A Tube Headphone Amplifier/Preamp with Relay-Based Input and Power Switching.

by Helmut Ahammer


[Warning: Like other projects using tubes for amplification, the circuits described in this article contain high voltages, and, therefore, the risk of a lethal accident is evident. Neither the author nor HeadWize is responsible for any damage or harm resulting from the construction of this project. DIYers should be familiar with and follow high-voltage safety precautions when building this amplifier.]

Tube amplifiers designed for headphones have the principal property that they can be used as preamplifiers too. In most cases, the output impedance of a tube headphone amplifier is (or should be) less than the output impedance of a tube preamp. Given this advantage, the use of a tube headphone amplifier for preamplification is not so critical concerning the input impedance of the power amp or cable impedance and cable capacitance. This project expands on the basic amplifier, with a slow turn-on for tube heater and plate voltages and input section using high quality relays.

The main intention of the project was the minimal use of elements in the audio path and the creation of a fully-featured amplifier with high audio impact. Therefore, additional discrete semiconductors and digital (logical) integrated circuits were brought in, serving only as helping devices. Only one tube in the audio path amplifies the audio signal. Here I have to say that I prefer the pure Class A topology, using only a single amplification stage with a parallel connected-output triode without overall feedback. This topology is not new and is for instance also mentioned in the Top-Level OTL Tube Headphone Amplifier and No-Compromise Tube Headphone Amplifier by Andrea Ciuffoli.

This amplifier features relay-based input and power switching. The stepped and slow turn-on for the power supplies result in less stress for the tubes and other components and reduces turn-on thumps that could damage headphones. Most tubes fail at turn-on. Without the slow turn-on, if the tube heater filament is cold, the resistance is lower and the in-rush current of a cold filament could be very high and cause the filament to break. And if the filament is not hot, it is better that the plate voltage is not applied. Applying the plate voltage with cold filaments can reduce the lifetime of the tubes.

I prefer relay input switching over input switching with a rotary switch for two main reasons. First, it is very hard to get a really good rotary switch, and when available, it would be very expensive. In contrast, there are many relays on the market, the contact material is normally very good, and they are relatively cheap. Second, it is possible to locate each input relay very near to the corresponding input jacks. Then there is the big advantage, that the wiring from the input to the switch is very short, and only one stereo cable must be routed through the amplifier. If the tape loop is used, then there is also a very short wiring for the audio path.

The circuits in this article are presented more or less as independent from each other, providing the possibility of an easy change if desired. The headphone amplifier could be built without the preamp sections (input relays and so on) or any part of the power supply could be changed (for example, valve rectified) or a preferred amplifier topology not presented here could be extended with the preamp sections.


Headphone Amplifier/Preamp

Figure 1

Figure 1 shows the circuit of the tube amplifier designed without global feedback. It is a pure Class A OTL design with a triode-connected pentode output section. The input section was built with the double triode E83CC as a differential amplifier. The big advantage of this circuit is that the output signal can be taken from the first or from the second triode. The output at the plate of the second triode isn’t phase inverting providing that the whole amplifier isn’t phase inverting. The E83CC is very often mentioned as a tube with excellent audio properties. There are selected types and different brands on the market. The E83CC is the high-grade type of the ECC83. It has a very high gain (µ = 100) and is therefore well suited for building up an amplifier with only one amplifying device.

To circumvent having a high voltage negative power supply for the current source (simply R4) of the differential amplifier and for improving the linearity, it is necessary to lift the cathode voltage and the grid voltage of V1 above zero volts. This is done by R7 and R8 acting as a voltage divider. This voltage divider could be built two-fold, each for one of the triodes, but since tubes have a negligible input current (gate current), the same divider gives the gate voltage for the first triode too through R6. With a gate-cathode voltage of about -1.2V and R4 = 4.7K Ohm, the plate current for each triode is about 1mA. This current increases the lifespan of the tube and gives very good results. The resistors R2 and R3 are the plate resistors and provide a plate voltage of about 170V (with a 280V power supply).

The higher the power supply voltage the higher the obtainable gain. With a power supply of 280V, the gain of the input section was measured to be about 22dB. R1 and R5 are gate resistors commonly used to reduce RF oscillations. Because the differential pair only uses one input, the other input must be grounded. C2 grounds the second input at audio frequencies. The gate is at a lifted potential for DC voltages, but is at ground for AC voltages. Like the input decoupling capacitor C1 and the interstage coupling capacitor C3, the gate capacitor C2 should be high performance too. Industrial standard MKP types (e.g. Epcos or Wima) are recommended.

The output stage is a cathode follower (gain < 1) with a triode-connected pentode EL84 instead of paralled triodes. If I had used triodes with high enough current (30mA or more), the plate voltage would have been, say, about 100V-120V. Then the voltage at the cathode resistor would have been in the range of 160V-180V. As the maximum cathode heater voltage must be less than 100V or in practical terms less the 80V, the ground plane of the heater supply would have to be lifted by 80V-100V. But lifting the ground plane with all the implemented control circuits (relay control, CMOS circuit, etc.) is not good. Furthermore, I dislike high voltages at the output capacitor. A fault of this capacitor could pose a lethal risk. Therefore I decided that the cathode resistor voltage should be not more than 80V. But what to do with the remaining 200V! There are no noval socket triodes (relatively cheap triodes) that can withstand this high plate voltage.

The solution I found was the triode-connected EL84 with a 210V plate voltage and about 35mA current – permitting a cathode resistor with lower wattage and lower cathode voltage! The drawback is the higher output impedance compared to paralleled triodes. The triode mode of the EL84 has lower output impedance and distortion compared to the pentode mode. R11 connects the second grid to the anode for the triode mode. The cathode current of 35.5mA is set with the cathode resistor R14 and the biasing resistors R12 and R13. The paralleled biasing resistors could be changed to one resistor with a value of about 180 Ohm and with a higher power handling capacity.

The output impedance of the cathode follower is approximately calculated with Rout = 1/S (S is the transconductance of the triode connected pentode). From the data sheet plate current/ gate voltage graphs in triode mode a value of S= 16mA/V, slightly higher than 12mA/V for the pentode mode is arrived and therefore Rout calculate to about 60 Ohms. This value works fine with headphones of 300 ohms impedance concerning the damping factor. By using headphones with about 30 Ohms, not only there is no damping, but furthermore the gain of the stage would be reduced. If gain isn’t that of importance low impedance headphones could give good results too, despite of the lack of damping.

Possible changes:

The resistor R4 with an actual value of 4.7K Ohms is not a real current source as the impedance of a current source should be much higher (about 1M Ohm). Implementing a better current source, it is possible to replace R4 by a choke or a FET current source. Using a BF245 with a resistor connected between Source and Gate could be replaced with no change for the rest of the circuit. A potentiometer could make the current adjustable. Bypassing R2 by a capacitor increases the gain of the input stage and would lower noise of the circuit. With the experience of the used power supply, which is very stable, noise was no problem and the gain without this capacity is well enough for driving 300 Ohm headphones.

With low impedance headphones, say 32 ohms, the output tube could be used with a current of up to 49mA instead of 35.5mA but then the voltage of the tube, the cathode voltage or the power supply voltage have to be changed. The impedance at the output is connected in AC terms parallel to the cathode resistor and the gain is reduced. Therefore, the maximum voltage swing is limited, and the current through the load is less. Adding an output transformer with a high primary impedance and a low secondary impedance could solve, but it would be better to integrate a transformer directly to the anode or cathode of the tube. Personally I prefer the transformerless approach, because these transformers need to be in the most cases very special ones and therefore are very expensive.

I thought of a using a real Class A MOSFET Follower instead of a tube output stage. Combining the gain (and the sound) of the E83CC tube with a MOSFET (low impedance) output is very interesting but actually I haven’t built such a circuit.

Heater DC Supply with Slow Turn-On

Figure 2

To minimize the possibility of hum problems (arising from AC heating of the tubes), a regulated power supply is used. Furthermore a slow turn-on was implemented (which could be found in similar form in many textbooks about tube circuitry) to limit the big in-rush current (figure 2). This slow turn-on increases the lifespan of the tubes. C1, C2, C3 and C4 reduce spikes from the rectifier diodes. Instead of the four capacitors (C5, C6, C7 and C8), one big capacitor could be used but several capacitors decrease the ohmic losses. The adjustable voltage stabilizer LM317 is used with the diodes D5 and D6, which are safety diodes in cases where negative voltages occur (mainly from turning off the amplifier).

The combination of R1, R2 and P1 set the output voltage. The combination of R4, C9 and Q1 gives the temporal behavior of the circuit at turn on. At the first moment C9 isn’t charged. Charging the capacitor through R4 gives a relative high voltage at this resistor and therefore Q1 is fully switched on. Q1 gives a very low parallel resistance and a very low overall resistance for the LM317 and therefore a very low supply output voltage. By charging C9 the overall resistance for the LM317 and the output supply voltage are raised. The output voltage before diode D7 is about 13.4V (the actual value is not critical) and is for the relay and control sections. The heaters of tube V1 (E83CC) of the left and right channel are parallel-connected. The heaters of tube V2 (EL84) for the left and right channel are connected in series, because they work only with 6.3V.

High Voltage DC Supply with Stepped and Slow Turn-On

Figure 3

The high voltage supply is turned on after a 40-second delay through a current-limiting resistor for slowing down the voltage ramp-up. For the first 40 seconds after turning on the amplifier, only the heater and the relay and control sections are powered. There is no voltage at all at the plate of the tubes (e.g., the output voltage of this supply is 0). After this 40 seconds, the control circuit turns on the relay Rly1 and the power supply for the plates begins to work (figure 3). As the MOSFET (Q2) configuration has also a slow turn-on function, the plate voltage rises slowly and, therefore, the tubes are not stressed. After 60 seconds, Rly2 shorts R1 and the voltage increase is accelerated. R1 and the contact of Rly2 could be placed at the primary of the transformer to limit the in-rush current of the transformer too.

The transformer is an easy-to-get 1:1 or 1:2 (depending on the primary voltage) transformer and should have a power rating of at least 100VA. R2 and R3 effectively increase the impedance of the rectifier diodes. This mimics in a minor manner a tube rectifier because tubes have a higher intrinsic impedance when compared to rectifier diodes. C1 and C2 minimize voltage spikes originating from the rectifier. C3 is actually built from six 150µF capacitors providing fewer losses. R4 and R10 are directly and closely connected to the capacitors to minimize the danger of charged capacitors if there is a fault (e.g., loosened connections to other circuit parts). Q1, D3, D4, R5 and DZ1 set the reference voltage of 285V, which is further smoothed by C4, R7 and C5.

Connecting the headphones (300 Ohms), there is absolute no audible hum or noise, except the volume pot is at the position 9 of 10. At this position it is able to hear the amplified input fluctuations but nevertheless no hum from the power supply. R9 is a safety resistor, if the connection to the output capacitors C6 is broken.

Relay-Based Input-Output Switching Section

Figure 4

The input-output switching section with input resistors for decoupling, a tape loop function (Rly7), the volume pot P1 (high quality recommended), the amplifier itself, an output relay Rly8, decoupled line outputs and the headphone output are shown in figure 4.

It is not necessary to buffer the inputs. There are different resistor values for the decoupling resistors mainly because today’s CD Players have 1-2Vrms outputs and many other inputs (like tuners, etc.) have outputs with less voltage. Therefore, I decided to halve the output voltage of the CD player, as then there is only a small change of volume when the inputs are switched between the CD player and an other units. The resistor values for the CD-input are relatively low because the output impedance of a CD player is normally relatively low too. The resistor values for the rest of the inputs are relatively high (the resistors to ground) because many tube amp owners have other tube gear to be amplified. For this case, the resistors should be not too small as the output impedances are normally higher. Principally, high resistances have the drawback of producing more noise, and the actual values of the decoupling resistors do not have to be exact.

The tape loop could be used to implement a signal processing circuit like EQ or crossfeed when using the amplifier for headphones. The input relays are switched by darlington transistors (BC517). The electrolytic capacitors (C7 – C10) minimize switching pops or crackles.

The tape loop relay (Rly7) and line output relay (Rly8) could be turned on with a normal switch, or with a momentary switch in combination with a NAiS VS5-24V electronic switching circuit and a bistable relay (Rly9 and Rly10) as shown in figure 4. The VS5-24V integrated circuit and a cheap polarised 2-coil relay (from any company) has the function of an expensive and hard to find stepper relay (a relay with two stable switched positions set by pulses).

The same circuit is used for the line output relay (Rly8), but in this case the relay can be switched only after the whole turn-on delay of 70 seconds. When the Tape/Line momentary switch is pressed, the second switch contact of Rly9/Rly10 (between the coil of Rly7/Rly8 and ground) is switched permanently. The other switch contact of Rly9/Rly10 between the two coils of Rly9/Rly10 is only necessary for the proper function of the VS5-24V. LEDs on the front panel indicate the status of Rly7/Rly8.

Turn-On Delay Control

Figure 5

Figure 5 shows the circuit for the turn-on delays which control the course of the turn-on procedure. The CMOS circuit 4060 is a binary counter and Q14 is the highest bit. C3 and R3 give the frequency of the oscillator and therefore the delay duration. At the start, C1 is not charged and pin 12 (RESET) of the 4060 is HIGH and resets the counter. Charging C1 sets pin 12 LOW because of R1, preventing a fault reset. The first time Q14 goes HIGH (after 40 seconds), Q1 is turned on. If, in addition, Q13 goes high (after 60 seconds), the NAND gates IC2C and IC2B turn on Q2. And finally if, in addition to Q14 and Q13, the pin Q12 goes HIGH (after 70 seconds), the gates IC2D and IC2A turn on Q3. D1 stops the oscillator to preserve this state. Changing the oscillator frequency would change the total duration of the whole delay, but would preserve the proportional timing. The formula for setting the frequency is: f = 1/(2.3 * R3 * C3). C1/R1 are reset at power-on.

This circuit starts every time in the same manner, and all the steps are cycled through with the depicted timing, regardless of the usage of the main switch (for example, if the main power switch is turned off and on improperly).

Flashing Led Signaling Slow Turn-On Procedure During Power-On

Figure 6

The circuit in figure 6 provides a rough visual display of status of the different steps of turn-on (and therefore representing the state of the amplifier during power on). Q1/Q2, D4/D5, C1/C2 and R2/R5/R6/R9 are the common components of an astable oscillator circuit. When the amplifier is powered on, the LED (D3) flashes slowly with the time constants given by C1, C2, and mainly R5 and R6. There is no second LED in connection with R2. This asymmetry ensures that the circuit starts oscillation under all circumstances.

After 40 seconds, the turn-on delay control for Rly1 (figure 3) also turns the transistor Q4 on. Then R6 is paralleled with a low resistance and gives a shorter time constant with C2 and, therefore, a faster flashing rate for the LED. Similarly, after 60 seconds, the Rly2 (figure 3) and the transistor Q3 are turned on and the LED flashes again faster. The diodes D1 and D2 prevent the toggle circuit from incorrectly turning on the relays. Playing with the values of R4, R7, C1 and C2, it is possible to change the flashing intervals of the LED. The depicted values are given for operation where the no-light durations are smaller than the light durations.

D1 and D2 prevent the toggle circuit from incorrectly turning on the relays. A “low” signal, produced periodically by Q1/Q2 could eventually be conducted through C1/C2, Q3/Q4 and R3/R8 to the coils of Rly1 and Rly2 during the delay time. Probably the resulting currents to the coils would be far too small to really turn on the relays, but I wanted to prevent it in any case.

And finally after 70 seconds, the output relay is turned on. Q5 and R1 stop the toggle function of the circuit and the LED operates in continuous mode. Overall this gives a very convenient optical control of the start-up procedure with one LED, which is located at the very left side of the faceplate directly above the main power switch.


The 6BQ5 and The Russian 6P14P (6P14P) in figure 1 can be substituted for the EL84 with very good results. The decoupling output capacitor (C4) should have at least 220µF with an appropriate voltage rating. A high voltage rating with a big amount of margin for this cap lowers the possibility of a failure (70V DC at the output!) and is therefore strongly recommended. Raising the value of the capacity gives better low frequency results. Bridging with an 1µF MKP type is recommended. All resistors are of 0.4W-0.6W type if not otherwise specified in the schematic.

In figure 2, if the BC557 transistor is not available, try 500mW types such as the BC556, BC327 and even the 300mW types like the BC177 should work as well. I’m not aware that there is a direct substitute for the ZTX758 in figure 3, because few PNP types are rated at 400V. If the ZTX758 is absolutely not available, then I would recommend 300V types such as ZTX757 or MPSA92. Substitutes for the IRF840 MOSFET include the IRF740, IRF830 or BUZ40B or any other N-channel MOSFET with a rating higher or equal than 400V, 4A should work. Types with less Rdson should be preferred. The fuse labelled 75 degrees (C) is a temperature fuse mounted inside the chassis which cuts the high power supply, if the temperature of 75° is reached. There are no direct substitutes for the BC517 darlington transistors in figure 4, 5 and 6, but the BCX38C could be used. For the BC107 transistor in figure 6, any small NPN transistor like the BC337, BC546 or 2SC1815 would also work.

The MOSFET IRF840 (figure 3) needs a heatsink with a temperature resistance of less than or equal to 6K/W, and the voltage regulator LM317 (figure 2) needs a heatsink of less than or equal to 4K/W. The “hot” parts of the amplifier (tubes, heatsinks, ev. transformers) should be equally distributed in between the chassis. Mounting the heatsinks in a way that they are an outer part of the chassis should be preferred.


At this time, there is really no information on the net about the NAiS VS5-24V module. I have a single data sheet from the local distributor. RS components and some local distributors here in Europe (e.g., Schuricht Elektronik) sell it. Furthermore I bought a small quantity (2 pieces) direct from the local NAiS distributor. The price is about Euro 9.30. There is no substitute, but I think it would be not so hard to develop a discrete version by any advanced DIYer. “Aromat” is the USA brand for NAiS, but NAiS does not know if Aromat sells the module. If not, it could be purchased from NAiS in Europe. I recommend contacting Aromat about the module and the availability.

If it is impossible to get the VS5-24V module, omit the Rly9/VS5-24V combo and replace the switch SW9b with a push button single pole switch. Electrically this switch could be a cheap one, because neither audio signals nor high currents nor high voltages would be switched. The same for Rly10.

All of the relays should have 12V coils with a resistance greater than 700 ohms. Rly1 and Rly2 have to be good power relays, such as the Siemens V23092 (any reliable monostable relay rated for 230VAC, 4A-6A works well in this application but cheap products should be avoided). They need only one switching contact (SPST). The switch contacts of Rly1 and Rly2 should NOT be moved after the rectifier to switch any high voltage DC – they should switch only high voltage AC. Normal “230V” relays are not constructed to switch 230VDC. I once had a bad experience with a relay specified for 320 VDC to switch 280VDC. The contacts melted together!

The use of high quality 12V relays for Rly3 – Rly8 is highly recommended (e.g., P2 Siemens or DS Nais DPDT types). For switching audio signals, it is important that the relay has a low and constant switch contact resistance at low voltages and currents. The resistance of a relay switch contact is specified with a minimum and maximum rating (the minimum rating should be as low as possible). For example, the Siemens P2 relay has a minimum voltage rating of 100µV (modern power relays (e.g., Siemens V23092) have a minimum voltage rating of at least 5V). The bipolar relays (Rly9/Rly10) have DPST switches. They do not have to be of the same quality, because they aren’t in the audio path and don’t switch high voltages or currents.

The part number of the Siemens P2 relay is V23079. DigiKey in the U.S. sells a Potter and Brumfield V23079 relay that is the same as the Siemens P2 relay. All Siemens relays have this type of number code. Despite that there are slightly different dimension values, I think it is really the same relay, especially since the coil current is exactly the same. The bistable relays Rly9 and Rly10 can be cheap products, because they don’t switch high power and don’t switch audio signals. They must be polarized, 2-coil types and must have at least DPST contacts.

Using a high quality momentary switch for the Tape and Line controls results in a very professional feel. I used the Apem type 18535CD, which costs about Euro 5.84.


As can be seen in the picture at the top of this article, this project could be built up in a relatively small case. The front plate was made of a 15mm ceramic plate. Ceramic is very heavy but absorbs vibrations very well and is therefore well suited for audio gear. The surface is manually fire lacquered giving a slight structure to the appearance. This plate is not 100% flat and uniform but therefore reflects on the other side the hand made aspect and individuality and completes for my opinion the tube approach very well.

To make the faceplate, I used white clay which deforms not so much when fired, and pressed it into a wooden form and dried it for about 4 weeks before firing. The wood form is a piece (about 40mm) of glue-pressed wood with material cut out with a router. The faceplate was fired in a kiln slightly above 1000°C for end stability (only drying the clay does not work). After firing the faceplate, I lacquered it with common spray lacquer (I didn’t use any glaze). The lettering is engraved with something like a small drilling machine and with high rotation. The white colour is the white clay under the lacquer. I tried drilling holes for the switches and pot in the faceplate both before and after firing and had different success, but actually I don’t know which method is better.

The chassis measures 435mm x 319mm x 142mm and consists of single aluminum plates mounted together simply with L-shaped aluminium profiles and screws. The thickness of the aluminium should be at least 2mm, and there should be enough holes in the chassis for cooling the components. Apart from that, many DIYers like to see the transformers and the big capacitors. I don’t, and therefore I constructed this chassis, where only the tubes are visible. The top plate extends over the tubes for protection. To mount the ceramic faceplate, I drilled little holes in the backside of the faceplate and glued little threaded bolts (threaded rods) into it. Then I mounted it with nuts to the aluminium chassis.


For testing, I would recommend measuring the voltages across the cathode resistors as they determine the current through the tubes (see the DC operating points shown figure 1). Also measure the heater voltage, the high power supply voltage and the voltages at the anodes. If all these voltages are OK, the amplifier should work properly. The start up procedure could be tested easily by hearing if the relays click at the right order in time. The proper function of relay Rly1 and Rly2 could be tested further by measuring the voltage right after the switch contact of Rly2 (before the rectifier). The slow turn-on of both the high voltage supply and the heater supply could be tested by measuring the voltages during the start up.

The voltage drop at the MOSFET Q2 (high voltage supply) should be in the range of 20-25V. When all the tube heaters are connected and warmed up, the output of the low voltage (heater) power supply is set to 12.6V with P1. When the power supply is turned on for the very first time, it is recommended to set the trim pot to the mid position and then adjust for 12.6V at the output. After connection of all the loads and after warm up, the voltage must be readjusted.

The heaters of V2 are connected in series across the 12.6V supply. If there are big differences in the heater resistances, it could be that one heater would run with a higher voltage. If each individual heater voltage does not exceed the specification (mostly ±10%), minor differences don´t matter. I have never had this problem, because normally I use pairs of tubes with the same “history,” and even during experimentation with different tubes (age, brand, etc) I had no problems. It could be, that even a tube from the same batch has a failure, and then the second tube would be influenced concerning tube performance or lifespan. Therefore, I recommend checking the heater voltages, when the tubes are turned on the first time.

Here are the output current measurements of the amp driving a 30-Ohm load and a 300-Ohm load:

30 Ohm resistance:
Maximum current 15.5 mA rms without clipping
Maximum current 32 mA rms with clipping

300 Ohm resistance:
Maximum current 10.5 mA rms without clipping
Maximum current 20 mA rms with clipping

I have compared this headphone amp to an old Technics receiver and to a hand-made solid state pre-amp built with a NE5534 opamp and a push-pull BD139/BD140 transistor output stage for driving headphones. The Technics receiver is the worst because of noise at the output and inferior audio quality. Mainly, there was no clear musical representation and therefore a very bad localisation of single instruments. My hand-made amps perform, in my opinion, very well. It’s hard to say why, but I like my tube amp more because of the overall tonal representation, especially during longer listening sessions. This is the reason why I presented my tube headphone amp and not my semiconductor amp. I’m using only my HD580 with this amp.

c. 2002 Helmut Ahammer.


A Precision Preamplifier-Power Amplifier System with Natural Crossfeed Processing.

by Jan Meier


“The better is the enemy of the good.”

The headphone amplifier with the natural crossfeed filter published on HeadWize in fact was my first DIY-project. I built this device because I was unsatisfied with the sound reproduction via the headphone-socket of my CD-player. But, as things go, I started to like constructing and decided to design and build some power amplifiers also. Having finished these, they sounded so good that I also made a new matching preamplifier with integrated headphone-amp.

Although the circuit of this new preamp basically is the same as that of the original headphone-amp, some modifications made it possible to increase the sound-quality substantially. The new preamp also has some more options as far as inputs and outputs are concerned. In this article I’ll briefly discuss each modification and leave it to the reader which modifications/options he wants to realize. For the basic preamp/headphone amplifier circuit, the reader is referred to the original article.

The matching 35W stereo power amplifier has 44 output stage opamps per channel and is not intended for a DIY novice. [Editor: the author also includes instructions for building a less ambitious 10W stereo amplifier.] In my opinion it really requires quite a lot of experience to build this amp properly. I had to drill/solder over 2000 holes/connections per amplifier. I made three of them, two for myself for biamping purposes and one for a friend. However, I have found the sound quality of the amplifier to be very rewarding. I was able to compare it with some very decent commercial amplifiers (DENON, LINN, NAIM), but these were completely outclassed by the new preamp-poweramp combo (an opinion shared by others).




Modifications to the original headphone amp circuit:

1. Breaking the ground loop
The preamp incorporates a ground-loop breaker. The ground-connection of the mains-socket is directly connected to the case of the preamp. It is connected to the ground-plane of the audio-circuit via a 4.7 ohm resistor in parallel with a 100 nF capacitor. This resistor prevents 50/60 Hz currents from flowing freely along the ground connections between the various audio components in a system, and thus eliminates the 50/60 Hz hum. Even if one component does not have a loop breaker, but all the other ones have, then there are also no ground loops and there is no problem. The 100 nF provides adequate RF-shielding. To prevent high voltages on the interconnect cables in case of a defective transformer, both inputs of the transformer are secured by a fuse (you never know which input is connected to neutral and which is connected to the alternating high potential).

The metal housing of the mains filter and the enclosure are both directly connected and are grounded to the mains. Normally the “signal ground” is also directly connected to this ground; however, in such situations a ground loop will occur if other equipment is connected to the preamp. By connecting the signal ground through a 4.7 ohm resistor, loop currents (and thereby hum) are greatly reduced. This implies that the preamp audio inputs and outputs MUST have floating grounds – their grounds cannot be directly connected to the enclosure.

2. Driving the opamps into class A operation
The output of each opamp is connected via a 1.5K ohm, 0.6 Watt resistor to one of the voltage-rails to drive the opamps into class A operation. At zero voltage output each output-stage now has to drive a 10 mA current and effectively works in class A. Only driving a low-impedance headphone at high volumes will result in the output stages leaving the class-A range.

By comparison, the output stage of a class B amplifier has two transistors that act like switches. One is opened to deliver the positive output currents, the other is opened to deliver the negative output currents. The switching behaviour going from positive to negative output currents (and vice versa) introduces distortion (for a very short moment the opamp is not able to “control” the signal) in the output that is readily heard (TIM-distortion).

With the output of the opamp connected via a resistor to one of the voltage rails, the DC output voltage will not change but one of the two output transistors will be opened to “dissipate” the current that flows through the resistor. As long as this current is higher then the current demand to drive the load, this output transistor will stay opened (and the other one will stay closed). There is no switching and therefore no distortion added.

This technique in principle does not limit voltage-swing, but it does limit the current swing. However, this should be no problem with my design. I enforce a DC output current of 10 mA. If higher currents are demanded by the circuitry (headphone) driven, the opamp will turn to class AB-operation. It is rather unlikely though that the preamp will need to output currents in excess of 10 mA, and if it does, sound levels will be so high that the distortion will not be heard. This modification resulted in a substantially improvement of sound quality, and can be easily added to the original design. Strongly recommended.

3. RF-shielding and prevention of oscillation
The + input of the first-stage opamps are connected to the potentiometer via two 1.5K ohm resistors. In the middle these two resistors are connected to ground by a 47 pF capacitor. Also 10 pF capacitors are added between the outputs and the inverting inputs of each opamp. These measures prevent high-frequency signals from entering the circuit and thereby increase stability and prevent high-frequency oscillations. I used polystryrol capacitors, but any other film-capacitors will also do.

4. Bass-enhancement circuit


I slightly modified the bass-enhancement circuit. The functionality has not changed, but now the feedback resistors are 10K ohms, and the outputs of the opamps are always connected by a 150 nF capacitor. This does not improve sound quality, but it does prevent annoying clicks when changing the settings of the bass-enhancement.

5. Decreased impedance of the potentiometer
Originally, a 50K ohm potentiometer was used. I found a lower impedance to sound marginally better – but only marginally. It is not worthwhile replacing a 50K ohms pot, if you already built the original circuit.



Additions to the original headphone amp circuit:

1. Inputs
There are now five inputs connected via a switch to the potentiometer. At the input jacks, the signal-pathways are connected to ground by 47K ohm resistors. This decreased the capacitive and inductive crosstalk between the various channels/inputs both audibly and measurably. Actually, I was rather surprised how much these resistors added to the sound quality.

2. Line out
For recording purposes, a non-volume-controlled output was added. The audio source can be chosen independently from the source being listened to. Note that there is no signal buffer and that it might be advantageous not to have these switches set to the same position, if a recording device is connected. Otherwise, the same source will be loaded by both the preamp and the recording device and cables.

3. Preamp out
A volume controlled output to drive a power amplifier. This output signal is not processed by the natural crossfeed filter.

4. Processor out
A volume controlled output to drive an amplifier (e.g., an electrostatic headphone amplifier). This audio-signal is processed by the natural crossfeed filter.

5. Headphone out
For connecting a dynamic headphone. The headphone jack I used has a built-in switch that disconnects the processor outputs, if a headphone is connected. It is made by Lumberg (part-number is KLBRSS 3 L) and can be ordered at Farnell in Germany (ordering number 838 550). The jack is directly mounted to the board.

6. Increased headphone output impedance
The headphone output impedance is normally near zero ohms. Optionally, the output impedance can be increased to 120 ohm by adding a resistor. Many headphones are designed to be connected to a source with a 120 ohm output impedance. Personally, I did not add these resistors to my preamp, but built a plug to connect preamp and headphone that has these resistors incorporated. My Sony headphones reacted very favorably to this increased impedance, whereas my Sennheiser HD600 became rather muddy. Simply try which suits your headphones/taste best. Since most dynamic headphones have a higher impedance at lower frequencies the increased output impedance results in an increased bass (with my Sony + 3dB!, Sennheiser + 1.5 dB).




The design

This all-opamp power amplifier has 44 output opamps per channel! Why 44 output opamps? My design goals were:

  • The output stage should be very fast.
  • The output stage should be linear, so the “control-opamps” would have an easy job.
  • The amplifier should be driven by a regulated power supply (unregulated supplies, as used in conventional amplifiers are IMHO a major source for a decrease in sound quality, because there is no infinite PSRR).

I wanted a completely regulated power supply for the output stage for currents up to 4 Amps. This implied using 4 pairs of LM317/LM337, since one voltage regulator only handles 1 Amp. I, therefore, would also need at least 4 pairs of output transistors per channel, since you can’t put voltage regulators in parallel to supply the same component with current. (There are voltage regulators that handle more than 1 Amp, but these are very expensive and require lots of heatsinking). So the choice was between [8 transistors + 4 opamps + heatsinks] or 44 opamps.

I also wanted to have a linear output stage with no distortion, which implied local feedback using one opamp per transistor-pair. For semi-class-A operation, 4 pairs of transistors allow for 4 different “switching” points (or more precisely, the output voltage where the opamp switches between the two transistors in the output stage). 44 Opamps allow for 44 different “switching points” (actually I only use 24 different points but this still is far better than 4). Each transistor-pair that is to be driven in class-AB dissipates heat and requires a heat sink. Opamps like the LM6171 don’t need a heatsink (unless you use the dual version at high currents).

Opamps are an ideal solution, but their current capabilities are too limited. I, therefore, placed 44 of them in parallel. To drive them in pure class A would demand a high DC-current (per opamp) and increase power dissipation. I decided to inject only a relatively small DC-current, so each opamp works in class AB.

By using different current values for the various opamps, each opamp will switch at a different overall current demand. At any time of operation, the major part of the output opamps will be in a non-switching state, and the “control-opamp” (which is working in class-A) is able to control the output signal continously. TIM-distortion is eliminated, although class-AB functionality is used. In contrast to a conventional class AB device, where there is no control during switching, there always will be “control” using many parallel output stages, each switching at different points. That’s why I called it semi-class A.

I wanted biasing currents for the output stage opamps between approximately 1.5 and 5 mA – not too high to cause excessive current drain and power dissipation, and not too low to start switching at very low sound levels. So RP resistors should be approximately within the range 3K ohms to 12K ohms. Then I simply selected values that were available in the catalogue. No sophisticated calculations.

When you look at the costs, I don’t think that the transistor solution is much cheaper than the all-opamp solution. High quality, high speed transistors as well as decent heatsinks do cost. Of course the second solution is more elaborate, but the way is our goal, so time is for free.



There are no special new construction techniques. This is not a project for novices. The enclosure shown is a real nice part but unfortunately also very expensive ($150). Like the preamp, the audio input and output jacks MUST have floating grounds, because of the loop-breaker circuit in the power supply.

The opamps that have the highest power dissipation in “standby” are the input opamps. They have to dissipate a current of 18V/1.5K ohms = 12 mA. Power dissipation is 216 mW. This is fully acceptable. Power dissipation of the output stage opamps is much lower, except when driving large signals into low loads. However, such power demands are transient. Power handling is one of the reasons I decided to use the LM6171 instead of its dual version LM6172. Note that 2 x 44 amps have quite a large total body surface so any heat is easily transferred to the air.

Originally, part of the amp got quite hot, not because of the heat dissipated in the opamps, but because of the heat dissipated in the voltage regulators. The total standby current for both channels is approximately 0.34 A. The power transformers are 2 x 18 VAC (50 Watts). With a voltage drop across the positive voltage regulators of 25V – 18V = 7V, power dissipation becomes 2.4 Watts. The negative regulators have to dissipate an equal amount of heat, so total power dissipation in the voltage regulators comes up to approximately 5 Watts. This is easily handled by the heatsink I made out of a aluminum sheeting. Even at high sound levels, the output voltage of the regulators does not drop below 23V, which means that the regulators can still do their jobs most adequately.

The amplifier uses 2 power transformers, not to have a completely independent supply for both channels (because they aren’t), but because I use a very slim enclosure and one big transformer would not fit. Each transformer “drives” 4 positive and 4 negative voltage regulators (4 pairs). Each voltage regulator pair, consisting of a positive and a negative voltage regulator, “drives” a group of 11 output stage opamps (44 opamps total). One additional pair of positive and negative voltage regulators (after a thorough LC-filtering) powers the the four input stage opamps of both channels. There are separate fuses for both transformers. The values of the fuses shown in the schematic are for 230VAC mains. For 110VAC, a value of 800 mA would be more appropriate.

A simpler 10W amplifier

The maximum output voltage of the amplifier is approximately 16V and the maximum current is about 6.6 Amps. To build a “smaller” power amplifier, reduce the number of output stage opamps to limit the current capability of the output stage. There will be a point where the amplifier will not be able to deliver the 16V. It all depends on the impedance of the speaker. For example, using 20 opamps will limit the maximum output current to approximately 20 x 0.15 = 3 Amps. With a 4 ohm loudspeaker, the maximum voltage is 12 V and maximum music power is 0.5 * I * I * R = 18 Watts per channel.

Which output stage opamps should be removed to reduce the output power? I would take the ones with the higher impedances to the power rails first, since this would drive the output stage for a longer period of time in pure class A. However, I think the sonic difference will be small, if some of the other opamps were removed.


The maximum power of the amplifier is primarily set by the supply voltage (18V). The maximum output voltage is approximately 16V. With 8 ohm speakers, the power rating is 16 Watts per channel. With 4 ohm speakers, the power rating doubles (32W per channel). Continuous power output equals the peak power output since, except for the supply voltage, power supply is “over-dimensioned”. Each LM6171 opamp is able to deliver up to 150 mA of current, we have 6.6 Amps per channel. With 16V output, the amp should drive loudspeakers down to 2.5 ohm. In this case, the output impedance effectively seen by each separate opamp is 44 x 2.5 + 10 = 120 ohms and does not represent a major problem (LM6171 is specified for impedances down to 50 ohms).

The noise of preamp and power amp measured at the loudspeaker connections with the volume at maximum was heardly noticable (no hum due to the regulated power supply) and unmeasurable for my multimeter (less then 0.1 mV!). SNR thus by far exceeds the specifications of the CD and is estimated to be better than 120 dB.

The “Phase” switch can be used to configure the both channels of the amplifier for biamping or can convert the amplifier into a monoblock with double the output power. It has three positions:

  • Position 1: each channel is driven by its own input buffer. This is the normal stereo mode.
  • Position 2: each channel is driven by one and the same input buffer. This can be used for biamping when both channels drive different units of the same loudspeaker. (Alternatively, one can connect the same output of the preamp to both inputs, but this solution saves cable and was given for free by the phase-switch.
  • Position 3: Each channel is driven by the same input buffer but the phase of one of the output channels is reversed. Connecting one single loudspeaker to the positive terminals of both channels (instead to one positive and one negative (ground) terminal) doubles the signal amplitude. This option is specially designed for high impedance, low efficiency loudspeakers. Advantage: Maximum output power per loudspeaker has increased by a factor 4 (approximately 64 Watts at 8 ohms instead of 16 Watts). Disadvantage: The stereo amp is converted to a mono amp (double costs).

Last week I listened to some real loud music. Due to the Analoguer filter (described in another article on HeadWize), I am simply able to sustain much louder sound levels now. I have used the amps with various loudspeakers (CHORD, KEF, QUAD, etc.) and although some of these speakers are quite hard to drive, the amp did not seem to have any problems with them. Especially the excellent bass-control was one of the first characteristics noticed by most listeners.


I am offering a DIY-kit for the headphone amplifier (NOT the preamplifier) with the updates discussed in this article. The completed headphone amp is shown above. If you are interested in the kit, please e-mail me.

I strongly recommend experimenting with these designs. As always, have fun!


10/11/2000: Corrected ground-loop breaker section in power supply schematics for preamp and power amp. For the ground-loop breaker to work properly, the circuit ground must be isolated from the metal enclosure, which is connected to the mains ground.

11/6/2000: Repositioned 10pF feedback capacitor around IC2 in preamp for greater stability. Also added pictures of headphone amplifier kit and updated picture of preamp-power amp combo at beginning of article.

c. 2002 Jan Meier.

Setting Sound System Level Controls.

by Dennis Bohn

2018 Note: Click here for the newer version on Rane site.


  • Decibel: Audio Workhorse
  • Dynamic Range: What’s Enough?
  • Headroom: Maximizing
  • Console/Mic Preamp Gain Settings
  • Outboard Gear I/O Level Controls
  • Power Amplifier Sensitivity Controls
  • Active Crossover Output Attenuators



Correctly setting a sound system’s gain structure is one of the most important contributors to creating an excellent sounding system. Conversely, an improperly set gain structure is one of the leading contributors to bad sounding systems. The cost of the system is secondary to proper setup. The most expensive system set wrong never performs up to the level of a correctly set inexpensive system. Setting all the various level controls is not difficult; however, it remains a very misunderstood topic.

The key to setting level controls lies in the simple understanding of what you are trying to do. A few minutes spent in mastering this concept makes most set-ups intuitive. A little common sense goes a long way in gain setting.

A dozen possible procedures exist for correctly setting the gain structure of any system. What follows is but one of these, and is meant to demonstrate the principles involved. Once you master the fundamental principles, you will know what to do when confronted with different system configurations.


Audio-speak is full of jargon, but none so pervasive as the decibel. Mastering gain, or level control settings also requires an understanding of dynamic range and headroom.

Dynamic range is the ratio of the loudest (undistorted) signal to that of the quietest (discernible) signal in a piece of equipment or a complete system, expressed in decibels (dB). For signal processing equipment, the maximum output signal is ultimately restricted by the size of the power supplies, i.e., it cannot swing more voltage than is available. While the minimum output signal is determined by the noise floor of the unit, i.e., it cannot put out a discernible signal smaller than the noise (generally speaking). Professional-grade analog signal processing equipment can output maximum levels of +26 dBu, with the best noise floors being down around -94 dBu. This gives a maximum unit dynamic range of 120 dB – a pretty impressive number coinciding nicely with the 120 dB dynamic range of normal human hearing (from just audible to painfully loud).

For sound systems, the maximum loudness level is what is achievable before acoustic feedback, or system squeal begins. While the minimum level is determined by the overall background noise. It is significant that the audio equipment noise is usually swamped by the HVAC (heating, ventilating & air conditioning) plus audience noise. Typical minimum noise levels are 35-45 dB SPL (sound pressure level), with typical loudest sounds being in the 100-105 dB SPL area. (Sounds louder than this start being very uncomfortable, causing audience complaints.) This yields a typical useable system dynamic range on the order of only 55-70 dB – quite different than unit dynamic ranges.

Note that the dynamic range of the system is largely out of your hands. The lower limit is set by the HVAC and audience noise, while the upper end is determined by the comfort level of the audience. As seen above, this useable dynamic range only averages about 65 dB. Anything more doesn’t hurt, but it doesn’t help either.

Headroom is the ratio of the largest undistorted signal possible through a unit or system, to that of the average signal level. For example, if the average level is +4 dBu and the largest level is +26 dBu, then there is 22 dB of headroom.

Since you cannot do anything about the system dynamic range, your job actually becomes easier. All you need worry about is maximizing unit headroom. Fine. But, how much is enough?

An examination of all audio signals reveals music as being the most dynamic (big surprise) with a crest factor of 4-10. Crest factor is the term used to represent the ratio of the peak (crest) value to the rms (root mean square – think average) value of a waveform. For example, a sine wave has a crest factor of 1.4 (or 3 dB), since the peak value equals 1.414 times the rms value.

Music’s wide crest factor of 4-10 translates into 12-20 dB. This means that musical peaks occur 12-20 dB higher than the “average” value. This is why headroom is so important. You need 12-20 dB of headroom in each unit to avoid clipping.


After all equipment is hooked-up, verify system operation by sending an audio signal through it. Do this first before trying to set any gain/level controls. This is to make sure all wiring has been done correctly, that there are no bad cables, and that there is no audible hum or buzz being picked up by improperly grounded interconnections (See Sound System Interconnection). Once you are sure the system is operating quietly and correctly, then you are ready to proceed.

  • Turn down all power amplifier level/sensitivity controls.
  • Turn off all power amplifiers. (This allows you to set the maximum signal level through the system without making yourself and others stark raving mad.)
  • Position all gain/level controls to their off or minimum settings.
  • Defeat all dynamic controllers such as compressors/limiters, gate/expanders, and enhancers by setting the Ratio controls to 1:1, and/or turning the Threshold controls way up (or down for gate/expanders).
  • Leave all equalization until after correctly setting the gain structure.


A detailed discussion of how to run a mixing console lies outside the range of this Note, but a few observations are relevant. Think about the typical mixer signal path. At its most basic, each input channel consists of a mic stage, some EQ, routing assign switches and level controls, along with a channel master fader. All of these input channels are then mixed together to form various outputs, each with its own level control or fader. To set the proper mixer gain structure, you want to maximize the overall S/N (signal-to-noise) ratio. Now think about that a little: because of the physics behind analog electronics, each stage contributes noise as the signal travels through it. (Digital is a bit different and is left to another Note and another day.) Therefore each stage works to degrade the overall signal-to-noise ratio. Here’s the important part: The amount of noise contributed by each stage is (relatively) independent of the signal level passing through it. So, the bigger the input signal, the better the output S/N ratio (in general).

The rule here is to take as much gain as necessary to bring the signal up to the desired average level, say, +4 dBu, as soon as possible. If you need 60 dB of gain to bring up a mic input, you don’t want to do it with 20 dB here, and 20 dB there, and 20 dB some other place! You want to do it all at once at the input mic stage. For most applications, the entire system S/N (more or less) gets fixed at the mic stage. Therefore set it for as much gain as possible without excessive clipping. Note the wording excessive clipping. A little clipping is not audible in the overall scheme of things. Test the source for its expected maximum input level. This means, one at a time, having the singers sing, and the players play, as loud as they expect to sing/play during the performance. Or, if the source is recorded, or off-the-air, turn it up as loud as ever expected. Set the input mic gain trim so the mic OL (overload) light just occasionally flickers. This is as much gain as can be taken with this stage. Any more and it will clip all the time; any less and you are hurting your best possible S/N.

Once you’ve set all the input gains, and then created the overall desired mix (involving all sorts of art and science I’m not going to get into), then you must set the output level controls in a similar manner: advance the output control until the output OL light begins to flicker. This is the maximum output level.

(Note that a simple single mic preamp is set up in the same manner as a whole mixing console.)


All outboard unit level controls (except active crossovers – see below) exist primarily for two reasons:

  • They provide the flexibility to operate with all signal sizes. If the input signal is too small, a gain control brings it up to the desired average level, and if the signal is too large, an attenuator reduces it back to the desired average.
  • Level controls for equalizers: the need to provide make-up gain in the case where significant cutting of the signal makes it too small, or the opposite case, where a lot of boosting makes the overall signal too large, requiring attenuation.

Many outboard units operate at “unity gain,” and do not have any level controls – what comes in (magnitude-wise) is what comes out. For a perfect system, all outboard gear would operate in a unity gain fashion. It is the main console’s (or preamp’s) job to add whatever gain is required to all input signals. After that, all outboard compressors, limiters, equalizers, enhancers, effects, or what-have-you need not provide gain beyond that required to offset the amplification or attenuation the box provides.

With that said, you can now move ahead with setting whatever level controls do exist in the system.

Whether the system contains one piece of outboard gear, or a dozen, gains are all set the same way. Again, the rule is to maximize the S/N through each piece of equipment, thereby maximizing the S/N of the whole system. And that means setting things such that your maximum system signal goes straight through every box without clipping. Here’s how:

Choose between one of the three methods OL LightOscilloscope, or AC Voltmeter described below. With the console or preamp set up as above, you now need a convenient sound source. Use an oscillator (built-in or external) and feed in a tone around 1 kHz. (Personally, I hate the sound of 1 kHz, so I prefer something lower, say, around 400 Hz.) Or you can substitute pink noise for the “OL Light” or “Oscilloscope” methods, but pink noise will not work for the “AC Voltmeter” method (the ACVM will not response fast enough to catch the peaks).

  • OL Light: Set the level of the oscillator (NOT the console or preamp’s output level; it has already been set!) by turning up it’s own control, if existing, or by using a spare channel on the console. Turn it up such that the output OL indicator just begins to light. It’s a large signal, on the order of +20 dBu, and would be very loud if you had not already turned off the amps. What you have now is the maximum expected signal level running through the system. From here on, everything will be set so it does not clip with this signal. Once done, the operator can run the system as loud as they want without fear of (feedback or) distortion.
    Oscilloscope: Using the OL light is a fast and convenient way to set this level. However, a better alternative is to use an oscilloscope and actually measure the output to see where excessive clipping really begins. This method gets around the many different ways that OL points are detected and displayed by manufacturers. There is no standard for OL detection. If you want the absolute largest signal possible before real clipping, you must use an oscilloscope. And, of course, if the unit or console does not have an OL indicator, then an oscilloscope is mandatory to establish the actually clipping point. (For a really clever alternative to an oscilloscope see “Piezo Magic,” The Syn-Aud-Con Newsletter, Vol. 24, No. 2, Spring, 1996.)
  • AC Voltmeter: If an oscilloscope is out of the question, another alternative is to use an AC voltmeter (preferably with a “dB” scale). Here, instead of relying on the OL indicator to tell you when you have a maximum signal, you choose a very large output level, say, +20 dBu (7.75 Vrms) and define that as your maximum level. Now set everything to not clip at this level. This is a reasonable and accurate way to do it, but is it an appropriate maximum? Well, you already know (from the above discussion) that you need 12-20 dB of headroom above your average signal. It is normal pro audio practice to set your average level at +4 dBu (which, incidentally, registers as “0 dB” on a true VU meter). And since all high quality pro audio equipment can handle +20 dBu in and out, then this value becomes a safe maximum level for setting gains, giving you 16 dB of headroom – plenty for most systems.

Outboard gear falls into three categories regarding gain/level controls:

  • No controls
  • One control, either Input or Output
  • Both Input & Output Controls

Obviously, the first category is not a problem!

If there is only one level control, regardless of its location, set it to give you the maximum output level either by observing the OL light, or the oscilloscope, or by setting an output level of +20 dBu as shown on your AC voltmeter.

With two controls it is very important to set the Input control first. Do this by turning up the Output control just enough to observe the signal. Set the Input control to barely light the OL indicator, then back it down a hair, or set it just below clipping using your oscilloscope. Now set the Output control also to just light the OL indicator, or just at clipping using the scope. (Note: there is no good way to optimally set an input control on a unit with two level controls, using only an AC voltmeter.)

For Rane digital audio products, like the RPM 26 DSP Multiprocessor where input A/D (analog-to-digital) metering is provided with the RaneWare software, setting the input level gain is particularly easy and extremely important: Using the maximum system signal as the input, open up the Input Trim box and simply slide the control until the 0 dBFS indicator begins lighting. This indicates the onset of “digital clipping,” and is definitely something you want to avoid, so this is the maximum gain point.


If your system uses active crossovers, for the moment, set all the crossover output level controls to maximum.

Much confusion surrounds power amplifier controls.

First, let’s establish that power amplifier “level/volume/gain” controls are input sensitivity controls. (no matter how they are calibrated.) They are not power controls. They have absolutely nothing to do with output power. They are sensitivity controls, i.e., these controls determine exactly what input level will cause the amplifier to produce full power. Or, if you prefer, they determine just how sensitive the amplifier is. For example, they might be set such that an input level of +4 dBu causes full power, or such that an input level of +20 dBu causes full power, or whatever-input-level-your-system-may-require, causes full power.

They do NOT change the available output power. They only change the required input level to produce full output power. (Okay. I feel better.)

Clearly understanding the above, makes setting these controls elementary. You want the maximum system signal to cause full power; therefore set the amplifier controls to give full power with your maximum input signal using the following procedure:

1. Turn the sensitivity controls all the way down (least sensitive; fully CCW; off).

2. Make sure the device driving the amp is delivering max (unclipped) signal.

3. Warn everyone you are about to make a LOT of noise!

4. Cover your ears and turn on the first power amplifier.

5. Slowly rotate the control until clipping just begins.

Stop! This is the maximum possible power output using the maximum system input signal. In general, if there is never a bigger input signal, this setting guarantees the amplifier cannot clip. (Note: if this much power causes the loudspeaker to “bottom out” or distort in any manner, then you have a mismatch between your amplifier and your loudspeaker. Matching loudspeakers and amplifiers is another subject beyond this note.)

6. Repeat the above process for each power amplifier.

7. Turn the test signal off.


Setting the output attenuators on active crossovers differs from other outboard gear in that they serve a different purpose. These attenuators allow setting different output levels to each driver to correct for efficiency differences. This means that the same voltage applied to different drivers results in different loudness levels. This is the loudspeaker sensitivity specification, usually stated as so many dB SPL at a distance of one meter, when driven with one watt. Ergo, you want to set these controls for equal maximum loudness in each driver section. Try this approach:

1. Turn down all the crossover outputs except for the lowest frequency band, typically labeled “Low-Out.” (Set one channel at a time for stereo systems.)

2. If available, use pink noise as a source for these settings; otherwise use a frequency tone that falls mid-band for each section. Turn up the source until you verify the console is putting out the maximum system signal level (somewhere around the console clipping point.) Using an SPL meter (Important: turn off all weighting filters; the SPL meter must have a flat response mode) turn down this one output level control until the maximum desired loudness level is reached, typically around 100-105 dB SPL. Very loud, but not harmful. (1-2 hours is the Permissible Noise Exposure allowed by the U.S. Dept. of Labor Noise Regulations for 100-105 dB SPL, A-weighted levels.)

Okay. You have established that with this maximum system signal this driver will not exceed your desired maximum loudness level (at the location picked for measurement). Now, do the same for the other output sections as follows:

1. Mute this output section – do not turn down the level control; you just set it!
If a Mute button is not provided on the crossover, disconnect the cable going to the power amp.

2. Turn up the next output section: either “High-Out” for 2-way systems, or “Mid-Out” for 3-way systems, until the same maximum loudness level is reached. Stop and mute this output.

3. Continue this procedure until all output level controls are set.

4. Un-mute all sections, and turn off the test source.

Congratulations! You have finished correctly setting the gain structure for your system.

Now you are ready to adjust equalization and set all dynamic controllers. Remember, after EQ-ing to always reset the EQ level controls for unity gain as required. Use the Bypass (or Engage) pushbuttons to “A/B” between equalized and un-equalized sound, adjusting the overall level controls as required for equal loudness in both positions.


Optimum performance requires correctly setting the gain structure of sound systems. It makes the difference between excellent sounding systems and mediocre ones. The proper method begins by taking all necessary gain in the console, or preamp. All outboard units operate with unity gain, and are set to pass the maximum system signal without clipping. The power amplifier sensitivity controls are set for a level appropriate to pass the maximum system signal without excessive clipping. Lastly, active crossover output controls are set to correct for loudspeaker efficiency differences.

Information contained in this work has been obtained by Rane Corporation from sources believed to be reliable. However, neither Rane nor its author guarantees the accuracy or completeness of any information published herein and neither Rane nor its author shall be responsible for any errors, omissions, or damages arising out of use of this information. This work is made available with the understanding that Rane and its author are supplying information but are not attempting to render engineering or other professional services. If such services are required, the assistance of an appropriate professional should be sought.

1. Murray, John & Pat Brown, “A Gain Structure Guide,” LIVE SOUND! International, pp. 18-24, Mar/Apr 1997. Thanks to John and Pat for inspiration and some content for this RaneNote.

Setting Sound System Level Controls PDF version.

2018 Note: Click here for the newer version on Rane site.

c. 1997, Rane Corporation.
From Rane Corporation site. Republished with permission.