hi-fi preamp, mixer, general purpose amplifier/ tone control building block, or you may find other applications just waiting for something like this.

The list of configurations possible is so broad as to make it difficult to cover them all. The secure site shows several configurations for the PCB, from the basic functionality described, right through to using it as a balanced line driver.

The configurations are extensive despite the simplicity, since various other projects can be used as the front end. For example, using 4 of the P94 PCBs would allow you to have two stereo line inputs (direct via pots), a stereo phono input (using P06) and a stereo mic input (with a P66 board), each with its own level and tone controls. Please see Project 94A for an alternate version that will suit many applications.

A stereo master volume control then lets you set the overall level, and the individual channel levels are set using their respective level controls.


Description

The circuit is very simple, and the PCB is nice and small (approx 50 x 75 mm). The idea is that one PCB would be wired with all components (Figure 1 and Figure 2), while the others only use the section shown in Figure 1. You can select the inputs you need, and add the appropriate input circuits, such as phono preamps, mic preamps, etc. Indeed, the range of uses is determined more by imagination than any "limitations" in the circuitry itself.

The first stage (U1) is a buffer, but provides a gain of 2 (6dB) as shown. The gain is easily changed by changing the value of R104 (and R204 in the "B" Channel) - a higher value gives less gain, and vice versa. I don't recommend that the gain be increased beyond about 4 times (12dB), or DC offset will become a problem. A value of 2k7 (2.7k) for R104/204 will give a stage gain of 3.7 (11.4dB) which should be more than enough. A microphone preamp is a must if very low level signals are intended.

Figure 1
Figure 1 - Input and Tone Controls

The second stage is a standard Baxandall feedback tone control, and will give an almost dead flat frequency response with the controls in the centre position. For stereo, use dual pots all round, but for mono, single pots will be needed. The tone control response curves are shown in Figure 3. The small markings on the pots (e.g. B1, B2 and B3) are references to the PCB connections

Figure 2
Figure 2 - Mixer Amplifier

The mixer is the common "virtual earth" mixing amplifier, and there is nothing special about it. Note that it is inverting, which complements the tone controls (also inverting) so the absolute signal polarity is maintained. As shown, the mixer also has a gain of a little over two times, and again this can easily be changed. Making R115/215 10k sets the gain at -1 (i.e. unity, but inverted). Note that R117/217 are not mounted on the board, but mount directly on the output level control.

Worst case output impedance is a little under 10k, so this unit is not suitable for driving long signal leads. VR104/204 can be reduced in value if you want, but if good quality low capacitance leads are used, I doubt that you'll have any problems.

All potentiometers are linear taper. The resistor values are selected to give a log "law" as described in Project 01 (where needed).

Figure 3
Figure 3 - Tone Control Response

Figure 3 shows the frequency response with the controls at 10% intervals. The centre frequency is deliberately set lower than the "industry standard" 1kHz, which (IMO) is an extraordinarily non-sensible place to set the bass turnover frequency. You will notice that there is a small "flat" section, between 500Hz and a little under 1kHz. Bass response may be changed by using a different value for C103/203 (higher value, lower frequency), and likewise C104/204 control the high frequency point (lower value, higher frequency). I expect that most users will find the values to their liking as shown, but it can be changed quite easily.

Photo
Photo of Completed PCB (No Wiring Shown)

The photo of the PCB shows the standard preamp connection, and you can see that the remaining mixing resistors have been omitted. This was done for testing (to make sure there were no errors on the PCB), and is a perfectly valid option for normal use.


Construction

If the ESP board is used, construction is very easy. It is small, but laid out very logically so it is easy to construct. No pots are mounted on the PCB - not because I like running wires (and I don't expect you do either) but because this gives you far greater flexibility for your version of the project. If I designed the board with the pots, then you would have to use the same type as I designed for, and the same spacings and layout. This is very restricting - especially if you can't get (or don't want to use) the same type of pot.

The power supply may be from +/-9V (for portable use), or +/-15V for use with the P05 power supply. Any dual supply may be used, so if you have one already, it may be used as long as the voltage is between +/-9V and +/-15V. Higher or lower voltages are not recommended.

I have shown the circuit with TL072 opamps, but you may use anything you like (must be an industry dual opamp though).

Opamp
The standard pinout for a dual opamp is shown on the left. If the opamps are installed backwards, they will almost certainly fail, so be careful.

The suggested TL072 opamps will be quite satisfactory for most work, but if you prefer to use ultra low noise or wide bandwidth devices, that choice is yours.

Remember that the supply earth (ground) must be connected! When powering up for the first time, use 100 ohm or 560 ohm "safety" resistors in series with each supply to limit the current - this will prevent (most) damage if you have made a mistake in the wiring.


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LED Audio VU Meter


Introduction
Meter

Thanks to Uwe
Beis for the
meter display

It is quite true that there are many variations of this circuit already on the WNet, but for the sake of completeness (and because I will (eventually) be producing PCBs for this version) here is yet another.

The LED meter is simpler and smaller than it's analogue counterpart, and is very common in audio equipment. This version is based on a National Semiconductor IC, and uses the logarithmic version. Each LED operates with a 3dB difference from the previous one, and a jumper is provided to allow dot or bar mode.

This project is also an essential part of the expandable analyser to be published soon (or perhaps "eventually"), and one meter circuit is used for each frequency band. There are many other uses for a simple LED VU meter. They are ideal as power meters on amplifiers, can be used with mixers (including the high quality mixer described in the project pages), preamps and any other application where it is important to know the signal level.


Description

The circuit is completely conventional, and is based on the application notes from National Semiconductor. The circuit is shown in Figure 1 and as you can see it uses a single IC and a few discrete components. The extra diode (D3) is included to ensure that the DC to the LEDs is almost unfiltered. C1 is included to make sure the IC does not oscillate, and is not a filter cap. This allows a higher LED current with lower dissipation than would be the case if the DC were fully smoothed, and full smoothing would also require a much larger capacitor. This increases the size and cost of the project - especially important if it is to be used for the expandable analyser, since there will be at least 10 meter circuits needed.

Figure 1
Figure 1 - The LED VU Meter Circuit

L1 to L8 will normally be green (normal operating range) and L9 and L10 should be red (indicating overload). This gives a 6dB overload margin when the unit is calibrated as described below. As shown, full scale sensitivity (with VR1 at maximum) is 12 Volts peak (approximately 8.5 volts RMS). This is designed for direct connection to the speaker output of an amplifier, but is still suitable for use with preamps if the sensitivity is changed.

JP1 determines dot or bar mode. With the jumper installed, the unit operates in bar mode, meaning that LEDs will light in a continuous bar. If the jumper is omitted, then only the LED corresponding to the current signal level will light. Dot mode uses far less current, but the display is not as visible.

Power comes from a 15-0-15 transformer (connected to AC1-Com-AC2). You can generally use the smallest one available, as average power is quite low. The peak current is about 120mA DC, so a 5VA transformer will be sufficient to power two meter circuits. One 15V winding goes to the terminal AC1, the other goes to AC2 and the centre tap is connected to Com (Common). The 10 ohm resistor isolates the earth connection to help prevent hum if the same transformer is used to power a preamp (for example).

The formula for sensitivity is somewhat complex, and is further complicated by the fact that the same resistors that change the reference voltage also affect the LED current. As shown, LED current is about 12mA. To save you the (very) tedious calculations, I have prepared a table to use to set the reference voltage (the reference voltage sets the signal level for the "all LEDs on" condition). This always needs to be slightly lower than the voltage to be measured, so that fine adjustments can be made with VR1. LED current is fixed at about 10-13mA for all voltages.

Ref. Voltage R3 (k) R4 (k) I led (mA)
12 (11.6) 2.2 15 12
10 (9.99) 2.7 15 10.2
8 (8.13) 2.2 10 10.4
6 (5.81) 1.8 5.6 10.5
4 (3.81) 1.2 2.2 12.9
2 (2.20) 1.2 0.82 11.9
Table 1 - Resistor Values For Different Voltages

Now, if the above looks too irksome, or fails to meet your needs, you can download a little calculator that will do exactly what you want, and can even check what values you will get from "real world" resistor values. Click here to download LM3915.zip (12,583 bytes), and extract the files into the directory (folder) of your choice. (Note, the program needs the Visual Basic 4 runtime libraries.)

The circuit only senses the positive signal (i.e. it is half-wave only). In most cases this is not a problem, because although audio waveforms are asymmetrical, the overall signal usually balances out over a period of time. If this is not desirable, a simple rectifier circuit using a dual opamp (a cheap one is quite OK) is shown in Figure 2, and can be added between the signal source and the input. This is not a "precision" rectifier, and as such will introduce a small error into the signal, causing the sensitivity of low level signals to be reduced. The lowest couple of LEDs will therefore not be exactly 3dB apart, but for monitoring purposes this error can be completely ignored.

If this is to be used, substitute a fixed 100k resistor for VR1 (from Pin 5 to ground) in Figure 1, and bring the signal into the IC via R1 as shown by the dashed line. VR1 in the signal rectifier will be used to change the gain rather than the meter circuit. R3 and R4 should use the values shown in Figure 1 for best accuracy.

Figure 2
Figure 2 - Simple Full Wave Rectifier and Preamp

The signal rectifier needs a +/- supply of 15 volts, and the audio signal is fed directly into the "Aud" input of the meter circuit. I suggest that the signal level to the rectifier be reasonably high (or use the "Set Gain" control to increase the gain of the first stage). This will minimise the errors from the "less than perfect" rectifier. The reason for not using a precision rectifier circuit is simply one of cost - there are more components, the opamp needs to be better than the 1458 specified, and the result is just not worth the extra effort. The speed of the circuit can be adjusted by varying the value of C3. With a high value (say 10uF), the meter will act more like a peak programme meter, holding the highest peaks for a relatively long time. The lower the value, the more quickly the meter will respond.

Please Note

Note - the input to this circuit must be less than 10V RMS at all times. Higher levels will be clamped by the protection diodes (D1, D2), but these cannot be relied upon for continuous protection against high level input signals. Excessive levels will destroy the opamp's input circuit. For higher voltage an input attenuator must be used, and an external limiting resistor (10k) in series with the input is recommended..

The gain of this circuit (as shown) is limited to a maximum of 11. At higher gain values, cheap opamps (such as the 1458) will be unable to amplify the highest frequencies due to their bandwidth limitations. This means that the lowest level signal you can have for a full scale reading will be about 1.3V peak, or about 900mV RMS. The maximum gain I would recommend is obtained using a 4.7k resistor for R3. This will give a gain of about 22, at which point the response will barely make it to 20kHz. This equates to a maximum signal sensitivity of a little under 400mV RMS. It is unlikely that this will ever be needed in practice, as it is far too low to operate any preamp or mixer and retain respectable noise performance.


Calibration

You have (of course) selected the resistors R3 and R4 to give a reference voltage slightly lower than the peak voltage to be measured. Now the meter can be calibrated to suit your application.

This could not be simpler. At the maximum level you wish to operate the equipment (as shown on an audio millivoltmeter or oscilloscope with signal applied), adjust VR1 so that the signal illuminates all the green LEDs (L1 is the most sensitive, and L10 indicates maximum level, so L1 to L8 should be lit). If the input is directly from a speaker output, an additional series resistor should be used at the "Aud" input terminal to reduce the level. This can be determined by calculation (I leave this to you) or by experiment. As a guide, for a 50W amplifier, the external resistance should be about 47k ohms.

If you are using the external signal rectifier, VR1 should have been omitted from the circuit as described above. Apply the signal voltage to the input of the signal rectifier at the maximum permitted level. Adjust VR1 (on the rectifier) to illuminate LEDs L1 to L8.

If you are calibrating the meter for a power amplifier, set the output to a level just below clipping. Adjust the level control until all LEDs are illuminated. This way, if the last LED (L10) lights when you are listening to music, you will know that you are very close to clipping, and the volume should be reduced.


  MeterMeterMeterMeterMeterMeterMeterMeter

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LED Audio VU Meter


Introduction
Meter

Thanks to Uwe
Beis for the
meter display

It is quite true that there are many variations of this circuit already on the WNet, but for the sake of completeness (and because I will (eventually) be producing PCBs for this version) here is yet another.

The LED meter is simpler and smaller than it's analogue counterpart, and is very common in audio equipment. This version is based on a National Semiconductor IC, and uses the logarithmic version. Each LED operates with a 3dB difference from the previous one, and a jumper is provided to allow dot or bar mode.

This project is also an essential part of the expandable analyser to be published soon (or perhaps "eventually"), and one meter circuit is used for each frequency band. There are many other uses for a simple LED VU meter. They are ideal as power meters on amplifiers, can be used with mixers (including the high quality mixer described in the project pages), preamps and any other application where it is important to know the signal level.


Description

The circuit is completely conventional, and is based on the application notes from National Semiconductor. The circuit is shown in Figure 1 and as you can see it uses a single IC and a few discrete components. The extra diode (D3) is included to ensure that the DC to the LEDs is almost unfiltered. C1 is included to make sure the IC does not oscillate, and is not a filter cap. This allows a higher LED current with lower dissipation than would be the case if the DC were fully smoothed, and full smoothing would also require a much larger capacitor. This increases the size and cost of the project - especially important if it is to be used for the expandable analyser, since there will be at least 10 meter circuits needed.

Figure 1
Figure 1 - The LED VU Meter Circuit

L1 to L8 will normally be green (normal operating range) and L9 and L10 should be red (indicating overload). This gives a 6dB overload margin when the unit is calibrated as described below. As shown, full scale sensitivity (with VR1 at maximum) is 12 Volts peak (approximately 8.5 volts RMS). This is designed for direct connection to the speaker output of an amplifier, but is still suitable for use with preamps if the sensitivity is changed.

JP1 determines dot or bar mode. With the jumper installed, the unit operates in bar mode, meaning that LEDs will light in a continuous bar. If the jumper is omitted, then only the LED corresponding to the current signal level will light. Dot mode uses far less current, but the display is not as visible.

Power comes from a 15-0-15 transformer (connected to AC1-Com-AC2). You can generally use the smallest one available, as average power is quite low. The peak current is about 120mA DC, so a 5VA transformer will be sufficient to power two meter circuits. One 15V winding goes to the terminal AC1, the other goes to AC2 and the centre tap is connected to Com (Common). The 10 ohm resistor isolates the earth connection to help prevent hum if the same transformer is used to power a preamp (for example).

The formula for sensitivity is somewhat complex, and is further complicated by the fact that the same resistors that change the reference voltage also affect the LED current. As shown, LED current is about 12mA. To save you the (very) tedious calculations, I have prepared a table to use to set the reference voltage (the reference voltage sets the signal level for the "all LEDs on" condition). This always needs to be slightly lower than the voltage to be measured, so that fine adjustments can be made with VR1. LED current is fixed at about 10-13mA for all voltages.

Ref. Voltage R3 (k) R4 (k) I led (mA)
12 (11.6) 2.2 15 12
10 (9.99) 2.7 15 10.2
8 (8.13) 2.2 10 10.4
6 (5.81) 1.8 5.6 10.5
4 (3.81) 1.2 2.2 12.9
2 (2.20) 1.2 0.82 11.9
Table 1 - Resistor Values For Different Voltages

Now, if the above looks too irksome, or fails to meet your needs, you can download a little calculator that will do exactly what you want, and can even check what values you will get from "real world" resistor values. Click here to download LM3915.zip (12,583 bytes), and extract the files into the directory (folder) of your choice. (Note, the program needs the Visual Basic 4 runtime libraries.)

The circuit only senses the positive signal (i.e. it is half-wave only). In most cases this is not a problem, because although audio waveforms are asymmetrical, the overall signal usually balances out over a period of time. If this is not desirable, a simple rectifier circuit using a dual opamp (a cheap one is quite OK) is shown in Figure 2, and can be added between the signal source and the input. This is not a "precision" rectifier, and as such will introduce a small error into the signal, causing the sensitivity of low level signals to be reduced. The lowest couple of LEDs will therefore not be exactly 3dB apart, but for monitoring purposes this error can be completely ignored.

If this is to be used, substitute a fixed 100k resistor for VR1 (from Pin 5 to ground) in Figure 1, and bring the signal into the IC via R1 as shown by the dashed line. VR1 in the signal rectifier will be used to change the gain rather than the meter circuit. R3 and R4 should use the values shown in Figure 1 for best accuracy.

Figure 2
Figure 2 - Simple Full Wave Rectifier and Preamp

The signal rectifier needs a +/- supply of 15 volts, and the audio signal is fed directly into the "Aud" input of the meter circuit. I suggest that the signal level to the rectifier be reasonably high (or use the "Set Gain" control to increase the gain of the first stage). This will minimise the errors from the "less than perfect" rectifier. The reason for not using a precision rectifier circuit is simply one of cost - there are more components, the opamp needs to be better than the 1458 specified, and the result is just not worth the extra effort. The speed of the circuit can be adjusted by varying the value of C3. With a high value (say 10uF), the meter will act more like a peak programme meter, holding the highest peaks for a relatively long time. The lower the value, the more quickly the meter will respond.

Please Note

Note - the input to this circuit must be less than 10V RMS at all times. Higher levels will be clamped by the protection diodes (D1, D2), but these cannot be relied upon for continuous protection against high level input signals. Excessive levels will destroy the opamp's input circuit. For higher voltage an input attenuator must be used, and an external limiting resistor (10k) in series with the input is recommended..

The gain of this circuit (as shown) is limited to a maximum of 11. At higher gain values, cheap opamps (such as the 1458) will be unable to amplify the highest frequencies due to their bandwidth limitations. This means that the lowest level signal you can have for a full scale reading will be about 1.3V peak, or about 900mV RMS. The maximum gain I would recommend is obtained using a 4.7k resistor for R3. This will give a gain of about 22, at which point the response will barely make it to 20kHz. This equates to a maximum signal sensitivity of a little under 400mV RMS. It is unlikely that this will ever be needed in practice, as it is far too low to operate any preamp or mixer and retain respectable noise performance.


Calibration

You have (of course) selected the resistors R3 and R4 to give a reference voltage slightly lower than the peak voltage to be measured. Now the meter can be calibrated to suit your application.

This could not be simpler. At the maximum level you wish to operate the equipment (as shown on an audio millivoltmeter or oscilloscope with signal applied), adjust VR1 so that the signal illuminates all the green LEDs (L1 is the most sensitive, and L10 indicates maximum level, so L1 to L8 should be lit). If the input is directly from a speaker output, an additional series resistor should be used at the "Aud" input terminal to reduce the level. This can be determined by calculation (I leave this to you) or by experiment. As a guide, for a 50W amplifier, the external resistance should be about 47k ohms.

If you are using the external signal rectifier, VR1 should have been omitted from the circuit as described above. Apply the signal voltage to the input of the signal rectifier at the maximum permitted level. Adjust VR1 (on the rectifier) to illuminate LEDs L1 to L8.

If you are calibrating the meter for a power amplifier, set the output to a level just below clipping. Adjust the level control until all LEDs are illuminated. This way, if the last LED (L10) lights when you are listening to music, you will know that you are very close to clipping, and the volume should be reduced.


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VU And PPM Audio Metering


Introduction

VU (Volume Unit) meters used to be the mainstay of audio metering systems, but they have been replaced by LED metering in a great many mixers and other applications. Even in software, the most common level meter is made to look like an LED meter. The Peak Programme Meter (PPM) was originally developed by the BBC to overcome the shortcomings of the VU meter, which is notoriously bad at showing the peak signal level. The VU meter is average reading, and the ballistics are important if an accurate reading is to be obtained.

Ideally, a VU meter is supposed to take 300ms to stabilise, and should show only minor overshoot. Very few so-called VU meters come even close to the specification, and the little units on tape machines and sometimes provided on power amplifiers generally bear no resemblance to a real VU meter except that the meter dial is divided into the proper number of divisions, and has a red section from 0VU to +3VU. Oh yes, it will also say 'VU' on the meter face as well.

PPMs are less common, although quite a few systems use LED arrays that are (more or less) PPMs. Some show both VU and Peak Programme on the same LED array, with one LED seeming to 'stick' at a higher level indicating the peak.


Description

The unit described here makes no pretence at being a real VU meter, and although it can also be used as a PPM, it does not meet the original BBC standards, which call for a linear meter and a logarithmic amplifier, with highly specified ballistics.

The term 'ballistics' refers to the absolute movement of the meter's pointer, and for true VU and PPMs there are detailed specifications for the movement of the meter needle in response to a signal ...

PPM:  A standard PPM has a 5ms integration time, so that only peaks wide enough to be audible are displayed. This translates into a response that is 1dB down from a steady state reading for a 10ms tone burst, 2dB down for a 5ms burst, and 4dB down for a 3 ms burst. These requirements are satisfied by an attack time constant of 1.7ms. The decay rate of 1.5 seconds to a -20dB level (IEC specified) is met using a 650 ms time constant.

VU:  A VU meter is designed to have a relatively slow response. It is driven from a full-wave averaging circuit defined to reach 99% full-scale deflection in 300ms and overshoot not less than 1% and not more than 1.5%. Since a VU meter is optimised for perceived loudness it is not a good indicator of peak performance.
Although these specifications are available as shown above, the meter movement itself will rarely behave itself well enough to meet the specs without a direct-coupled amplifier to control the meter's mechanical components. This would needlessly complicate the project, which needs to be able to indicate average and peak power or signal levels, but for the purposes of this exercise does not need the absolute accuracy of the real thing. Since properly damped meters are rare and expensive, I have chosen to use a standard readily available meter movement. This will be quite satisfactory for the intended purpose.

The amplifier / rectifier is a simple LM1458 or similar dual opamp, and buffers the rectifier circuit. Active rectification is needed so the diode voltage drop does not cause huge inaccuracies, but by amplifying the signal first, we can use a simple rectifier and reduce the overall component count. The diodes must be germanium types as specified, or the low levels will suffer significant deviation from the ideal. This is not intended as a precision instrument, but will be much better than the units that are available (OK; most of the units that are available - if you really want to spend $400 or more for a single meter, then it will be better than this - but by how much?).

Figure 1 shows the typical internal circuit of simple (cheap) VU meters. A single diode is used in some, but the better ones will generally use a tiny selenium bridge rectifier or a germanium diode bridge. Although a capacitor is shown, few budget VU meters will include it. As a result, the meter movement itself is uncontrolled in most of these meters, so overshoot is often huge, and the reading is almost useless. Because of the diode forward voltage, many of these meters also fail completely to register low level signals (< -20 dB).

In contrast, Figure 2 shows the meter's ballistic control for this project, which involves a low value resistor in parallel to damp the movement, and a capacitor to provide some additional damping and better averaging when in VU mode.

Figure 1
Figure 1 - Simple VU Meter Circuit

There are two different time constants. One gives the averaging needed for VU metering and is present at all times. The other is switched in by SW1, and the 100 ohm resistor (R5) ensures that the rectifier will charge the capacitor in 10ms, with a 1 second decay time. This is used for PPM mode, and allows you to see at a glance what the peak level is. The difference between VU and PPM readings can vary greatly, but will typically be between 10 and 15dB, depending on the signal source.

The rectifier circuit is as simple as I could make it, but will still have quite good performance. The diode voltage drop of a germanium device is only about 200mV (as opposed to 650mV for a silicon diode), so the low level inaccuracy is minimised. 'Proper' full wave rectification could have been used with the standard opamp precision rectifier, but this would increase component count needlessly. The circuit shown is a reasonable compromise.

Figure 2
Figure 2 - Complete VU / PPM Circuit

The values of R6, C1, C2 and C3 may need to be adjusted, depending upon the ballistics of the meter movement you use. Because meters vary so widely in this respect, it is only possible to provide representative values, although they should work quite well in practice. In order to get exact VU meter ballistics, it will be necessary to test the meter with a 300ms burst waveform at full scale (+3VU). It should reach 99% of full scale with up to 1% of overshoot before dropping back to zero.

By definition, a moving coil meter movement responds to the average value of applied current. If the movement is seriously underdamped, the (moving) average will still cause excessive pointer activity - C3 is designed to help damp this. Should you be fortunate enough to have a well damped meter movement, it may be possible to omit C3.

Opamp The standard pinouts for a dual opamp are shown (top view of device). It is suggested that a bypass capacitor (typically a 100nF ceramic or polyester) be connected between pins 4 and 8, as close to the opamp as possible. Although the opamp specified is relatively slow (by comparison to 'premium' devices), it is still a good idea to use a bypass cap to prevent possible instability at high frequencies.

The circuit has 2 inputs shown. The signal input has an impedance of 10k, and may be adjusted for full scale with a signal of about 500mV. The speaker input is optional (just leave it out if it's not needed), and has a maximum sensitivity of 5V. It can be adjusted to allow for any higher voltage to suit nearly all power amps. If both inputs are not required, just leave the unwanted one out of the circuit.

The first opamp is an inverter (with gain control to allow calibration), and the second stage is also an inverting buffer with a gain of -1. D1 therefore rectifies the negative half-cycles, and D2 rectifies the positive. This gives full wave rectification so that both positive and negative peaks are measured. This is important, because audio signals can be very asymmetrical, which causes significant error in the indicated level. The lowest levels are unimportant in this application, so the diode induced inaccuracy is of no consequence, but for lowest error, use the suggested OA91 (or similar) germanium diodes. LM1458 dual opamps have been specified, and these will be more than adequate for this meter. Better opamps may be used, but there will be little or no improvement in performance.

We must ensure that the capacitor can be charged quickly for use as a PPM using the opamp output current, which is typically only about 20mA, so a reasonably small capacitance is needed. Even so, with the meter movement loading of about 3500 ohms, the decay time will be much too fast. Increasing the capacitance will make it that much harder to charge the cap quickly enough, so Q1 acts as a buffer. D3 and the associated resistor provide a forward bias so the transistor will be conducting at the first sign of any signal. The negative terminal of the meter circuit is held at -0.65V, so the transistor already has the needed 0.65V emitter to base bias. D1 and D2 will prevent the meter from registering any significant deflection with no AC signal present. Any small amount that is visible can be corrected with the meter's mechanical offset adjustment.

Figure 3 - Click to Enlarge
Figure 3 - Meter Scale (Example Only)

A sample meter scale is shown in Figure 3, and this can be used as a template, or you can re-size the image and print it onto suitable material and stick it to the meter face. If you are really lucky, you might even be able to obtain a 50uA meter movement already calibrated in VU, but don't count on it.

A 50uA meter movement is about the ideal, and these are very common. Like all analogue movements they are not cheap, and a unit of reasonable quality will be in the order of $20 or so. These will typically have a DC resistance of about 3500 Ohms. If the one you get is markedly different, you may need to adjust the 4.7k and 220 Ohm resistors (R6, R7 and R8) so that full scale is achieved in VU mode with an input of 5V into the speaker input. The calibration control must be at maximum resistance for this test.

The output voltage from the rectifier for full scale is designed to be about 5V, and the 4k7 and 220 ohm resistors around the meter provide the attenuation needed and give excellent electrical damping. The capacitor in parallel with the meter also helps to damp the movement, preventing (at least to some degree) overshoot and undershoot. The exact value might need to be changed to suit the movement you have, since as noted above, their ballistics are somewhat variable. This circuit is much faster than the standard 300ms (or 150ms as used by some manufacturers). I happen to think that this is too slow to be useful, and I think that you will be happy with the final result. C2 (47nF) is optional. When this is in place the meter will show a slightly higher than normal reading for a given signal, and this will give a more meaningful VU reading in most cases.


   MeterMeterMeterMeterMeterMeterMeterMeterMeterMeterMeterMeterMeterMeterMeterMeter

                                                                                                     

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Microphone Circuit Test Oscillator


Introduction

So, a reader sends me an e-mail, and says ....

Say, I am looking for a tone generator schematic.  Specifically, one that is mounted on an "XLR" style plug, to test mic lines.  These are great when installing multiple mic lines, to sort out which one is which; you can test the lines with only one person, too!  Do you have one?  It has got to be cheaper than buying one, and they can't be that difficult to build, can they?  Thanks!
Well, it turns out I don't have one, but I could see the circuit in my head as I wrote the reply.  Using my trusty opamp test board (see Project 41), I whipped it up in about 10 minutes.  And here it is ....


Description

This unit would be mounted in a small plastic or preferably metal box, with a 9V battery, level control, a male XLR connector (same as on a mic) and a switch.  Current drain is low, since the circuit only uses one dual opamp.  There is no need for a high quality device, and a 1458 is all that is needed.

Figure 1
Figure 1 - Mic Circuit Test Oscillator

The first stage is the oscillator itself.  This is a simple three stage phase shift oscillator - a circuit that is remarkably uncommon - which is to say I have never seen it used elsewhere.  I designed it for another project a few years ago, and I don't understand why it is not in any opamp application notes.  Maybe I invented a new circuit :-D

If you want to tune it, you can use a 50k pot instead of R1.  I suggest that if tuned, set it to A-440 Hz.  Frequency stability is not wonderful, and it changes by a few Hertz as the battery discharges, but this is unlikely to cause problems - it is a test oscillator, not a tuning standard.  As shown, frequency will be about 430Hz, depending on the accuracy of the capacitors.

The phase shift network (R1-C1, R2-C2 and R3-C3) serves two purposes.  First (and for an oscillator, most importantly), it shifts the phase of the output signal so the feedback is positive, causing oscillation.  Secondly, since it is a three stage filter, it attenuates the signal and filters the output square wave so the signal at pin 2 is a reasonable sine wave.  Distortion (if you really care) is about 3% or so - I didn't measure it this time, but I recall having done so before.

The second stage is the output buffer, and the signal is simply split to supply the two mic leads.  The metal case should be connected to pin 1 (earth) on the XLR connector.  The output level control must be a linear type, as the circuit loading will create a good approximation to a log pot.  Maximum output into a typical microphone input will be about 100mV (unloaded oscillator output on mine was 140mV).

Not much to it - the whole circuit can be built on a small piece of veroboard, and the battery, pot and XLR connector will take up far more room than the oscillator.  There is no LED indicator for power, as this would draw more current than the circuit.  To prevent accidentally turning it on, a slide switch is suggested.  They are a pig to mount compared to a toggle switch, but are much less easily bumped.  If you can get a pot with a switch, this would be even better, but these are now hard to get - especially as linear.
 
NOTE Make sure that you do not connect any of the internal circuit to the case.  As shown, the unit will be quite happy on phantom powered (48V) mic lines - this will not be true if you connect to the case.


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Direct Injection Box for Recording & PA Systems
Updated 08 Jan 2005


Introduction

A Direct Injection (or DI) box is a very handy piece of equipment for any public address rig or recording studio, whether for band or general use. It will allow you to connect the output from guitar amps, keyboard mixers, tape machines and just about anything else directly to the mixer, without using a microphone, and with no hum loops.

The unit described will convert unbalanced inputs (such as from a guitar or bass amp) to balanced, allows the level to be set to something reasonable, and comes in two flavours. There is a completely passive version that uses a transformer to create the balanced send, or an active unit which can be operated from a 48V phantom feed or a couple of 9V batteries.


Description

Firstly, for those who may not know about phantom feed, Figure 1 shows how it is done. The 48V supply in the mixer is connected to both signal lines, so causes no current flow in transformers since both ends of the winding are at the same DC potential. At the remote end, the current is tapped off the lines using a resistance value suitable for the electronics. Again, this is done with both signal lines to ensure that there is no DC imbalance in the circuit.


Figure 1 - Phantom Powering

After filtering (and in some cases regulation as well), the DC is then available to power the circuit that drives the AC signal down the very same pair that provides the power. In all cases the shield must be connected at both ends, since this provides the DC return path (hence no earth lift switch).  In this example a microphone has been used, but the same concept applies to virtually anything that can function on the limited power available.

Figure 2 is the passive version of the DI Box, which is very easy to build. The only problem is that to get good sound quality, you will need a good transformer, and these are expensive. As can be seen, the input is simply two 6.5mm phone jacks to allow a speaker lead to pass through the unit. The output is a male XLR connector, and is balanced. Have a look at the Jensen Transformers (or any other audio transformer manufacturer) web site to track down a suitable unit. There are many other manufacturers, but I don't know them all.  See if you can find one in your country. The transformer is 1:1 ratio, and needs to be rated for 600 ohm operation (or higher).


Figure 2 - Passive DI Box

The switch selects either line or speaker level from the phone jacks, and the 1k pot allows you to set the level when using a speaker source. When using speaker input, the attenuator is variable to allow for the widely differing output levels available from amps. No "earth (ground) lift" switch is provided - these are often used to completely isolate the signal source, for those occasions where there is a hum loop created between the mixer and the stage equipment. Instead, there is an earth isolation circuit (the 10Ω resistor and the 100nF cap), which will be more than enough except in the most extreme cases.  The earth lift is only fully effective when the transformer circuit is used, and will prove worse than useless in an active unit.

The active unit uses the 48V phantom feed available in many mixers, but can be run from batteries if this is not available. To ensure that there is no unnecessary battery loading a LED has not been included.

The connections to the XLR have been shown on all the drawings, and the pin numbers are clearly marked on the connector, designations are ...

    Pin 1 Earth (Ground)
    Pin 2 Hot (+ve signal)
    Pin 3 Cold (-ve signal)

Note that in some cases (especially with older equipment of US origin), pin 2 is 'cold' and pin 3 is 'hot'. This connection scheme is not recommended, and should not be used. The above is as close to an official standard as you will find, and should be used in all cases.


Figure 3 - Active Phantom/ Battery Powered DI Box

An earth lift switch cannot easily be used with phantom powering without excessive complexity, and has not been included.  The 10 Ohm resistor and 100nF cap will be quite sufficient in all but the most stubborn of cases.

The opamps require some degree of protection from the applied 48V when the unit is connected, and this is provided by the diodes from the opamp outputs back to the power supply. Without these it is possible to damage the opamps as the output capacitors charge. Because some degree of mucking about would be normally be needed for the output capacitors to make the unit truly universal, these are specified as bipolar (non-polarised) types - standard electrolytics must not be used.

All resistors should ideally be 0.5W 1% metal film for lowest noise and best matching. Capacitors must be rated at 25V or more, and diodes are 1N4148 or similar. If you need as much level as you can get and don't care about a bit of distortion, then a low power opamp (such as the LM358) can be used. These draw a lot less current, so the supply voltage will be higher. This allows more signal before the opamp clips. Bear in mind that many low power opamps can supply less output current than the TL072, so you may not get any real benefit. This does not apply to the LM358 - it can supply more than enough current (and more than can be provided by the phantom power scheme).

Two versions of the active unit used to be shown here, but by using bipolar output caps the unit can be dual-purpose. When plugged into a phantom supply, make sure that the switch is in the phantom position to eliminate unnecessary battery drain. Likewise, always leave the switch in the 'Phantom' position when not in use.

If you want to make the unit phantom or battery only, simply leave out the parts that you don't need. For battery only, you don't need R8 and R9, and D3 (24V zener) can also be omitted. If the unit will only be used with phantom, then you can omit the Phantom/Battery switch and the batteries.


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High Quality Audio Mixer


Introduction

This project is probably the most ambitious so far, and can be expected to be very expensive.  On the positive side, it is also capable of excellent performance, and can be tailored to suit your exact specifications.  There are several different input modules in the series, the first being the microphone and line input.

The project is presented in parts, and Part 1 shows the mic/line module and has some background information on noise measurements and the general philosophy behind the project.

Stages 2 and 3 are now complete, and are described below, and Stage 4 is available, but still under construction.


Description


Since the project is presented in stages, this is an index for the various modules, and as the system is developed will also be the place to look for updates and other information.

    Updated
Stage 1 Microphone / Line input module - Includes optional 48V phantom feed, and shows three different input stage configurations.  You can choose from transformer input, or two different electronically balanced circuits.  Also shows the tone control circuits, peak level indicator and all channel to group/master switching, faders and pan pots. 07 Jan 2001
Stage 2 Basic Mixing Modules - These are used for mixing the stereo sends from each of the Mic/Line modules, either as sub-mixers or the main mixer (the same unit is used for each).  Also included are the Auxiliary Mix Module and balanced line driver circuits, and the first stage of the power supplies. 23 Oct 1999
Stage 3 Power Stages - This section shows the power supply regulators - both +/-15V main supplies and 48V phantom supply, and the headphone power amps. 26 Nov 1999
Stage 4 Bits and Pieces - The next installment will describe the pre-fade listen and other headphone mixing and switching, as well as the talkback mic amp, phono and auxiliary input modules.  (See NOTE below.)  Stage 4 is under construction, and contains descriptive text only at this time.

The additional modules, metering and a complete system layout will be added to the list as they are developed.

NOTE:  There has been a surprising amount of interest in this project, with a common requirement being a smaller version (an almost equally common request has been for a bigger version, too).  Scaling the project up is not really a problem, but it is difficult to know what you can leave out to make a small mixer of 6 channels or less.

It seems I have also managed to confuse a few people with some of the links between the various stages.  I used two different terms for the same one in one instance, and the links made sense to me at the time, but this has not helped some of my readers.  Sorry about this, and I will try to find and fix the errors (whether of nomenclature or common sense).

There is also some concern about the time taken for Stage 4.  I have been busily doing other things, and managed to push the mixer to the "back burner" - I am almost sorry I started this, but I will have to continue - after all, I got this far.  Bear with me, gentle reader - it won't happen overnight, but it will happen. (After all, I do have a "real" job - one has to eat ... )


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Headphone Amplifier


PCBs PCBs are available for this project now

Introduction

Firstly, I'd like to stress that the intended use of this circuit is only one of many possible applications. Apart from the obvious usage as a headphone amplifier, the circuit can be used for a range of applications where a wide bandwidth low power amplifier is needed. Some of the options include ...

  • Reverb drive amplifier - ideal for low and medium impedance reverb tanks
  • High current line driver - suitable for very long balanced lines
  • Low power speaker amplifier - better performance than small integrated amps
  • ... and of course, a headphone amp.
In short, the amp can be used anywhere that you need an opamp with more output current than normally available. Since most are rated for around ±20-50mA, general purpose opamps are not suitable for driving long cables or anywhere else that a relatively high output current is needed.

As a headphone amplifier, this design is very similar to others on the ESP site, but the main difference is that this one (and P70) has been built and fully tested. The design is fairly standard, and every variation was checked out before arriving at the final circuit. A photo of the prototype is shown below, and at only 64 x 38mm (2.5 x 1.5 inches) it is very small - naturally, the heatsink is not included in the dimensions.

The amplifier is capable of delivering around 1.5W into 8 ohm headphones, and 2.2W into 32 ohms - this is vastly more than will ever be needed in practice. The use of a 120 Ohm output resistor is recommended, as this is supposed to be the standard source impedance for headphones. Unfortunately, many users have found that their 'phones perform better when driven from a low impedance source.

Photo
Prototype Headphone Amplifier

The circuit is based on an opamp, with its output current boosted by a pair of transistors. Distortion is well below my measurement threshold at all levels below clipping into any impedance. Noise is virtually non-existent - even with a compression driver held to my ear, I could barely hear any, and I couldn't hear any with headphones.

WARNING
Headphones are rated in dB SPL at 1mW, and this amplifier (like many other similar headphone amps) is capable of producing extreme SPLs. The levels obtainable are sufficient to cause almost instantaneous permanent hearing damage! Never operate the amp at very high levels, and never switch the amplifier on with signal while wearing you headphones.

Always start with the volume control at minimum, and gradually increase the level until it is comfortable, but not too loud. Because of the very low distortion, it is easy to increase the level too far without noticing. Your ears are precious - safeguard them at all times.

Note the warning above - this is serious. Most headphones are capable of at least 94dB SPL at 1 mW, with some as high as 107dB SPL. Even 10mW is enough to create sound levels capable of causing hearing damage, so you must be very careful to avoid damaging levels.

Continuous dB SPL Maximum Exposure Time
85 8 hours
88 4 hours
91 2 hours
94 1 hour
97 30 minutes
100 15 minutes
103 7.5 minutes
106 < 4 minutes
109 < 2minutes
112 ~ 1 minute
115 ~ 30 seconds
Table 1 - Maximum Exposure to SPL

Note that the exposure time is for any 24 hour period, and is halved for each 3dB SPL above 85dB. The above shows the accepted standards for recommended permissible exposure time for continuous time weighted average noise, according to NIOSH (National Institute for Occupational Safety and Health) and CDC (Centers for Disease Control) [1]. Although these standards are US based, they apply pretty much equally in most countries - hearing loss does not respect national boundaries.


Description

The amplifier itself is fairly conventional, and is very similar to another shown on this site (see Project 24). This amplifier does not include the active volume control, because in general it is far easier to get a good log pot (or simply 'fake' the pot's law as described in Project 01). Likewise, it does not include the cross-feed described in Project 109. If this is desired, it is very easy to implement on a small piece of tag board, or even 'sky hook' the few components off the bypass switch. Full details of how to do this will be included in the construction guide when PCBs are available.

The output transistors are biased using only resistors, rather than constant current sources. Extensive testing showed that using current sources made no discernible difference to performance, but increased the complexity and PCB size. Using separate caps for each biasing diode does make a difference though - and although it is relatively minor, the use of the two caps is justified IMHO.

The bias diodes should be 1N4148 or similar - power diodes are not recommended, as their forward voltage is too low. This may result in distortion around the crossover region, where one transistor turns off and the other on. As shown, crossover distortion is absolutely unmeasurable with the equipment I have available.

Figure 1
Figure 1 - Headphone Amplifier Circuit Diagram

Above is the schematic of one channel. Resistors and caps use the suffix 'R' for the right channel. The second half of the dual opamp powers the right channel. Note that the volume control shown is optional, and is not on the PCB. If needed, it may be mounted in a convenient location and the output connected to the inputs of the board as shown. D1 and D2 (L and R) are 1N4148 or similar.

One of the reasons the amp is so quiet is that the entire board runs from a regulated supply, so hum (in particular) is eliminated. Although an unregulated supply can be used, this is not recommended. The supply should be separate from that used for your preamp, because of the relatively high current drawn by the amplifier (at least with low impedance 'phones). A P05 preamp supply can be used, and will ensure optimum performance.

The prototype amplifier has flat frequency response from 10Hz to over 100kHz. Distortion is below my measurement threshold with any level or load impedance, and output impedance is almost immeasurably low. Your headphones may be designed to operate from a 120Ω source impedance (many are), so this may be added if it improves sound quality. Adding any series resistance will reduce the available power, but it is already far greater than you can use. Without series resistance, the minimum power into various load impedances is given below (based on ±15V supplies).

Impedance Power (Direct) 120 Ohm Feed
8 Ohms 1.5 W 35 mW
32 Ohms 2.2 W 99 mW
65 Ohms 1.1 W 136 mW
120 Ohms 595 mW 149 mW
300 Ohms 238 mW 121 mW
600 Ohms 119 mW 82 mW
Table 2 - Output Power Vs. Impedance

This is not especially comprehensive, but will cover the majority of headphones in common use. In all cases, the available power is more than needed ... not so you can damage your hearing, but to allow adequate headroom for transients.


Construction

As noted, PCBs will be available for this project soon, and this is the recommended way to make the amplifier. While it may be possible to build it using Veroboard or similar, there is a high risk that it will oscillate because of the very wide bandwidth of the amplifier. A capacitor may be added in parallel with R4 (L and R) to reduce the bandwidth if stability problems are encountered. Although I used an NE5532 opamp for the prototype, the circuit will also work with a TL072, but at reduced power. You may also substitute an OPA2134 or your favourite device, taking note of the following ...

The standard pinout for a dual opamp is shown on the left. If the opamps are installed backwards, they will almost certainly fail, so be careful.

The suggested NE5532 opamp was used for the prototype, and performance is exemplary. Devices such as the TL072 will be quite satisfactory for most work, but if you prefer to use ultra low noise or wide bandwidth devices, that choice is yours.

Construction is fairly critical. Because of the wide bandwidth of the NE5532 and many other audio grade opamps, the amplifier may oscillate (the prototype initially had an oscillation at almost 500kHz), so care is needed to ensure there is adequate separation between inputs and outputs. Even a small capacitive coupling between the two may be enough to cause problems.

As shown in the photo, this amplifier needs a heatsink. While it can operate without one at low power using high impedance headphones, you need to plan for all possibilities (after all, you may purchase low impedance 'phones sometime in the future). The heatsink does not need to be massive, and the one shown above is fine for normal listening levels. An aluminium bracket may be used to attach to the chassis - I recommend 3mm material. Note that the heatsink should always be earthed (grounded).

The output transistors must be insulated from the heatsink. Sil-Pads™ are quite suitable because of the relatively low dissipation, but greased mica or Kapton can be used if you prefer. If you use the suggested 3mm aluminium, you can drill and tap threads into the heatsink, removing the need for nuts.


Testing

Connect to a suitable power supply - remember that the supply earth (ground) must be connected! When powering up for the first time, use 56 ohm "safety" resistors in series with each supply to limit the current in case you have made a mistake in the wiring. These will reduce the supply voltage considerably because of the bias current of the output transistors.

If the voltage at the amplifier supply pins is greater than ±6V and the output voltage is close to zero, then the amplifier is probably working fine. If you have an oscilloscope, check for oscillation at the outputs ... at all volume control settings. Do this without connecting your headphones - if the amp oscillates, it may damage them.

Once you are sure that all is well, you may remove the safety resistors and permanently wire the amplifier into your chassis.



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Portable Headphone Amplifier


PCBs will be made available for this project if there is sufficient demand


Introduction

The modern day dynamic headphone drivers are very efficient. Just a few milliwatts are sufficient enough for reaching SPL that can easily render you with permanent ear damage. Caution, therefore, is not just a recommendation, it is a necessity.

Headphones are by far the most affordable of all audiophile equipment. The quality of reproduction and SPL offered by even moderate headphones can easily be regarded as a performance standard for the most desirable of loudspeakers.

Still, headphone listening is not as blissful as it might have been expected. The headphone outputs of most commercial systems receive very little attention from the manufacturers. This neglect manifests itself in the form of cheap quality sound and frustrations for the listener. A dedicated headphone amplifier can easily cure these ailments.

In my case, it all started when I got myself a Sennheiser PMX 60 headphone. When connected to my Sony portable, the sound left a lot to be desired. As I increased the volume, the bass simply disappeared while the treble became a ringing in my ears with all the hostility of a raging gale. I tried the ‘phones with my IPAQ and this time the sound was even worse.

If you use Grado or any other low Z (≤ 32Ω) headphones, then this may very well be your song I'm singing. The built-in headphone outputs of most systems, by their very design, cannot keep up with the high current appetite of a low Z headphone.

My design goals for this amp were quite straightforward:

  • Punchy bass on demand
  • Portability
  • Low listener fatigue

The amplifier, as it now stands, sports three opamps per channel, one as the voltage gain stage and the rest as current amplifiers. That's a total of three dual opamps for stereo. There is also a crossfeed network sandwiched between these two active stages.

Crossfeed
To locate and externalise sources of sounds, we use both of our ears. The sound from a source on the right (say, the right speaker) is heard not only by the right ear, but also heard, delayed and attenuated, by the left ear. The brain compares the delayed and attenuated sound with the original to deduce the exact location of the sound source.

Of course, this is some what an over simplification as reflections at the ear pinnae and from the walls of the listening area also contribute complex information important to the localisation process. All the info from these sources is furthered by the movements of the head.

When listening to a headphone, all these sources of info are absent. Transducers mounted directly on the ears cause the unnatural 'super-stereo effect', where one ear doesn't hear, in any form, what the other is hearing. The perceived spaciousness, which doesn't occur in normal listening conditions, might be very impressive in the beginning but quickly fatigues the listener with headaches and occasionally, dizziness.

This is where a crossfeed comes in. It is an acoustic simulator of the simplest from. The crossfeed electronically mimics the inter-channel interactions of the real world by delaying and attenuating the signal from one channel and feeding it to the other.

The use of the crossfeed results in a realistically spacious sound stage where instrument locations seem more natural. The perceived depth also lowers the listener fatigue considerably.

The crossfeed presented was originally designed by a Swedish audio engineer named Ingvar Ohman. It was published in an article called "Den Lilla Stereo-kontrollboxen SP12" in the December 1994 issue of the "Musik och Ljudteknik" ("Music and Audio Technical Society") magazine.


Description

The headphone amplifier circuit is shown in Fig.1. As you can see, it is a very simple design requiring you to detach yourself from the wonderful world of weekend chores for just a few hours… I promise!


Figure 1 - Schematic of Headphone Amplifier

U1 is the gain stage and, as shown, has a gain of 4. The gain can be adjusted by changing the value of 3.3k resistor. A gain of more than 11 is not recommended.

SW1 bypasses the crossfeed network. I have reconfigured the original crossfeed schematic so that now the 100k resistor always bridges the bypass switch and thereby reduces any 'crackle' or 'click' or whatever you may call them. Don't omit these 100K resistors as they form a part of the crossfeed network and omitting them would bear undesirable results. Note that R6 and R9 are indicated as 4.53k, however the use of 4.7k resistors will be perfectly adequate in practice.

U2 and U3 are paralleled as current-boosting amps. This doubles the output current into the load as established by Burr-Brown's AB-051 application note: Double The Output Current To A Load With The OPA2604 Audio Opamp.

    Meraj suggests that the value of the output resistors might require a little bit of experimenting for optimally matching the amp with your headphones. However as shown, output impedance is close to zero, and changing R12 and R13 will not affect impedance. Most headphones are designed for an impedance of 120Ω, and I suggest that a 120Ω resistor be installed in series with the output.

The power supply pins were not shown in the diagram for clarity. These pins are bypassed by 10uF and 100nF decoupling caps.

Only one channel is shown, so two units are needed for stereo.


Construction

For the prototype, I used Veroboard and have found the amp to be very tolerant of layout. I've made boards based on the prototype and ESP may make them available to others when there is enough demand to offset costs. Needless to say, these boards would make construction a breeze. I used 1/4W carbon resistors throughout. Considering the level of ambient noise that a portable system has to put up with, the volume level would usually be high enough to make it impossible to discern noise from signal - however I still recommend metal film resistors for best results.


Figure 2 - Work In Progress

I chose the NE5532 for this project. Since the source is a PDA's internal DAC, I didn't see the need to use premium opamps. Of course, if it makes you feel better, you can always use higher quality (expensive) opamps. Just make sure the opamp is capable of driving low impedances. LM6171, OPA2134, OPA2132, OPA134 and OPA4134 (dual) are some possible substitutes. It's likely that there are others. IC sockets are therefore a good idea if you have plans to upgrade the opamps.

The volume pot should be a linear type and would give, with the 15k resistor in parallel, the benefits outlined in ESP's A Better Volume Control.

The crossfeed is on a separate board in the prototype. I mounted it vertically on the main board using hot-melt glue. All the switches, jacks and volume control were also mounted on the enclosure using a hot-melt glue gun. I used generous amounts of hot-melt glue around the bases of all the capacitors as they are more susceptible to lead and track breaking due to vibrations.

For the enclosure, I chose what used to be a part of a plastic school lunch box. I measured and marked the spots for the cuts I had to make. A sharp hobby-knife, a drill bit, a tabletop vise and a steady hand were all that I needed for the job. When working with plastic, it's a very good practice to measure twice and cut once (he spake from bitter experience).


Figure 3 - Testing … testing …1, 2, 3 …

The belt clip was made for Nokia and came into my possession when I bought a 7110 aeons ago. If you use a belt clip, level it to make sure that the amp, with all the jacks sticking out, doesn't get in the way of your belly when you sit down. Trust me, it can be very painful!

By far the most expensive part of this project is the paint job. I went through a can of flat black and a can of clear lacquer to get the finish.


Power Supply

My prototype uses two 9V alkaline batteries to give 9-0-9V supply. I get around 20 hours of operation at normal portable listening levels. The effect of the demise of a few pairs of alkaline batteries on my wallet has decided me to switch over to rechargeable batteries. A battery charger is now under construction.

This amp can also be powered by Project 05 using a 15-0-15V transformer. A 5VA transformer should have oomph enough for the job.

A star ground was not necessary for my battery powered version but is recommended for a mains powered one. Use the common point of the power filter caps as the ground return and employ a ground loop breaker if you use a metal enclosure.


The Fruits of Labour?

The first thing that you notice about the sound is the authority of the low frequency. The sensation of the kick drum's 'kick' (pun intended) on the earlobes greatly enhances the listening experience. Only now do I realise the full potential of the Sennheiser PMX 60.

With the crossfeed on, the vocal that used to seem to be right on top of one's nose is pulled forward. The perceived depth in the sound stage and the bass is very much dependent on the source material. For some materials, loss in bass is experienced with the crossfeed on. This is due to the cancellations of the unrealistic, out-of-phase signals.

I find the crossfeed to be satisfactory for listening to classical and pop, rock gets mixed results and death metal is less confusing because of the cleared up sound stage.

At the time of writing, I had just finished a 15-0-15V power adaptor based on the Project 05. The improvement in the sound brought forward by the increased voltage is just amazing! As my ears were recovering, I was on my way to hunt down an enclosure that would house the project along with four 9V batteries. That's ±18V - I must be crazy!


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Headphone Adaptor for Power Amplifiers


Introduction

This simple project is nothing more than a handful of resistors and a double pole, double throw switch, but will reduce the output of almost any amplifier to a nominal level of 5V RMS, and maintains the recommended 120Ω source impedance. This is designed to suit most headphones currently made, as they are generally designed to operate from that impedance.

Naturally, this is not always suitable (some manufacturers have chosen not to adopt the standard for one reason or another), but will suit most headphones very well.

The level of 5V RMS was chosen to ensure that the power amplifier will not clip when driving the headphones, but is much too high for normal listening. As always, you may make changes to suit your preferences, but be aware that nearly all headphones are capable of sound levels that will cause permanent hearing damage, so always be mindful of this.

Warning: This unit is not designed to be used with bridged amplifiers! If there is a warning on your amp that states that the -ve speaker terminals must not be grounded, then you must not connect this adaptor, or the amplifier will be damaged. If in doubt, find out first from the manufacturer or distributor - assumptions can be very costly!


Description

The project could not be simpler - it basically consists of a switch to disable the main speakers, and the attenuator to set the correct level and impedance. Figure 1 shows the circuit diagram of a single channel, and this is duplicated for the second channel.


Figure 1 - Schematic for One Channel

The only hard part in all of this is choosing the resistor values that will give you as close as possible to the correct voltage and impedance for all typical amplifier powers. Table 1 saves you the tedium of working this out, and all attenuators use standard value resistors. The nominal voltage and actual output impedance are also shown, and as you can see, the variation is very small.

Power - 8Ω R1 R3 Zout Vout R1 Power
20 W 180Ω 47Ω 119Ω 5.2 0.33 - 0.5W
30 W 270Ω 47Ω 122Ω 4.9 0.47 - 0.5W
40 W 330Ω 33Ω 121Ω 4.8 0.54 - 1W
65 W 470Ω 22Ω 118Ω 4.7 0.74 - 1W
100 W 560Ω 22Ω 121Ω 4.9 0.95 - 1W
150 W 680Ω 18Ω 120Ω 5.7 1.37 - 2W
250 W 1kΩ 12Ω 119Ω 5.1 1.79 - 2W
Table 1 - Resistor Values for Different Power Amplifiers

The table shows the nominal amp power (8 ohms), and the values for R1 and R3 (marked with a * in the schematic). The actual voltage available to the headphones is also shown (Vout) as is the maximum power for R1 and the recommended power rating for that resistor. R2 is fixed at 120Ω for all power levels. Should you need more (or less signal) for your headphones, you may simply use the values for the next lower (or higher) amplifier power. For example, if your amp is 60W and you want less level for the headphones, use the values for a 100W amp.


Construction

To construct the circuit, you will need a double pole, double throw switch to disconnect the speakers, assuming that this is not already available. Do not be tempted to use a rotary switch, unless it is rated for the maximum amplifier output current - most are not. A heavy duty toggle or rocker switch is recommended, with a minimum current rating of 10A.

As shown, when the speakers are disconnected, the headphone adaptor is connected and vice versa. This prevents power being fed to headphones for no good reason, and also prevents "extraneous" sound when you are listening to the speakers. The entire adaptor may be installed in a separate box, with a speaker switch, headphone socket(s) and speaker in and out connectors.

This approach is assumed in the schematic, and will generally be the easiest way to provide headphone capabilities for an amplifier that does not have this ability. If more than one set of headphones is required, you must use a separate attenuator for each output - do not simply parallel headphones.

The "tip" of a stereo phone plug is the Left channel, the ring is the Right channel and the sleeve is Earth (Ground). If your amplifier has a balance control, you can check that the jack(s) are correctly wired by using the balance control to mute one channel.


Testing

Before connecting the unit to your amplifier, make sure that there are no wiring faults that present a short to the amplifier terminals. This can be tested with a multimeter, and you should also verify that the switch connects and disconnects the headphone attenuators and speakers in the correct manner.

The real test is to connect your amplifier and headphones, and verify that the level is correct, and that everything works as it should. This must not be done until you have checked your wiring thoroughly, and verified that there are no shorts - especially across the speaker leads!


Note Carefully Note: In use, make sure that the amplifier volume is set low to start with. Headphones vary considerably in impedance and sensitivity, and it is virtually impossible to determine the correct setting in advance.

It is very important that you always maintain a safe listening level - as stated above, headphones can produce extremely high SPL (Sound Pressure Level) - more than sufficient to cause permanent irreparable hearing damage!


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DoZ Headphone Amp -
A New Use For The Class-A Power Amp

 
Updated 03 Oct 2005

PCB Please Note:  PCBs are available for this project.  Click the image for details.

The quiescent current can be quite unstable with variations in the supply voltage. Normal changes in the AC mains can cause Iq to shift above and below the preset value. A simple modification is now included on the PCB that virtually eliminates the problem (or reduces it to the point where it is immaterial).


Introduction

You really need to see the original article - Project 36 - to see all the design details for this project. The project presented here is simply a modification of the original design, with much lower power dissipation and adapted specifically as a headphone amplifier. The circuit is identical to the original Death of Zen amp, except for the output transistors.

DoZ Photo
Photo of Assembled Rev-A Board

Class-A is ideal for this application, since headphones are such an intimate way of listening. An amplifier for 'phones should be as clean and free from crossover distortion as possible, and must also be quiet. A background of hiss and hum does nothing to enhance the listening experience.

Headphone amps are somewhat misunderstood, but in reality there are few points that need to be made. Most 'phones are designed to be operated with a source resistance of 120 ohms, and damping factor (as applied to conventional loudspeakers) is largely irrelevant. The actual source impedance should have very little (if any) effect on the frequency response or dynamic behaviour, since there is no cavernous enclosure and no heavy cones to try to control.

The IEC 61938 international standard recommends that headphones should expect a 120 ohm source (5V RMS maximum) - regardless of the headphone's own impedance. If the manufacturer followed this standard, the 120 ohm resistor used in this circuit will not affect sound.

Power requirements are usually in the 10 to 100mW range, and this is quite sufficient to cause permanent hearing damage. With the current set for 330 mA as suggested, this amp will be able to drive a minimum of 2 (but probably 3) sets of headphones at once. With 40 Ohm 'phones, it can give a maximum power of over 150 mW, so caution is needed to prevent hearing (and headphone) damage. Even with 8 ohm 'phones, power will be about 110mW - more than enough to have you asking people to repeat everything they say!

Please Note Caution! Just in case you missed it, headphones are easily capable of causing permanent irreparable hearing damage. Modern dynamic headphones are very efficient (typically well over 90dB SPL per milliwatt) and will reach full volume with just a few milliwatts of input. A mere 100 mW will therefore provide a peak SPL of around 110dB SPL. The recommended maximum exposure to this sound level is less than 5 minutes in any 24 hour period !

Based on a maximum voltage of 10V RMS and a feed resistance of 120 ohms, the following table shows what peak power you should expect into various impedance headphones. Reducing the feed resistance will increase the power applied, probably to the detriment of your ears and the headphones themselves.

Impedance (ohms) Power (mW)
8 48.83
16 86.51
32 138.50
40 156.25
65 189.92
100 206.61
Table 1 - Power Vs. Impedance

You might need to adjust the value of the feed resistor(s) if you have really low sensitivity headphones, but unless it is absolutely necessary - don't !


Description

The final circuit for the DoZ headphone amp is shown in Figure 1. It is almost identical to the original (well, apart from the output transistors and size of C3, it is identical), and there is no longer the need for massive heatsinks and TO-3 output transistors. As shown, there are outputs for 2 sets of headphones. Needless to say, only one channel is shown - the other is identical.

For final testing you will need a multimeter. As shown in the power supply circuit below, use a 10 Ohm resistor in series with the power supply positive lead. When you measure 1 volt across this resistor, this means that the amplifier is drawing 100 mA. The resistor remains in circuit, providing a useful reduction in supply ripple. You will lose about 3.3 V at operating current, and a 5W resistor is sufficient - it will get slightly warm. The output resistors (120 Ohm) should be rated at at least 2 Watts - a pair of 220 ohm 1W resistors in parallel will do just fine (the absolute value is not critical).

Figure 1
Figure 1 - DoZ Headphone Amplifier

Although MJL4281 transistors are shown in the circuit diagram, you can use cheaper devices for a headphone amp. If you want the highest possible reliability and best performance, those shown are a very good choice. Alternatives are TIP35 (A, B or C), MJL21194, or TIP/MJE3055. TO3 devices can also be used, but must be mounted off the PCB.

C3 should be 470uF to 1,000uF. The higher value is recommended if you intend to drive multiple sets of headphones. The value of C3 is determined based on the use of 120 ohm feed resistors to the headphones. You will need to use a higher value if you use a lower resistance (not recommended, but some 'phones seem to prefer lower source impedance).

D1, D2 and R11 are optional but highly recommended. Full details for determining the zener voltage and resistance for R11 are given in the construction page. R13 may be omitted if desired. It helps to stabilise the bias current, but a side effect is slightly increased distortion.

Please Note Q3 and Q5 (the output transistors) must be on a heatsink (see below), and even for headphone use, Q2 and Q4 may require a small heatsink.

A quick circuit description is in order. VR1 is used to set the DC voltage at the +ve of C3 to 1/2 the supply voltage (20V for a 40V supply), by setting the voltage at the base of Q1. The 100uF cap ensures that no supply ripple gets into the input. Using a larger value will prevent any thump into the headphones as C3 (the output capacitor) charges, but there may be a period where excessive output current is drawn. The voltage rise is slow enough that there is little audible noise heard as the amp is powered on. Q1 is the main amplifying device, and also sets the gain by the ratio of R9 and R4. As shown, gain is 13, or 22dB, providing an input sensitivity of about 1V for full output.

Q4 is the buffer for the output transistor Q5, and modulates the current in Q2 and Q3. VR2 is used to set quiescent current, which I found needs to be about 330 mA for best overall performance. C4 and R6 are part of a bootstrap circuit, which ensures that the voltage across R6 remains constant. If the voltage is constant, then so is the current, and this part of the circuit ensures linearity as the output approaches the +ve supply.

If the DoZ PCB is used, the output components (C3, the two 120 ohm 2W resistors, and the 1k resistor to earth) are mounted "off-board". The output resistors are best mounted directly to the headphone jack, and the remaining parts can be mounted anywhere convenient.

Before applying power, set VR1 to the middle of its travel, and VR2 to maximum resistance (minimum current). Be very careful - if you accidentally set VR2 to minimum resistance the amp will probably self destruct - more or less immediately.

With an ammeter in series with the power supply (or measure the voltage across the 10 Ohm power supply resistor), apply power, and carefully adjust VR2 until you have about 330mA. Set VR1 to get 15V at the +ve of C3, and re-check the current. As the amp warms up, the current may increase, and you need to monitor it until the heatsinks have reached a stable temperature. If necessary, re-adjust VR2 and VR1 once the amp has stabilised. If you use a heatsink smaller than about 2°C/W the amp will overheat and will be thermally unstable - this is not desirable (note use of extreme understatement :-)

I used a 30V (nominal) supply, and was able to obtain 150mW into typical 40 ohm headphones at the onset of clipping. Like the original, clipping is a lot smoother than most solid state amps, and the amp has no bad habits as it clips.

Figure 2
Figure 2 - Wiring of a Headphone Plug

Figure 2 above shows how to wire a standard stereo headphone plug. The tip is the left channel, the ring is the right channel, and the sleeve is earth (ground). Use an ohmmeter or continuity tester to determine the channel designations of the solder lugs inside the jack plug body. With a headphone jack, insert a headphone plug with known wiring scheme and use an ohmmeter or continuity tester to match the jack connections to the plug. Use this scheme when wiring the socket(s) to ensure that Left and Right channels are not reversed. The proper connections are shown in Figure 3.

Figure 3
Figure 3 - Phone Jack Wiring

Test Results

On the basis of the tests, I would rate this amp at 150 mW into 40 Ohm headphones, although I did get a little more. Distortion probably rises with increasing level, but I have no way of knowing, as it is so low - even at 10V RMS output into a 50 ohm load the distortion was about the same as the residual of my oscillator, which means that it must be below 0.04%, but I have no idea just how low it gets.

I simply used components as I found them, and did no matching or any selection. All test results are based on the prototype, which uses ordinary resistors, a couple of old salvaged computer caps for the high values, and standard electrolytics for the others. The input capacitor is an MKT polyester type or you can use a standard electrolytic if you want to (the positive goes to the junction of R1 and R2).

Supply Voltage 30V
Suggested Quiescent Current 330 mA
Maximum power (40 ohm 'phones) 350 mW
Output Noise (unweighted, 1k ohm source) <1 mV
Distortion @ 1kHz, 10V RMS at output < 0.4%
Output Impedance 120 ohms
Frequency Response (-0.5dB @ 100 mW) <20Hz to >50kHz
Table 2 - Measured Performance of Figure 1

I could hear no noise at all, even with a very basic power supply. The output noise level I measured was about 0.5mV, but it is not easy to measure accurately at such low levels. There appeared to be no residual hum that I could see on the oscilloscope, even with averaging turned on.

The amp will also tolerate an indefinite short circuit across the headphone socket(s) with no ill effects, and even (blush) reverse polarity. I accidentally connected the supply up backwards while testing the original, and thought "Oh, no. Now I'll have to rebuild the blessed thing" (if the truth be known I thought something much shorter!). However, I connected the supply the right way 'round, and away it went, as if nothing had ever happened. This is not an experiment I suggest to others.

The design is also unaffected by quite a few component variations. When I first started testing the original DoZ amp, there were no emitter-base resistors in the current source, and when I added them, I simply readjusted the two pots to get everything back where it was. I retested distortion after making the changes, and could measure no difference.

I have also designed a simple, high performance preamp circuit (all discrete Class-A), which is very nice indeed (see Project 37). The distortion is very low, and frequency response is excellent.


Bias Stability

As the supply voltage changes with normal variations in AC mains voltage, the quiescent current also shifts. This is not desirable, and is easily solved with the addition of a resistor and a zener diode (or a series string for odd voltages). If you are using a regulated supply, this mod is not needed. These parts are provided for on the Revision-A PCB, and the construction notes give the information needed to calculate the Zener voltage and series resistor.


Heatsink

As I have said before, this amp needs a fairly good heatsink, as do all Class-A amplifiers. Even 'though this amp runs at very low current, a good heatsink is recommended. Thermal resistance should ideally be no greater than about 2°C/W, so with a dissipation of about 10W the heatsink will be 20 degrees above ambient temperature. This is still quite hot, and a larger heatsink will not hurt one little bit :-)

If you can't keep your fingers on transistors, then they are hotter than I like to operate them - I know they will take much more, but it shortens their life. A small heatsink is also recommended for the drivers, as they get surprisingly warm without one.


Power Supply

A suitable supply for a pair of DoZ headphone amps is shown below. I must firstly give this ...

WARNING: Mains wiring must be done using mains rated cable, which should be separated from all DC and >signal wiring. All mains connections must be protected using heatshrink tubing to prevent accidental contact. Mains wiring must be performed by a qualified electrician - Do not attempt the power supply unless suitably qualified. Faulty or incorrect mains wiring may result in death or serious injury.

A simple supply using a dual 25V secondary transformer will give a voltage of around 35V. Allowing for the voltage drop across the 10 ohm resistor, this will give a typical supply voltage of a little under 30V for each amplifier. The actual voltage is influenced by a great many things, such as the regulation of the transformer, amount of capacitance, etc. For a pair of amps, a 50VA transformer will be (just) sufficient. Feel free to increase the capacitance, but anything above 10,000uF brings the law of diminishing returns down upon you. The performance gain is simply not worth the extra investment.

The amp is quite tolerant of supply ripple, and a simple supply will almost certainly be fine. A suitable power supply is shown in Figure 4, or for the perfectionist, use the capacitance multiplier circuit (Project 15). There really is no need for anything more than the circuit shown below - supply ripple is less than 12mV RMS when loaded, and no hum was heard at all. The added advantage of the circuit shown is that it will self correct (to some degree) variations in quiescent current with supply voltage.

Figure 4
Figure 4 - Suggested Power Supply

For the standard power supply, as noted above I suggest a 50VA transformer as a minimum - 100VA is preferred. For 115V countries, the fuse can remain as 2A, and a slow blow fuse is required for toroids because of the inrush current of these transformers. If using a conventional laminated transformer, then fast blow fuses should be OK.

IMPORTANT ! Note that the secondary windings are in parallel, and the dots indicate the start of each winding. When windings are paralleled it is imperative that the phasing is correct, or the main fuse will blow. In some cases, the transformer may be damaged by the overload.

The supply voltage can be expected to be higher than that quoted at no load, and less at full load. This is entirely normal, and is due to the regulation of the transformer. In some cases, it will not be possible to obtain the rated power if the transformer is not adequately rated.

R2 and R3 should be 5W wirewound types, the bridge rectifier can be a 5A type if you want (35A bridges are cheap enough, and the latter are preferred), and filter capacitors should be rated at a minimum of 50V. Wiring needs to be heavy gauge, and the DC must be taken from the capacitors - never from the bridge rectifier.

As shown, a separate feed is used for each channel. I strongly recommend this approach to ensure that there is no low frequency interaction between the amps.



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امروز می خوام به سوال یکی از بازدیدکنندگان جواب بدم. در مدار امروز که توسط یکی از بازدید کنندگان سایت به ما معرفی شده می پردازیم.این مدار یه شوکر ولتاژ بالا می باشد که قادر به تولید یه شوک با ولتاز 75000 ولت و توان 25000 وات می باشد ( خداییش مدار جالبیه با این همه ولتاژ بالا شوک ان باید چند ثانیه بدن را بی حس کنه). البته یکی از کاربرد های این دستگاه برای دفاع می باشد ....


ادامه مطلب
+ نوشته شده در  86/09/02ساعت   توسط احد دانشمند  |