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

by Jan Meier

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“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).

THE PREAMPLIFIER-HEADPHONE AMPLIFIER

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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

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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.

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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 POWER AMPLIFIER

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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.

Construction

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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.

Operation

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.

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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!

Addendum

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.

The Lindesberg Portable Headphone Amplifier With Crossfeed.

by Toni Kemhagen

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I thought that my portable headphone amplifier with crossfeed would be an interesting project for DIYers. My intention when I started the project was to build a “state of the art” amplifier, with sound quality being the first priority, but still be portable and have relatively long battery life. I am very pleased with the result. The only disadvantage is that the amp is a little larger and heavier than would be optimal for a portable due to the fact that it uses four 9V batteries. The completed amplifier is still very small – it measures just 80mm x 150mm (3.2″ x 6″).

The headphone amp is a 2-stage design with Analog Devices opamps. The are two completely separate ±9V battery supplies, one for the input buffers and voltage amplifiers and the other for the output stage only. I added a crossfeed filter designed by a Swedish engineer named Ingvar Ohman. The level of crossfeed in the Ohman filter is higher, for example, than in Jan Meier’s crossfeed filter, but has better effect.

The Headphone Amplifier Circuit

Different opamps are more or less sensitive to impedance mismatching. This headphone amp design corrects for opamp input errors and distortion due to impedance mismatching, so that both the + and – inputs of the AD823 opamps see the same impedance. For example, in the “System Applications Guide” from Analog Devices, page 8-71, there is a distortion test with the OP275 opamp that shows how a mismatch of 10K ohms/910 ohms at the inputs can triple the device’s output THD+N. Also see Walt Jung’s article, Minimizing Input Errors, in Electronics Design, December 18, 1998. [Editor: And see, Designing an Opamp Headphone Amplifier.]

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The headphone amplifier has 2 stages. The Analog Devices AD823 opamp in the first stage works as an input buffer with a gain of 2. The 2.49K resistor compensates for the feedback impedance. The compensation network (0.1uF + 1.87K + 100K) at the – input of the opamp balances the impedance of the input network (with the pot rotation set at the usual position for listening).

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The input compensation network is a compromise. With the 0.1uF capacitor, there is still a slight mismatch in the 30Hz to 500Hz range. The optimal would be 1.0uF which would match the impedance over the entire frequency range. I use 0.1uF because there is limited space in the enclosure and because it is most important that the impedances are matched at DC and upper frequencies. Otherwise, if the enclosure had enough space, I would use a 1.0uF capacitor. Also, the impedances are matched with the crossfeed on. When the crossfeed is off, there is a slight mismatch, but I always listen with the crossfeed on.

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I do not know how much the AD823 is sensitive to impedance missmatching. I have not tried using the AD823 without the compensation, only with it. In my project, I have taken every practical step to minimize distortion.

The second and third stage (AD823 and AD812 opamps) is a composite amplifier with a gain of five. The inputs of the AD823 in the composite amp are impedance matched to compensate for the feedback network and the crossfeed network. The local feedback at the AD823 in the second stage sets the open-loop bandwidth to about 100kHz. The advantages of a composite amplifier are:

  1. No thermal coupling from the output stage to the controling input amp.
  2. The possibility of having separate power supplies to the AD823 and the AD812 opamps, so that the AD823s can get the cleanest power. Also, I think this amplifer is much less sensitive to the high impedance of the batteries.
  3. The ability to combine the JFET-input of the AD823 and the current feedback operation of the AD812.

Again, see Walt Jung’s article, Minimizing Input Errors, in Electronics Design, December 18, 1998. [Editor: And see, Designing an Opamp Headphone Amplifier.]

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The headphone amplifier has a total gain of 10. If you want to modify the total gain to 5, for example, the best way is to change the gain of the first stage to 1.

The Ohman Crossfeed Filter
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This crossfeed filter that I use is a design by a Swedish audio engineer named Ingvar Ohman. The design was published in an article called “Den Lilla Stereo-kontrollboxen SP12” in the December 1994 issue of the magazine “Musik och Ljudteknik” (“Music and Audio Technical Society”). The “SP12” in the title is nothing particular – only numbers and letters to give the project a name. In the article, Ohman is tried to be a little funny and explained that SP12 could stand for “Stereo Processor According to a 12-Year-Old Idea.”

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The graph above shows the frequency response of the Ohman filter with the right channel as the main signal and the left channel as the crossfeed. The filter has an input impedance of 210K ohms to 16K ohms from DC to beyond the audio range. The output impedance varies from about 50K ohms (DC) to 3.5K ohms (high frequencies). Because the filter has a high output impedance and because the input and output impedances change so much with frequency, the filter should be isolated between two buffers. There is no voltage drop – a mono signal goes straight through without loss.

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The maximum delay between the main signal and crossfeed is about 250uS at frequencies less than 100Hz. Ohman said that it is more important to have a correct frequency response for the crossfeed, and less important to have time delay. The 250uS that separate the right and left channels after crossfeeding correspond to 8.5cm of travel in air. For a person with ears that are 17cm apart (the average), the time delay exactly corresponds to a listening angle of 30 degrees for loudspeakers (which is the stereo standard).

Though the crossfeeding circuit may seem to make low frequencies almost mono, it is time-coding at low frequencies that is totally optimal and what is heard in reality. Low frequencies arrive at both ears with practically identical volume, not only the in-front sound, but also sound incoming at 30 degrees, which corresponds to about 250uS delay.

Ohman tried to add time delay to the higher frequencies, but did not like the sound (similar to a commercial crossfeed product that had full-spectrum time delay) due to the serious comb filter effects on mono signals. In order to compensate for too little time delay, extra channel separation in mid frequences is added – a trick to fool the human ears to believe that the time delay exists.

I think Ohman developed the filter for the recording engineers that have to use headphones. It matches most people’s hearing. He made many calculations and measurements and tests on many people. It has a larger amount of crossfeed than, for example, the filter by Jan Meier. Anyone who finds the Meier filter to be too subtle should try this circuit. The result is alot more depth in the audio image! I find that the Ohman crossfeed circuit is especially good with classical music, because it gives more depth to the audio image. Even if you do not build the portable headphone amplifier, please try this crossfeed circuit.

The Power Supply
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The output current buffers have their own ±9VDC power supplies, total separate from the ±9VDC power supplies of the voltage amplifiers and input buffers. The separate supplies mean that the input buffers and the voltage amplifiers that control the output signal have a cleaner power source. I use rechargeable NiMH 9V batteries (150maH capacity). You can use alkaline batteries and get longer battery life, but they will be rather expensive of course.

The AD823 has a low quiescent current of 2.6mA and a minimum power supply of ±1.5V. The AD812 idles at 4mA, has a short circuit current of 100mA and a minimum supply voltage of ±1.2V. There are a total of four AD823 opamps, so the battery supply for these devices will last about 150mAH / (4 * 3mA) = 12.5 hours. The battery life for the two AD812 opamps will be 150mAH / (2 * 4mA) = 18.75 hours in the idle state. I use a 4PDT switch to turn the power off – one pole for each battery.

The AD812 power supply will drain faster when driving headphones, depending on the headphone efficiency, headphone impedance and the listening volume. I use a Sennheiser HD600 (300 ohms, 97db/mW) and the batteries go out at approximately the same time. But with 30ohms cans, maybe the batteries that go to AD812 drain faster. A 30-ohm headphone will drain ten times more current than a 300-ohm headphone (with the same sensitivity).

I have checked the battery life more exactly. This data is for normal listening volume. The data is calculated from current measurements:

AD812 batteries AD823 batteries
300 ohms (HD600) 18 hours 12 hours
60 ohms (HD570) 16 hours 12 hours
30 ohms 14 hours 12 hours

In general, the batteries to AD823s drain a little faster than batteries to the AD812s, but the battery life for the AD812 batteries depends on the volume level and type of music being played:

Classical (low volume) Rock (high volume)
300 ohms (HD600) 19 hours 16 hours
60 ohms (HD570) 18.5 hours 10 hours approx.
30 ohms 18 hours 7 hours approx.

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I am using a “wall-charger” to charge the NiMH 9V batteries, which takes about 12 hours. The schematic above shows a recharger circuit for the headphone amplifier that equalizes the battery life when the amp is turned off and will fit in the amp’s enclosure. (I have not built this circuit yet.) Most people do not listen as long as 12 hours at a time. On average, someone may listen for a few hours and then turn the amp off. During the time between the listenings, this circuit will equalize battery levels, which will be ready for the next listening session with the same charge status.

For example, if the batteries to the AD823s have drained faster, then the AD812 batteries will charge the AD823 batteries through the 2 x 68 ohm resistors. And after a few hours, the charge status for all of the batteries will be equal. If all of the 9V batteries are at the same charge, it is simple to recharge all batteries at the same time using the AC charger. If you recharge with low current (aproximately 15mA), the charging time will be about 14 hours.

Construction

I mostly get my parts from a distributor here in Sweden: ELFA. They have most of things DIYers need, and you do not have to be a company to buy from them. But they do not have the AD823 and AD812 opamps, and the output inductors. The capacitors in the signal path are Icel 1% polypropylenes. The power supply capacitors are Panasonic FC Electrolytic and Wima polyester MKS2. The resistors are all 0.6W, but I think 1/4W resistors will work too. The volume control is a Bourns model 91, plastic type, 10K log taper (ELFA order number 64-256-07) and costs 142,5 SKR (approx. $15 US). The 4PDT switch for the power supply (ELFA order number 35-203-01) costs 99,25 SKR (approx. $11 US).

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The output inductor can be the Miller 4608. I wound the 3.9mH inductor myself, because at the time, I could not find anywhere to buy one. Here is how to wind the inductor:

  1. Put a little grease on a 5.5mm drill bit.
  2. Shrink a small length of shrink tubing on the drill bit (at least 17mm when shrunk).
  3. Wind 31 turns of 0.4mm diameter, enamelled copper wire around the tubing – tightly and the every turn tightly together. Keep the length of the coil to 14.5mm.
  4. When done, hold both wire ends in left hand and apply a few drips super glue along the length of the coil. Hold it a few seconds until the glue gets hard.
  5. Slide the shrink tubing off the drill bit and trim to about 16.5mm.

The final specs of the inductor are:

  • inner coil diameter: 6.7mm
  • outside coil diameter: 7.5mm
  • coil length: 14.5mm
  • 31 turns of 0.4mm enamelled wire

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The headphone amp has a separate ground plane, because it is easy to ground all components. On the downside, it is not so easy to modify the circuit, because you must desolder all of the ground connections before the ground plane can be removed. The enclosure is grounded at the backplate via a 10ohm resistor connected to the ground card. The 10 ohm resistor isolates any RF interference that the enclosure picks up from the air.

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The size of the circuit card is 76.5mm x 100mm. The size of the ground card is 75mm x 100mm. The ground plane is a standard copper pc board. After that I decided where all the holes should be, I held the main pc board and the ground plane together and drilled through both boards at the same time. Then I separated the boards and soldered all the parts to the main board first and then to the ground plane.

If a component has leads that are too short, you can solder a longer lead to the component to be able to solder it to the ground plane. It is hard to solder onto the large area of copper foil on the ground plane, because it dissipates the heat of the soldering iron quickly. It is good practice to mill a bit around each solder point to reduce the heat dissipation.

I do NOT recommend the ground plane card for the beginning DIYer. The better way to have a ground plane may be with a double-sided pc card. If I build another amp, I will not use a separate ground card. In this case, I will try to make an ordinary star ground on the card with the components.

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The enclosure is the Bobla “Alubos” model ABPH 800-0150 and measures 80mm x 150mm x 32mm (3.2″ x 6″ x 1.28″). The box and the front and back plates are sold separately at ELFA:

  • Box: ELFA order number 50-792-49. Cost: 212,5 SKR (approx. $23 US)
  • Front and back plates: ELFA order number 50-793-06. Cost: 73SKR (approx. $8) each.

The total cost is $39 ($23 + $8 + $8) – rather expensive but very nice! I find this case beautiful, and the four 9V batteries fit perfectly. It has a rubber rings around the front and backplates, so the case will not scratch the table.

The Results

The finished amplifier weighs 500g mostly due to the batteries. I have only used the amp with my home stereo and do not need a portable amp at this moment. When it is used as a portable amp, I do not think it is too heavy to carry, if the listener is interested in high quality audio.

I have tried the amp with the Sennheiser HD600 (300 ohms) and the HD570 (60 ohms), and the amp works great with them. The sound quality is very good. I have not been able to find any distortion. You can hear a very small amount of noise, if you turn the volume up to maximum with no music playing. I do not think that is a problem.

The audio image of the Ohman filter has very good depth. It has more crossfeed than Jan Meier’s filter. With the Meier filter, I still have an “in-head” feeling, and I do not feel satisfied until I have enough crossfeed. But, with the Ohman filter, the ambience is reduced. On classical recordings, I get a feeling of depth in the audio image that outweighs the reduced ambience.

The Ohman filter works best when the recording has wide stereo imaging and lots of ambience. At first, when I listen to some recordings with very narrow stereo image, I think it sounds a little dull. But when I have been listening for a while and get the feeling for the better audio image and can hear the instruments in their more proper place, I prefer the Ohman crossfeed in the long run. In other words, it is nice to move the ambience that you hear in an extreme stereo image to its proper position. You then can get a feeling of the original ambience in the audio image. It is also more relaxing and comfortable to listen with Ohman crossfeed.

It works on pop/rock music also. But in general, in pop/rock music recordings, there is no natural audio image from the place where the recording took place, as there is in a classical recording, when you can hear the ambience from the room. So, if there is very little stereo separation in a pop/rock recording, I do not use the crossfeed. For me, the lack of ambience in the recording demands a little wider stereo image.

But with most pop/rock recordings that have a very wide stereo image – with channels separated in each ear, this crossfeed is perfect!

Well, those are my thoughts, but of course, it is a matter of personal taste.

c. 2001 Toni Kemhagen.

Addendum

1/27/01: Gus Wanner has prepared an Microsoft Excel application for simulating the operation of the Ohman crossfeed filter, so that DIYers can test the effects of component value changes on the filter’s frequency response, time characteristics and channel separation. Wanner provides simulations for 3 levels of crossfeed (3dB, 6dB and 10dB). He writes:

The attached Excel 5.0/95 spreadsheet is an electrical analysis of Ingvar Ohman’s crossfeed network as published in Toni Kemhagen’s article on construction of the Lindesberg Portable Headphone Amplifier With Crossfeed. The spreadsheet includes plots of left and right channel output levels, channel separation, interchannel delay time, network input impedance and network output impedance. I have also made included entries for source resistance and load resistance.

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This crossfeed network is interesting in that it is the first one I have seen which deals with the problem of excessive high frequency channel separation when listening to normal stereo recordings with headphones. Since the optimum value of interchannel crossfeed is different for different recordings, I have included additional network analysis sheets for 3dB, 6dB, and 10dB low frequency channel separation. The shape of the channel separation versus frequency characteristic for these alternative networks follows (as closely as available component values permit) that of the original Ohman network specified in the referenced article. Since the crossfeed resistor (R4) is the same for all networks, a simple 2 pole 5 position switch would provide the ability to switch between the different networks for different degrees of crossfeed.

I appreciate the need to implement source impedance matching for opamps which are sensitive to source impedance when used in the non-inverting mode. Obviously, if source impedance is not constant (as with a gain control potentiometer at the input of an amplifier or, as here, with multiple crossfeed networks), source impedance matching becomes a major problem. A better solution would be to use opamps which are not as sensitive to this phenomenon. The manufacturer’s data sheets generally provide this information as part of their application notes.

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One note about these networks – as currently parameterized they are very sensitive to load resistance. It might be desirable to reparameterize the networks by dividing all resistances by 4 and multiplying all capacitance values by 4, thus lowering the output impedance by the same factor. If FET-input opamps are used, the existing values will not be a problem.

The worksheets are protected (without a password) to prevent accidental deletion of calculations. Input signal magnitudes, source and load resistances are entered on the “original” network page and are automatically carried over to the other pages. Note that these spreadsheets make use of the Excel complex number capabilities which are included in the “Analysis Package”. This comes with the Excel package but may need to be manually installed using the original Excel installation CD or floppies. Current editions of Quattro Pro also include complex number analysis capability and may be able to load this spreadsheet.

Download Gus Wanner’s MS Excel Simulator for the Ohman Filter

5/29/01: Gus has prepared a new version of his Ohman Filter Simulator. Computationally, it is the same as the earlier simulator, but includes a schematic for adding a variable Ohman crossfeed filter to the Landesberg (or any other amp). Thanks to David Richard Meddings for the submission.

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Download Gus Wanner’s MS Excel Simulator for the Ohman Filter

10/21/2001: The author presents an update to his amplifier design incorporating a sound image width control that operates separately from the Ohman crossfeed filter. He does not recommend modifications to the Ohman filter itself as a means of changing the width of the sound image.

  • It is wrong to change the setup of the Ohman filter when you want to tune the stereo image. The Ohman filter is made to mimic a normal loudspeaker setup, and the differences in people’s hearing normally deviate only few percentages from the Ohman filter response. Modifying the Ohman filter’s response is not an option because then the filter is not an Ohman filter anymore; it’s just an “effect-box” that does something strange that may sound good to the user. Only one setup, the original Ohman filter setup, is to be used as a “true headphone monitoring system.”

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My variable Ohman filter uses a stereo expander in the feedback path of the input buffer before the crossfeed processor, so that you will be able to tune the stereo image without changing the setup of the Ohman filter. Imagine that you are listening with loudspeakers and you listening to a recording with a narrow stereo image. When you put the loudspeakers farther apart, the time difference between the ears then has to increase. You see the same thing in the graphs below.

The filter takes the signal from one channel and puts it at the other channel’s negative input and, therefore, amplifies the differences between the channels, like a spatial expander. When the switch is set to “original,” you have the original Ohman crossfeed network working. When the switch is set to +1,+2, +3 and +4, both the stereo expander and the crossfeed are working, and the width of the audio image increases more and more. The widest sound image is at setting +4. When the switch is set to “stereo”, both the stereo expander and the crossfeed are disconnected, and the amp is in normal stereo mode. The switch is a 6-position, 2-pole shorting type – for example the ELMA typ01 (Elfa order no. 35-493-18).

I am very pleased with the result of this Variable Ohman crossfeed filter. I find it most useful to fine-tune the stereo image when the image is too narrow. It works with all types of music. But the most recordings donīt need to expanding, only if the sound image is too narrow or “boring”.

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Frequency Response of Variable Ohman Filter (Freq. vs. Volts)

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Time Response of Variable Ohman Filter

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Channel Separation of Variable Ohman Filter

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Frequency Response of Variable Ohman Filter (Freq. vs. dBV)

11/17/2002: Corrected polarity of electrolytic caps for negative power supply.

An Acoustic Simulator For Headphone Amplifiers.

by Chu Moy

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This article describes modifications to an acoustic simulation circuit for headphones that appeared in a magazine article called “Improved Headphone Listening” by Siegfried Linkwitz (Audio, December 1971). It is a simple RC-type filter that creates a more realistic sound image in headphones by electronically mimicking the “shaded” interchannel crossfeed of normal hearing. The circuit was based on a design published by Benjamin Bauer ten years earlier, but the Linkwitz version does not use inductors and is less sensitive to load impedance. (Crossfeed filters should not be confused with virtualizers, which use digital signal processing to simulate binaural or externalized 3D sound.) These modifications improve the sound of Linkwitz filter and optimize the circuit to work with headphone amplifiers.

I tested the modifications with my DIY pocket headphone amplifier (shown above), the Musical Fidelity X-Cans (v.1) headphone amplifier, Sennheiser 465 and Sennheiser HD600 headphones and the Stax SRS-3030 electrostatic headphone system. The source was a Panasonic SL-SX500 CD player. The filter can drive either a headphone amplifier or headphones directly with the headphone amplifier as the source. I tested the original filter directly connected to the HD465 phones and fed by the X-Cans. The modified filter was tested both ways: as the source to headphone amplifiers and directly connected to both headphones. Putting the filter before the amplifier eliminates any impedance interaction between the filter and headphones.

For more information about the original Linkwitz circuit, please refer to the Audio magazine article cited above. For more information about the Bauer circuit and acoustic simulators generally, see Technologies For Presentation of Surround Sound in Headphones and The Psychoacoustics of Headphone Listening. For information on commercial crossfeed filters, see A Quick Guide to Headphone Accessories.

Background

At work, I listen to music with headphones several hours each day. My system is a Musical Fidelity X-Cans amplifier and a pair of Sennheiser HD465 headphones (discounted 50% from a liquidation house!). The source material is either CD or FM radio. Although the audio reproduction is excellent, I cannot listen to this system for more than four hours without suffering headaches, caused by the super-stereo effect that is characteristic of headphone listening.

Stereo recordings are meant to be heard through loudspeakers. Headphones create a soundfield that is unnaturally spacious, in which some sounds seem to be crowded around each ear. A few months ago, I did some research into the problem. Headphone amplifiers from HeadRoom Corporation had a built-in crossfeed circuit to mitigate this unpleasant effect. However, I did not want to replace the X-Cans amplifier. Then I ran across a stereo crossfeed circuit in an old issue of Audio Magazine, which seemed to do something very similar to HeadRoom’s audio image processor.

I upgraded the design with higher quality parts, which I ordered from Digi-key Electronics and Mouser Electronics. The circuit was assembled on a 2″ x 1.75″ piece of printed circuit protoboard and put in a PacTec case from Radio Shack (RS 270-211). The front panel had two mini-stereo jacks (for the input and output), and a bypass toggle switch. (See A Pocket Headphone Amplifier for more information about purchasing parts.) The output of the X-Cans went to the input of the filter, and the headphones plugged into the output. After a month of listening with the Linkwitz circuit, I became dissatisfied with the sound quality. It did eliminate the super-stereo effect, but introduced other sonic problems.

First, the high frequencies were severely attenuated compared to the original signal, despite the built-in treble boost in each channel. The imbalance imparted a muffled effect to all kinds of music. With some vocalists (such as Barbra Streisand – yes, I am a fan), voices took on a “thick” quality. I suspected that the effect was caused by phase anomalies at the crossover point interacting with phasey artifacts in recordings.

Second, the level of crossfeed was a bit excessive for my tastes. The soundstage was pulled inward, away from the ears, but I felt as though I was sitting in the back of an auditorium with heavily padded walls. Flipping the bypass switch restored the spaciousness and reverb at the expense of the benefits of the crossfeed.

Third, the filter did not drive headphones well with a headphone amp input. The original circuit was designed to be connected to the output of a power amplifier. Running it from the speaker outputs of a 15W receiver produced ample volume. Headphone amps, on the other hand, huff and puff along at a relatively measly 100mW maximum output. A few Thevenin computations also revealed that the output impedance of the filter was slightly frequency dependent and varied from 61 to 73 ohms. My Sennheiser HD465s were rated at 60 ohms. These factors explained why I had to keep the X-Cans volume control at 75% rotation to achieve acceptable listening levels.

The Modifications

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Figure 1

The original circuit (figure 1) crossfeed frequencies below 700 Hz (figure 2). Linkwitz noted that the low frequency blending raised the bass response in each channel by about 3dB. He designed a 2dB treble boost to compensate partially for the increase, reasoning that full compensation was not needed because the low frequencies in each channel weren’t always in phase. On some recordings, the filter appeared to reduce the amount of deep bass due to phase cancellation. However, it also overemphasized the lower midrange and imparted a heaviness to the sound. Even though the separation between channels increased above 700 Hz, there was some high frequency attenuation – probably due both to phase effects at the crossover point and the fluctuating output impedance.

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Figure 2

The most obvious solution was to increase the treble boost, but that would not have affected the circuit’s high drive requirements or the apparent width of the soundfield. A voltage divider at the input (R1/R2) set the original attenuation factor at about 1:6. The crossfeed signal was summed into the output resistor (R2) of the divider. Increasing the value of the output resistor would increase the voltage output, but would also increase the output impedance and vary the level of crossfeed. Not a good idea.

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Figure 3

Lowering the input resistor (R1) would increase the output voltage, while keeping the crossfeed level constant (but having the effect of reducing percent of crossfeed in each channel, because a smaller R1 increases the level of the main signal). If crossfeed level remained constant, decreasing the value of R1 would “widen” the soundstage and create a smoother response with the existing treble boost. Also, a smaller R1 value would minimize output impedance fluctutation. After experimenting with several R1 values, I set R1a to 200 ohms (40% of the original value).

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Figure 4

Since all recordings are not the same, I added a “PERSPECTIVE” switch (S1) to customize the processing with an alternate R1 value. At R1b = 150 ohms, the low frequency separation between channels goes up to about 10dB and the overall output increases by about 2dB. Toggling from R1b to R1a, the soundstage appears to move further away (lower output, more narrow soundstage, slightly softened highs). After a period of listening, I still heard a slight thickness in the lower midrange, which I suspected was due to the threshold frequency of the treble boost being a bit low. Decreasing R3 to 910 ohms moved the threshold to about 800Hz and cleaned up the midrange emphasis. It also gave a touch more treble boost for a more balanced, clearer sound overall.

I settled on the R1b setting (low crossfeed) as the default, and use R1a when the sonic presentation would otherwise be too wide or if the recording is too bright. The PERSPECTIVE switch has a distinct effect with these R1 values, with the high crossfeed setting effectively simulating greater distance from the soundstage. On good acoustic recordings, the high crossfeed produces a palpable sense of depth in the headphone image.

Figure 3 shows the final schematic. The “Low-Z” version (where “Z” is short for impedance) is the most versatile. It is the version that I built and can drive headphones directly, because it has a low output impedance. The rest of this article mostly refers to the Low-Z version. If the simulator will be used ONLY as an input stage to a headphone amplifier, consider building the “High Z” version, which scales the resistor and capacitor values by a factor of 10 (x10 resistors and ÷10 capacitors) to get a higher input impedance (about 2000 ohms), which is a better match for the line outputs of preamps and other audio sources. The headphone amp itself should have an input impedance of 5K ohms or higher. Several DIYers have built the high-Z filter. See the addendum for details. Do NOT scale the parts if the simulator will ever drive headphones directly.

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Figure 5

A comparison of the original to modified signal levels (figure 4) shows that the modified output is about 3dB louder in each channel because of the new attenuation factor of 1:3. The low frequency separation is 6 dB, 3 dB wider than before. The overall output impedance is lower and is flatter over the audio range: 51 to 60 ohms. My calculations indicate that the high and low frequencies are 25% closer in level than the original. The improvement in sound is so great that I suspect the reduction of high frequency phase effects due to the wider separation also contributes significantly. The modification increases the crossfeed threshold frequency by about 8% and treble boost threshold frequency by 10%.

Construction

Assembling this acoustic simulator is fairly straightforward, and would be a good project for the intermediate DIY beginner. I used a printed-circuit protoboard from Radio Shack, which since has been discontinued. Instead, I recommend the Vector Circbord from Mouser Electronics (Stock No. 574-3677-6), which is tin-plated. If possible, layout the circuit as compactly as possible on a breadboard first to get an idea of where the components will go. The circuit is fairly simple, but the 16 capacitors and resistor can be a challenge to place.

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Then cut a small square (about 2″ x 1.75″) of the protoboard with a section of the foil pattern that best suits the breadboard layout. To cut out the board, a coping saw will work fine. If the DIYer is building my pocket amp project (which uses a similar-sized board), another good method is to score the board with a utility knife and break off the section needed. This method can result in some waste, unless the DIYer is building more than one project that will be installed in the same type of enclosure.

  1. With a ruler, draw, on the non-foil side, a line parallel to the long side of the Circbord, about 1.75 to 2 inches from the edge. Be sure that the selected section contains a useful foil pattern.
  2. Score the Circbord several times with a utility knife and ruler along the drawn line.
  3. Position the scored line of the Circbord (foil-side down) over a table with a sharp edge – the marked section should hang over the edge of the table.
  4. With one hand, press down on the Circbord against the table to anchor it.
  5. With the other hand, apply several sharp blows to the area of the Circbord overhanging the table. The section should snap, but still remain hinged because of the copper foil. Be careful not to pull off the foil.
  6. To separate the section, cut through the foil by scoring with the utility knife.
  7. Repeat this procedure on the Circbord section itself to get a board about 2 inches long.

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The 2.75″ x 4.6″ x 1″ enclosure is Pac Tec model K-RC24-9VB and comes in the colors Bone and Black. I purchased it at Radio Shack, and it is the same case I used for the pocket headphone amp project. It has a 9V battery compartment and both opaque and red plastic front panels. Radio Shack stores no longer stock the case, but it can be ordered from RadioShack.com (RS 910-1096, black only). Digikey and Mouser also sell these cases.

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The headphone jacks are enclosed units for 1/8″ stereo plugs. Radio Shack sells a version of these jacks (RS 274-249). I ordered higher quality units that have spring-loaded contacts from Mouser Electronics (Stock No. 161-3502). The DPDT switches are micro-mini toggles from Radio Shack (RS 275-626). It may be more convenient to mount the switches and jacks before wiring them, because some of the resistors may have to be wired behind the front panel and not on the protoboard, as space allows. I chose the red plastic panel because the opaque panel was too thick to mount the headphone jacks. Resistors that are mounted behind the front panel should have their leads insulated to prevent short circuits. I cut plastic tubes to length from the insulation on my hookup wire (24 ga.). Keep the wiring as short and neat as possible (which is not easy to do) to minimize RF pickup.

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In Linkwitz’s schematic, the original C2 is a 1.3uF non-polar electrolytic, but this value is difficult to find in a film capacitor. I used a 1.2uF film capacitor. Many DIYers like the sound of the simulator with the parts values shown, but others have written to tell me about their modifications to the circuit. I encourage anyone interested in building this simulator to experiment with parts values to get the best sound for that person’s hearing preferences and characteristics. The easiest way to customize the simulator is to build it first on an experimenter’s breadboard and then permanently solder the connections after the best sound is achieved.

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I strongly recommend auditioning the filter on a breadboard first, before finalizing the component values. The most popular modification is to decrease or increase the values of R1a/R1b in the range from 50 ohms to 330 ohms. Again, the lower the value of R1, the wider the soundstage will be. Another popular modification is to decrease C2 to 1uF, which can give a bit more depth to the soundstage due to the higher threshold frequency. The crossfeed threshold frequency is given by: Fcrossfeed = 1/[2*pi*C2*[R4 || (R5 + {R2 || R1})]]. The treble boost threshold frequency is approximately 1/[2*pi*C1*R3]. The amount of treble boost can be increased/decreased by decreasing/increasing R3 (to keep the threshold frequency constant, choose C1 so that R3*C1 = 0.00018).

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Download Gus Wanner’s Excel Application

For DIYers who want to customize the simulator but don’t have the time or patience to build many versions, Gus Wanner has prepared the above Microsoft Excel 2000 application, which can instantly plot changes in frequency response, time delays and other circuit characteristics as component values change. To use this spreadsheet, Excel must have installed the engineering functions in the analysis toolpak (which comes with the Excel package but is NOT automatically installed by the standard MS Excel install program). To install this toolpak, use the add-ins submenu of Excel. Verify that the correct toolpak is installed by clicking on Insert|Functions options and look for the engineering functions menu. Excel 97 should work the same way.

Mr. Wanner describes the application as follows:

The analysis is straight-forward, using Thevenin equivalent impedances and voltage sources as explained on the “analysis” tab. Note that Excel does NOT have any formatting capability for complex numbers (they display with maximum precision all the time, taking a huge amount of space). I have “cut them off” (for display only) by putting blanks or other values into adjacent columns.

The spreadsheet is parameterized for a medium and high crossfeed case designed for use with a low power amplifier (20 watts/ch or so) and a load of 70 ohms (the Sennheiser HD-25 impedance). The network sheets are protected (to prevent my accidentally wiping out formulas and values); there is no password, so simply unprotect the sheets (Tools|Unprotect) to change the values.

See the addendum for more details about this application.

The Results

The best sound quality was obtained with the Linkwitz filter driving the input of my pocket headphone amplifier or the Stax electrostatic amplifier. To optimize the 3D effect with supra-aural headphones, wear the earpieces as forward as possible, to enhance in-front imaging (slightly down and forward on the ears). This positioning helps direct the sound waves to enter the ears at an angle as happens with normal hearing, instead of going straight to the eardrums. The technique does work with circumaural headphones, but the effect is not as pronounced.

The modified Linkwitz filter sounds much more open and clearer than the original, especially when the crossfeed level was set to low (R1a). The highs were back! The bass was stronger and better defined. Instruments and vocals were focused and had “air” around them again (Barbra’s voice sparkled). Although the soundstage was wider than with the original filter on both low and high crossfeed settings, the crossfeed was still effective and subtlely pulled the image forward. Reverb no longer “bounced” off my ears the way that unprocessed headphone reverb does, and the bass was more centered like loudspeaker bass.

With the PERSPECTIVE switch set to high crossfeed, the soundstage narrowed and the top treble softened, yet there was more depth, more dimensionality – as though it had been moved further back. Recordings that polarized the stereo presentation with instruments or vocals to the extreme left and right had an substantially improved sense of aural continuity.

Sound Quality with Simulator Before Headphone Amp

The first set of extended listening tests were with the simulator as the source to the headphone amplifier and set to low crossfeed. In the HD465, there was just a trace of the emphasis in the lower midrange and a softening of the high end, which still made for a natural presentation. The HD465 is a very fine supra-aural headphone, but the looser ear coupling thins out the sound somewhat and highlights treble anomalies from the crossfeed. Regardless of these slight tonal shifts, the simulator’s effect still generated a very pleasant, forward soundstage.

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The HD600 has a more even frequency response than the HD465. The extended low end lends the sound a lushness. The effect of the simulator in the HD600 was so smooth that I could call it “ethereal.” Switching the filter on and off, there was only a hint of tonal impact in treble. It was almost seamless, like a video “morphing” transition. The HD600’s larger, more enveloping soundfield (due to the different ear coupling) created a vibrant, dimensional acoustic space that portrayed instruments and vocals with greater accuracy and tonal fidelity than the HD465.

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My new (actually used) Stax SRS-3030 headphone system arrived just in time to be included in the latest revision of this article. The system consists of the Lambda SR-303 electrostatic headphones and the SRM-313 amplifier. I quickly auditioned the Stax with the modified Linkwitz filter installed between the Panasonic CD player and the SRM-313 amplifier. With the simulator switched off, the SR-303 headphones presented a spatial, open soundfield without the “in-each-ear” sensation of other headphones, which I attribute to the oversized Stax transducers that sit at a slight angle on the ears. The sound was brighter than with the HD600s, not as lush. The Stax imaging seemed larger, but still two dimensional – essentially forming in a straight line between my ears.

With the modified Linkwitz filter switched in (low crossfeed setting), the image instantly took on depth and dimensionality. On recordings with a front vocal, instruments seemed to localize around and behind the voice. The treble softening was more noticeable than in the HD600 (though less noticeable than with the HD465) but did not appreciably alter the frequency balance of the SR-303s. The Staxes had tighter bass than the HD600. There was a touch fullness in the bass frequencies that conveyed the energy of the performances well. After a sustained period of auditioning with a variety of music from classical to pop, in every case I preferred leaving the filter on. The filter greatly enhanced the realism of the Stax presentation. Overall, the Stax/modified Linkwitz was a very successful pairing.

Sound Quality with Simulator After Headphone Amp

The sound quality from headphones directly connected to the Linkwitz filter varies, depending on the impedance response of the headphones. In theory, the filter should drive high impedance headphones more consistently than low impedance headphones, but that was not the result of my listening tests. Headphones are not the same as loudspeakers, and tend to have flatter impedance curves. So long as headphone impedance remains flat over the audible range or never falls below a threshold (say 10 times the filter’s output impedance), the headphone amplifier’s current output is the dominant factor in sound quality. Where headphone impedance falls below that threshold in the audible range, the Linkwitz circuit may introduce some coloration due to impedance fluctuation.

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The modified Linkwitz is much more efficient for directly driving headphones than the original filter. My X-Cans headphone amp could now comfortably drive the filter and the HD465 with the volume control at around 30% rotation. Substituting my pocket amp for the X-Cans was similarly gratifying. Since the HD600 has a higher impedance which should have made it better load for the Linkwitz filter, I expected it to sound more tonally consistent with the filter than the HD465. In fact, the HD465 had the better interfacing experience. Directly connected the Linkwitz, the HD600 had a distinctly dark sound. By comparison, the sound of the HD465 was similar to that when the filter was put before headphone amp. The lesson here is that there is no easy way to predict the success of the filter directly driving headphones. The best way to determine the sound quality in this configuration is to try the headphones with the filter.

Conclusion

The modified Linkwitz circuit sounds natural and spacious with a forward-projecting image. The filter’s effect is more subtle compared with the original design, and is, therefore, less intrusive into the musical experience. On many recordings, it adds an almost three-dimensional quality to the presentation. Headphone listening is now definitely less fatiguing. I can listen to my headphone systems for hours every day without headaches. If I do feel the need to take a headphone off, it is usually because my ears hurt from having the earpieces physically pressed against them for a long time.

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The filter works best as the input to a headphone amplifier and pairs well with my pocket headphone amplifier in the matching enclosure (shown above) for a very nice portable listening system. Both units are lightweight and will fit into many portable stereo carrying cases. The unused battery compartment in the acoustic simulator is good for storage of small accessories such as a headphone plug adapter. Finally, the price for the modified Linkwitz circuit is hard to beat, especially if one already has a headphone amplifier. The upgraded parts, case, switches and jacks came to about $20. A headphone amplifier with the modified Linkwitz circuit could be built for less than $40. If you build the Linkwitz filter with or without my modifications or have other mods you’d like to share, please don’t hesitate to e-mail me.

Thanks to Tyll Hertsens for his helpful comments.

c. 1998, 1999, 2000, 2001 Chu Moy.

Addendum

8/15/98: For better in-front localization with the Linkwitz filter, try wearing the headphones slightly forward and lower on the ears (supra-aural phones are the easiest to position this way). Experiment with the positioning to obtain the best localization. The goal is to get the sound to enter the ears at an angle, which is closer to the way normal hearing works. With the right recordings, this technique can produce a stunning sense of depth. It also works without crossfeed, but does not sound as natural.

10/6/98: Updated comparison of HeadRoom circuit to Linkwitz circuit. I also want to report that depending on the recording, the R1a setting of the perspective switch (“further from the soundstage”) can render a more 3D sound image, although the apparent width of the presentation would be fine without the additional crossfeed.

10/25/98: Added discussion about placing filter in front of headphone amp to eliminate any impedance interaction between the filter and headphones. Also clarified a few points throughout article.

11/16/98: Added image and description of portable headphone system. Also received report from user that the filter can drive Grado headphones directly with good results.

6/22/99: Added graph of time delays for the modified Linkwitz filter.

8/20/99: Mika V鳵r鄚iemi built the acoustic simulator and pocket amp in a single aluminum enclosure. He experimented with various values of C1 and R1 and found that C1 = 1uF and R1a = 50 ohms, R1b = 100 ohms had the widest soundstage and least effect on the high frequencies (Mika used the original R3 = 1000 ohms). “[B]efore I was positioned in the middle of band playing music. Now I’m in the front row as close as you can be…. Music just sounds realistic and that’s what I was looking for.” A more complete description of his work can be found in the DIY Workshop Forum.

8/26/99: Here is the parts layout and wiring diagram for Mika V鳵r鄚iemi’s simulator/amplifier project. Pictures of the finished amplifier can be see in A Pocket Headphone Amplifier.

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11/23/99: Added more guidelines for customizing the simulator. Also, Chester Simpson has created a version of the modified Linkwitz with scaled parts values for headphone amps with high input impedances (greater than 250K ohms). See his article A Soundfield Simulator for Stereo Headphones.

12/9/99: Siegfried Linkwitz (himself!) e-mailed me the equation for calculating the crossfeed threshold frequency, which I have added to the article. Check out his new website: Linkwitz Lab.

1/12/00: scrazy@gcn.net.tw built this version of the pocket amp, which has a 10K ohm volume control and an acoustic simulator front-end that is based on the circuit by Chester Simpson (see design by Fred Peng below). Full details can be found at DIY Zone (in chinese only). His system consists of a Rega Planet CD Player and Audio Technica ATH-f15 headphones.

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1/13/00: Fred Peng based his headphone amplifier on the acoustic simulator by Chester Simpson (which is based on the Linkwitz design), except that he replaced the R4,R6 combination in Simpson’s circuit with a 100K ohm resistor and added a unity gain input buffer stage made from an OPA134 and a high current output stage made from a PMI BUF-03 buffer. When compared with a McCormack Micro Headphone Drive, the BUF-03 driving his Grado HP-1 headphones with the simulator bypassed sounded better in the high and low frequencies than the McCormack, but the McCormack was better in the mid frequencies. With the simulator switched in, the sound was more relaxed, the low frequencies more centered, and the soundfield moved from inside his head to outside. He is very satisfied with the result and is planning to make another simulator for his Stax Lambda headphones. Full details and schematics (in chinese only) can be found at DIY Zone.

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1/28/00: Added figure 1a. Thanks to Siegfried Linkwitz for sending me the graph!

5/1/00: Gus Wanner has sent in a Microsoft Excel Spreadsheet application that analyzes changes to component values of the modified Linkwitz circuit (see the text above for instructions to download). He writes:

I enjoyed your article on your modification to the Linkwitz crossfeed network. Since I have a low power (20 watt/channel) high quality amplifier integral to my McIntosh C-40 audio control center, I wanted to develop a version of this network to work with the C-40 and my Sennheiser HD-25 phones (and other headphones with greater than 60 ohm impedance). The HD25s have a maximum input level of 200mW. With the HD25s at 70 ohms, this will require a voltage of approximately 3.7 volts across each channel or approximately 11.8 volts into the crossfeed network. This voltage level corresponds to an amplifier output of 17.5 watts into 8 ohms.

To aid in doing the design, I developed a complete analysis of the Linkwitz network using the complex number analysis capability built into the MS Excel spreadsheet (and I think also available in newer versions of Quattro Pro). The spreadsheet allows you to enter values for the various components, and immediately computes the resulting levels, channel separation, and delay times for frequencies from 20 – 20,000 Hertz. Graphical plots for these parameters versus frequency are included as well.

The component values on the spreadsheet are the final values for a crossfeed circuit I designed for use with my McIntosh C-40 and my Sennheiser HD-25 headphones. The circuit is relatively insensitive to load impedances 70 ohms or greater, so it would work with other headphones as well. The modified Linkwitz crossfeed filter works great with my Sennheiser HD25s.

3/14/01: Major rewrite of article. Added detailed comparison of sound quality of filter placed before and after the headphone amplifier and review of Stax SRS-3030 headphones with filter. Added new high resolution pictures.

3/14/01: Coffin Lin put his version of the pocket headphone amplifier (with a Linkwitz crossfeed front-end) in an old TV remote control case. He modified the filter by making R2 and R3 adjustable, instead of R1. The component values in his version of the filter are:

R1: 30K ohms
R2a, R2b: 15K, 10K
R3a, R3b: 50K, 100K
R5: 33K
C1: 3,300pF
R4: 33K
C2: 10,000pF

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The resistors are Dale RN55D. About making R2 and R3 adjustable, he says:

I mistook R2 for R1, but on the Excel worksheet simulator, R2 can still alter some balance. I think that adjusting R3 is more effective than adjusting R2 (I forget which switch is for what resistor.) One has more stereo (good for dance and rock) and the other is more natural (good for jazz).

3/25/01: Changed the value of R3 in figure 3 to 910 ohms (originally 1K-ohms) to remove emphasis in lower midrange and to increase the treble boost. This update results in a more balanced, clearer sound. I STRONGLY recommend it.

3/27/01: R2 and R3 incorrectly drawn in figure 3 from 3/25 update. Fixed.

11/24/2001: Mark D. Johnson writes:

I just finished building your Acoustic Simulator and have been auditioning it over the last several days on my Sennheiser HD600s (even now I’m listening to Miles Davis as I type this). I just want to tell you how much more I’m enjoying my music and how much less fatiguing it is to listen for long periods of time.

The thing I like best of all is the three dimensionality I hear in recordings that was never present before. As a drummer, the most amazing thing to me is that I can actually hear (whether true or not) the location of individual drums/cymbals being played – and not just sound coming from a point source called “drums.” On the latest Dianna Krall CD I could hear even elevation changes taking place as different cymbals were struck. Again, whether or not this is just a byproduct of the design I cannot be sure, but it sure makes listening more enjoyable.

5/21/2002: J. Ian Ramsey (from the forums) built a pocket amp and the high impedance version of the Linkwitz acoustic simulator in separate enclosures. He obtained most of the parts from RS Components Ltd. For the 120nF cap (C2), he paralleled two caps: 100nF and 22nF. He writes:

I made the Simulator in the high impedance version as recommended by Chu in the project notes. I have increased the gain of the amp from x11 to x17 by putting a 1.5K resistor in parallel with R3 (1K) to compensate for the insertion loss of the simulator which is only used between the source and the amp.

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I laid out the simple circuit for the acoustic simulator in stripboard. The simulator effect is very subtle and at first I was unsure if I had made mistakes in the layout, which were preventing the correct circuit action. Listening to this year’s Grammophone magazine Award winner – Vaughan Williams ‘A London Symphony’ confirms the following:

  • With the simulator switched to B the whole soundfield is very gently more centre focused and there is a slight loss of ambience.
  • Switched to A, the field narrows and the volume level drops slightly thus making the sound image appear to be heard from a greater distance – just as Chu remarked in the article.

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Overall I could easily live without the simulator as my HD600 headphones with the cmoy amp are very, very good. As an intellectual exercise, the concept of the simulator is satisfying in the way it addresses some of the headphone effects against speakers and this design does not degrade the sound – so I will most likely continue to use it.

Chu was spot on when he said to spend time with it in the design values as one would soon tire if the values were changed to produce a more dramatic effect. The subtlty soon gives way to a distinct change with it in and out and between mode A and B. Congratulations and thanks to Chu Moy for these brilliant designs.

5/21/2002: Phidaeaux (from the forums) built a pocket amp and the high impedance version of the Linkwitz acoustic simulator in a single enclosure. The white LED power indicator is mounted on the circuit board INSIDE the transparent Serpac enclosure. He writes:

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Ok, I just finished this bad boy, and man oh man, am I pleased! This thing sounds so good. I’m only driving 32-ohm phones (Sennheiser HD497s) but the difference in sound quality is VERY obvious over my SlimX MP3/CD player’s built-in amp. It feels good to have power to spare, instead of driving the stock amp to its limits. Also, the crossfeed is very nice. I didn’t really notice at first, but now that I can sit down and play with it, I really like how it sounds. smile

First off, thanks much to PRR, cmoy, tekir, tangent, and everyone who had either helpful suggestions, or had problems in the past, that I could read about and avoid doing myself. Cmoy and tangent’s info were both very useful too, for both the obvious reason (the schematics) but also the piles of helpful tweaks and pictures they had.

Technical notes! I’m using all the ‘usual suspect’ parts: Panasonic pot, Digikey jacks, metal film resistors, polypro caps (except for the power supply caps, which are electrolytic). I’m using a Burr-Brown OPA2134 dual op-amp (in a nice machined socket).

The crossfeed circuit is a pre-amplifier modified Linkwitz (the same circuit that cmoy uses as an independent item) and has had its resistor values all multiplied by 10, and its capacitor values divided by 10. This was done to raise input impedence a bunch, because its not actually driving headphones, it just connects directly to the amplification stage. The two ‘perspective’ settings are 2000 ohms, and 1500 ohms.

I thought about replacing one of those with a pair of mini pcb-mount multiturn pots, so I would have a ‘default’ setting and then a ‘custom’ setting I could adjust by opening the case. Or maybe put a ‘stereo’ pot controling those values, and cram it somewhere else on this case. Who knows. Anyway, I’m satisfied with those two values for now, I’ll wait before I do any ‘tinkering’. <smile>

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Anyway, to those who said it could not be done, I managed to fit the following panel items in the case: input and output jacks, one on each side of the case. I’ll get some right angle adapters I think to clean it up a bit, so it doesn’t have the plugs sticking off to the side like cowlicks. Power switch, crossfeed bypass switch, crossfeed perspective switch and Panasonic EVJ volume control (with pretty aluminum knob) are all on the front panel. The volume control, and the crossfeed switches are touching each other. They are quite literally pressed up against each other inside that panel!

White LED. Looks good shining out from inside the case. Bright too. I was going to use a blue, but the blues were less efficient than this white (oddly enough). I’m running it with a 2.2k resistor. Current through it seems to be about 3mA. That could stand to be lower, and the LED is very bright right now. If I make another modification, it’ll be to dim that LED a bit, and save some power. You can see neat little shadows inside the case from the components, light glinting off of things inside there. smile Very cool.

I also really like this Serpac H-65 transparent blue case. I’m all about form AND function. I know that people usually get on one side of the fence or the other. They either LOVE pretty little gadgets, regardless of how they actually work. And then there are people who say “Screw it, does it work right? Then it can be ugly, I don’t care.”

For me, I want it to look good, work well, and feel ‘right’. If I’m going to be using something every day, it needs to be ergonomically designed. Engineers are notorious for totally forgetting the fact that real people have to USE the things they build. I work with equipment that was NOT inuitively designed so often; its like a breath of fresh air to find equipment where someone actually sat down to use it for a while, and thought ‘huh, this knob should really go down here on this side, instead of up here… that would make a lot more sense.’

It’s so easy to figure out where controls should go. Everything flows left to right on mine. Left is the input, right is the output. Starting from the left on the panel, you can turn it on or off, then you can choose crossfeed on or off, then you can raise or lower the crossfeed, then you can change the final volume. I don’t need labels, cause it makes sense. And then, I love this case. Its curvy and sexy, but still has quite a bit of internal volume. The transparent blue doesn’t scream “I bought this at Radio Shack like a nerd!” but rather, “This is a modern piece of technological wizardry.” But at the same time, you can peer inside and see the parts, making a muted statement that it was a DIY project. The LED inside is nice too. Don’t have to use up a panel spot, and the inside of the device lights up all pretty. Form AND function. Together at last!

Anyway, I don’t mean to lecture you guys; you all build very good things. smile I got the ideas for this from seeing various other projects people have built. I just urge you all to take note of the stuff you deal with every day, and while you are thinking about all the technological aspects of it, to give a thought to intuitive design and control placement, the delicate art of making something ‘easy to use’ regardless of what its actually doing, etc.

Also, read the book “The Design of Everyday Things” by Don Norman. It talks about doors, VCRs, ovens, and all sorts of things you use every day, but don’t think much about. Ever push on a door when you are supposed to pull? Everyone has! But why? It’s a simple matter to make the operation of a door obvious; you don’t even need ‘push’ and ‘pull’ signs that people have to look at and read. A well designed door gets used right each time, without anyone even noticing. It’s all about intuitive engineering and human-centered design.

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Oh, more technical jibba-jabba. Gain on the amp is set to 11. Current measured while in use, at a moderate volume, is 12mA. Not too shabby. Voltage between ground and the rails is 4.40V and -4.41V while in operation. Not bad, if I do say so myself! This single 9V is plenty to drive the 32-ohm HD497 to utterly insane levels. No need to give it more voltage.

An Enhanced-Bass Natural Crossfeed Filter.

by Jan Meier

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[Editor: This article is a follow-up to A DIY Headphone Amplifier with Natural Crossfeed by Jan Meier and resulted from an e-mail discussion with Chu Moy – the author of An Acoustic Simulator for Headphone Amplifiers.]

If you have an amplifier with a mono-switch, then here is a little experiment: listen to a stereo recording (by headphone) in stereo mode, and then press the mono-button and watch the bass. If you hear the same way I do, then you will notice that the bass suddenly seems to have weakened – it has become less pronounced. The effect is similar to that what is heard with a crossfeed filter, only much stronger. Listening in mono does introduce cancellation of low frequencies, but there is also cancellation higher frequencies (which is generally is even stronger since, with normal stereo recordings, low frequencies are more in phase). With the crossfeed activated, a weak cancellation will only be present at low frequencies, but at all frequencies, the sum of the sound pressures at both eardrums always equals the sum of the pressures in stereo mode!

At first I also wondered about the apparent loss of bass, but actually, it is this unnatural, larger then life-size, uni-directional bass, that counts for most of the annoying effects of headphone listening. I know, the crossfeed sound is nothing for a bass-freak. One should not expect a punchy bass, only a relaxation of the sound. It’s like listening to loudspeakers – a balanced speaker does not jump at you at first hearing but is rather colourless/neutral/unobtrusive. The rewards come while listening for longer periods of time. A good speaker does not fatigue, and this exactly is the strength of the natural crossfeed filter.

To add bass or not to add bass….that is the question. I believe that most bass-losses are due to psychoacoustic effects, but after thinking it over more carefully, enhancing the bass response of the natural crossfeed filter could be legitimate, because headphone sound is optimized without using crossfeed. If there really is a psychoacoustic effect (a uni-directional bass is unnatural and I believe that, with headphones, this emphasizes its existence), then the effect has been (unconsciously) corrected for in the sonic design of the transducers.

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Figure 1

The crossfeed design by Siegfried Linkwitz (see An Acoustic Simulator for Headphone Amplifiers by Chu Moy) includes a bass boost to compensate for low frequency cancellation. Figure 1 is a graph of the frequency response of both the direct-signal and of a mono-signal that is given on both signals simultaneously. Responses were calculated for a 60 Ohm load (such as headphones) and for a very high output load (e.g., a headphone amplifier).

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Figure 2

As with the natural crossfeed filter, the direct signal with the Linkwitz filter shows a signal loss at lower frequencies, (-1.0 dB at 60 Ohms, -0.35 dB at 50k Ohms). However, more important is that a mono-signal at frequencies below 700 Hz is increased by up to 1.3 dB at a 60-Ohm load and up to 1.9 dB (!) at 50k Ohms. The delay times for the Linkwitz design (figure 2) are fairly natural, as the crossfeed signal has similar filter frequencies and thereby should have similar delay times as the natural crossover filter.

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Figure 3

I designed a modified version of the natural crossfeed filter that has a frequency response very similar to the Linkwitz filter. It can be found in figure 3. The crossfeed level is medium. It easily can be tested between a CD-player and headphone amplifier (there is no insertion loss). I guess it sounds very similar to the Linkwitz filter, but is a little bit easier to realize. I have never implemented this Linkwitz equivalent, being fully satisfied with the original natural crossfeed filter.

The Linkwitz equivalent can be substituted for the filter in my headphone amplifier design. The bass EQ switch (S1) was not intended to compensate for any apparent loss of bass due to the crossfeed. It simply should compensate for the natural roll-off of the transducers. Such filtering is nothing new. With my headphones, I use position “3.” In this position, only the very lowest frequencies are amplified. Even with the Linkwitz equivalent, it could be very nice to keep the bass extension switch. Since it can be switched off, it will not hurt and the extra work/costs are little. It is, however, a matter of personal taste. Hi-fi purists do not like unnecessary equalization and might want to leave it out.

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Figure 4

With the Linkwitz equivalent filter, it is not possible to set the crossfeed level with a 2 x 6 switch. Not only do the resistors and capacitors in the two outer networks (Z1) have to be switched, but also the central capacitor to ground. A possible option is to use a 3 x 4 switch (see figure 4) and leave the first two crossfeed positions out (they have a very weak effect, and I never use them).

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Figure 5

To customize the time delay and amplitude profiles of the Linkwitz equivalent, download the circuit simulation spreadsheets below (in Quattro Pro 1 and 3 formats). Change the values B4..C9 at the first page according to figure 5 and the corresponding lines in the pictures will change. Similarly you can change the values in D4..E9 for a direct comparison of the different options.

Download circuit simulation spreadsheet (in Quattro Pro 1 and 3 formats)
Download circuit simulation spreadsheet (in Quattro Pro 1 and 3 formats)

(Please remove the .xls extension on the files above.)

Yesterday, I did some listening with popular music with heavy bass using the original natural crossfeed filter. The bass in these recordings was more “centered”. As expected, I could not/hardly notice any specific loss of bass. I present the Linkwitz equivalent design as an example of how the various filters can be tested to personal taste, by putting them between a CD player and a power/headphone amplifier, before eventually building the headphone amp project.

Addendum

6/22/99: Added figure 2.

7/26/99: Added figure 5 and instructions for using spreadsheet circuit simulator.

5/4/00: Jasmin Levallois built this version of the Pocket Headphone Amplifier (see article by Chu Moy). It features an input gain stage, the Meier enhanced-bass natural crossfeed filter and an output buffer. He writes:

Finally I got some free time to complete my project…. I got a lot of work to do for school during the last few weeks and I didn’t have time to work on my amp. This weekend I decided to take one day to transfer the amp from the breadboard to the pc board. I used about the same circuit as Jeff Medin. The input stage has a gain of 10, the output stage is a voltage follower, and in the middle I put the Meier bass-enhanced crossfeed circuit.

I used 2 OPA2132 opamps, but if I had to do it again I would use 2 OPA2134. An OPA2132 costs $6.99 while an OPA2134 costs $2.67. Since there is almost no audible difference between both opamps, I would go with the OPA2134 to save money. Since the second stage has no voltage gain, I decided to omit the capacitor in front of the output stage. I also removed the resistor in front of the output stage, and I don’t hear any noise from the output stage. The only noise I can hear, sometimes, is coming from my CD player.

As you’ll see on the photos, the inside of my amp is very messy, but, hey, its my first electronic project. Fortunately, even if it’s messy inside, the outside looks pretty good. I really like this Serpac Enclosure (Digikey part no. SRH65-9VB-ND); it looks ways better than the PacTec case.

The photo of the battery compartment is to show that the Serpac enclosure has a 9v Battery compartment with battery contacts. It’s easier to remove the battery with that kind of battery compartment than the PacTec Enclosure. Also the Serpac enclosure is just about the same size as the Pactec enclosure except that it’s a bit longer, and the height is a little bit less. This might be a problem for the electrolytic capacitors. I would recommend the Philips ones with this enclosure rather than the Panasonic Z series because the Philips electrolytic caps are much smaller.


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Download parts list for Levallois Amplifier (MS Excel format)

5/6/00: Gus Wanner developed a low impedance version of the enhanced-bass filter to drive his Sennheiser HD600 headphones directly from a power amplifier, and has prepared a MS-Excel application to model the enhanced-bass filter (both low- and high-impedance versions). DIYers can change component values and instantly see the effects of their changes on the filter’s frequency response, time delay profiles, etc. Wanner writes:

Compared to my Sennheiser HD25s, the HD600s have a lower sensitivity (the HD25s produce about 105 db SPL at 1mW into 70 ohms, while the HD600s produce about 97 db SPL with 1mW into 300 ohms), a higher impedance, and a maximum input level of 200mW. With the HD600s, 200mW requires about 7.75 volts across each phone. I measured the impedance magnitude versus frequency for both the HD25 and HD600 headphones. The measurements were made using an audio oscillator in series with a 300 ohm 1% metal film resistor. By measuring the voltage across each headphone and the voltage across the 300 ohm resistor, it is possible to compute the impedance.

I have attached a spreadsheet which provides my data and the impedance plots. It is interesting to note that both headphones have an impedance peak around 100 Hertz; in the case of the HD600s this peak has a magnitude of almost 600 ohms! I also looked at both the current and voltage waveforms on a scope; for both phones the waveforms are in phase for the entire audio range. Slight phase shift can be seen at 20 Hz and at 20,000 Hz, but I would say that the impedance of both of these phones is predominantly resistive.

To handle the HD600 inefficiency, I decided to look at Jan Meier’s crossfeed networks. Jan’s networks have the advantage of low insertion loss, but tend to be sensitive to both source and load impedances. I created a new spreadsheet with an analysis of Jan’s enhanced bass crossfeed filter, using his original component values (designed for use within his headphone amplifier) and a low impedance version designed for use with a nominal 300 ohm load impedance.

This spreadsheet is attached hereto as well; you can see a performance comparison between my final low impedance compenent values and Jan’s original. Because of the large impedance variation of the HD600s at low frequencies, I incorporated their impedance versus frequency into the design as wel. I did not assume a constant headphone load impedance in my analysis of the network, but actually put in my measured data of impedance versus frequency to see how the response of the network would be affected by the Senn HD600 peak around 80 Hertz. For other headphones, you can substitute the impedance magnitude versus frequency (if you have it) into the appropriate column on the spreadsheet. Alternatively, you could substitute the manufacturer’s nominal impedance value (e.g., 120 ohms or whatever) at each of the test frequencies.

Components for my crossfeed networks were purchased from Digikey – the capacitors are Panasonic ECQ-E(F) series metalized polyester film types which are available in values up to 10 microfarads at 100 WVDC at reasonable prices. Non-polarized electrolytics are NOT recommended for this application. The resistors are Ohmite TA Series “Power Chip” ™ thick film on an alumina substrate 5 Watt rating. These resistors have basically no inductance (50 nanohenries at 1 MHz!). Non-inductive wirewound resistors could also be used, however standard wirewound resistors are not recommended due to their inductance.

With this network, the HD600s have plenty of volume with my 20 watt/channel monitor amp and sound wonderful – the closest thing to electrostatic speaker sound I have heard! Again, note that the low impedance version of Jan Meier’s network is more sensitive to load impedance than the low impedance version of the modified Linkwitz network. With the HD600s, to get 200mW requires about 24.5 volts into each channel of the modified Linkwitz crossfeed network, which corresponds to an amplifier output of about 75 watts into 8 ohms! I did, in fact, hook up the modified Linkwitz crossfeed network to my 150 watt/channel power amplifier, and the Hd600s sounded great. But I wanted to use the HD600s with my McIntosh C40, so I had to look for another solution.

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Download Gus Wanner’s MS Excel modeler for the Meier enhanced-bass filter 1
Download Gus Wanner’s MS Excel modeler for the Meier enhanced-bass filter 2

c. 1999, 2000 Jan Meier.
The author’s website: Meier Audio.

A DIY Headphone Amplifier With Natural Crossfeed.

by Jan Meier

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For me, listening to music is a very private affair. Throughout my life, my musical tastes always have been somewhat different from that of the other people I lived with. Consequently, I learned to appreciate good quality “cans.” This appreciation has increased since, as a teenager and as a student, I never had the financial means to buy loudspeakers that could stand up to the sound quality offered by my Sennheiser and Beyerdynamic headphones.

Nonetheless, listening was not all heaven on earth. The in-head localization phenomenon did not please me. With recordings presenting a wide soundstage, some instruments are heard in one of the two audio channels only. This is most annoying, like a bee buzzing in one’s ear.

Some 15 years ago I experimented with electronic crossfeed by bridging the left and right outputs of the headphone channel of my amplifier with resistors. Although the crossfeed thus produced cured the “buzzing bees,” the sound became extremely dry and I dropped the idea.

In the last few years, a number of systems have appeared on the hi-fi market that also produce crossfeed, but in a much more refined way. The analogue headphone amplifiers of HeadRoom and the digital systems of Sony (VIP 1000), Sennheiser (Lucas), and AKG (Hearo 777) are well-known examples. These systems all more-or-less cure the problem of the in-head localization experienced with headphones.

These systems work by simulating the mechanisms that a person uses to locate and externalize sources of sound:

First, the sound of a source to the right side of the listener (e.g., the right loudspeaker in a stereo setup) not only reaches the right ear, but attenuated and delayed, is also heard by the left ear. The level of attenuation and the delay time of this crossfeed signal provide important directional information.

Second, the soundwaves are partly absorbed and partly reflected by the listener’s head. Especially, the reflections at the ear pinnae interfere with the soundwaves that directly enter the ear canal and amplify or attenuate specific frequency components. As these reflections depend on the direction of the soundwave, the “color” of the sound changes with the direction of the source.

Third, reflections of soundwaves on the walls of the listening room produce reverberation that conveys an extra feeling of space.

The information obtained by these mechanisms is further refined by movements of the head. Changes in sound levels, delay times and sound color refine our sense of direction. For a demonstration blindfold a friend and ask him to locate a ticking clock that you have hidden in the room. He will start turning his head, although he can’t see it. With his head in a fixed position, he will find an exact localization much more difficult.

All these mechanisms are missing when we hear music using headphones. The transducers are directly coupled to our ears. The sound of the right (left) transducer will not reach the left (right) ear and the reflections on the oracles have changed and hardly interfere with the original soundwave. Moreover, the sound-sources are attached to our head, so head movements no longer add information. Reverberation is not present.

In principle, digital sound processors can simulate the mechanisms described, but the results are, thus far, not fully satisfactory, because the reflections on the pinnae are very complex and listener-specific.

Fortunately, the mean directional information is provided by the differences between what we hear by our two ears. A “natural” crossfeed from the right (left) audio signal to the left (right) transducer, with an appropriate attenuation and delay, will reduce most of the adverse symptoms of headphone listening considerably.

A straightforward approach to mimic crossfeed is to take the original stereo signal, attenuate its amplitude and have it delayed. Then cross the two channels and add the processed signals to the original stereo signal. In a mathematical formula:

Vleft,out(t) = Vleft,in(t) + .Vright,in(t-t0a<1

Vright,out(t) = Vright,in(t) + .Vleft,in(t-t0)

The HeadRoom systems work like that (with a being a frequency dependent parameter). For a more detailed information just take a look at their most enjoyable homepage (http://www.headphone.com). The people at HeadRoom are very fine engineers, but have not revealed the schematics of their circuitry. So when I decided to build my own headphone-amplifier, I had to design an appropriate crossfeed-filter myself.

The crucial part in the crossfeed-filter is the realization of the required time delay of approximately 300 ms. Although standard solutions for signal delay can be found in many text books on electronics, a fixed frequency-independent delay with headphones has one major drawback: the so-called Comb-effect.

A conventional crossfeed filter, such as that realized by HeadRoom, mimics the sound of a left or a right sound source most adequately, but the frequency-spectrum of a source in front of the listener is unnecessarily disturbed. For this in-front source, the left and right audio signals are equal: a mono signal. In principle, these signals need no crossfeed. However, with conventional solutions, there still is, and the audio signals at the headphone-transducers become:

Vleft,out(t) = Vright,out(t) = Vleft,in(t) + a .Vleft,in(t-t0a <1

Especially in the high frequency range, the delayed crossfeed signal interferes with the original input and attenuates specific frequencies. The frequency-curve is no longer flat but shows a larger number of dips (the Comb-filter effect). HeadRoom compensates for the overall attenuation with a filter that gently lifts the higher frequencies, but the dips in the frequency-curve do not disappear.

With a fixed delay, the Comb effect can not be eliminated, so I decided to make the delay of my crossfeed filter frequency dependent. For localization of a sound source, the delays of the frequencies below 2 kHz are the most important, and therefore should have a natural 300 ms delay. For higher frequencies the delay is reduced. Moreover, by also slightly shifting the original audio signals in the other direction and by giving them a small, frequency dependent attenuation before the crossover signals are added, the mono signals will be left undisturbed! A music signal that simultaneously is found on both channels is left unchanged, and a signal on only one input channel is partly transferred to the other channel, with an appropriate time delay at low frequencies.

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Figure 1

Take a look at the basic crossfeed circuit shown in figure 1. The crossfeed is performed by only three resistors and two capacitors! It’s hard to believe, but the circuit really does the job!

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Figure 2

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Figure 3

For those interested in the technical details, figures 2 and 3 show the amplitudes and the time delays of both the crossfeed-signal as well as the (direct) audio signal. The amplitude of the crossfeed-signal decreases with frequency, and thus mimics the shadowing effect of the head at higher frequencies. In the lower frequency-range, the time delay between the crossfeed and the direct signals is 320 ms, and thus mimics the natural delay for a loudspeaker seen at an angle of approximately 30 degree by the listener. By choosing different parameter values for resistances and capacities the crossover signal can be “tuned”, but with the values shown it works well for my ears with 95% of all my recordings.

SOME MATHEMATICAL STUFF

With a conventional crossfeed filter, the direct signal equals the original input signal, and the crossfeed signal is realized by attenuation and delay:

Vinput(t) = cos(2pft)

Vdirect = Vinput(t) = cos(2pft)

Vcrossover = a(f).cos(2pf(t-tdelay))

a(f) << 1

With a mono signal, both direct signals are equal and both crossfeed signals are equal. The output signals become:

Vout(t) = Vdirect + Vcrossover = Vinput(t) + a(f).cos(2pf(t-tdelay))

The second term interferes with the input/direct signal and results in a number of dips in the frequency spectrum at those frequencies where: 2pftdelay = (2n+1)p. This is the so-called Comb-effect.

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Figure 4

With the “natural crossfeed” filter, the crossfeed signal is also realized by attenuation and delay, but now the direct signal is also (slightly) attenuated and slightly time-shifted in the other direction (figure 4):

Vinput(t) = cos(2pft)

Vdirect = A(f).cos(2pf(t+tdirect))

Vcrossover = a(f).cos(2pf(t-tdelay))

A(f) = sin(2pftdelay)/(sin(2pf(tdirect+tdelay)) » < 1

a(f) = sin(2pftdirect)/(sin(2pf(tdirect+tdelay)) << 1

tdirect << tdelay

2pf(tdirect+tdelay) < p/2

(This last condition guarantees that a(f) and A(f) will change monotonically with the frequency.)

The result is that, with a mono signal, the sum of the direct and the crossfeed signals equals the original input signal and there is no Comb-effect:

Vout(t) = A(f) cos(2pf(t+tdirect)) + a(f).cos(2pf(t-tdelay)) = cos(2pft) = Vin(t)

The condition 2pf(tdirect+tdelay) < p/2 requires that the effective delay (tdirect+tdelay) be shortened for higher frequencies. Natural delay times of 300 ms only can be realized for lower frequencies.

THE NATURAL CROSSFEED NETWORK

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Figure 5

The basic network is shown in the figure 5. It contains a chain of three passive networks, of which the outer two have the same impedance value. It easily can be shown that:

Vleft,out(t) = (Z1 + Z2)/(2*Z1 + Z2) * Vleft,in(t) + Z1/(2*Z1 + Z2) * Vright,in(t)

Vright,out(t) = (Z1 + Z2)/(2*Z1 + Z2) * Vright,in(t) + Z1/(2*Z1 + Z2) * Vleft,in(t)

The crossfeed signals are given by the last terms of these two equations.

Using

Z1 = R1 // C1 = R1 /(1+iwR1C1)

Z2 = R2

the transfer function of the crossover becomes:

Z1/(2*Z1 + Z2) = (R1/(2*R1 + R2))/(1+iwC1R1R2/(2R1 + R2))

This is the transfer function of a first order low-pass Bessel filter with a filter frequency of:

f = (2R1 + R2)/(2pC1R1R2)

Using:

R1 = 1000 Ohm
C1 = 470 nF
R2 = 2200 Ohm

the filter frequency becomes f = 650 Hz.

To derive the time delay of the crossover signal:

f (phase shift of the transformation) = arctan(C1R1R2/(2R1 + R2)) and
t (time shift) = /(2pf) = f/w

For low frequencies:

f = wC1R1R2/(2R1 + R2)
t(w~0) = C1R1R2/(2R1 + R2)

For high frequencies:

f = p/2
t = 1/4f

To derive the time shift of the direct signal:

Amplitude A(f) = 1 – Z1/(2Z1+Z2) = (R1+R2+iwC1R1R2)/((1+iwC1R1R2/(2R1+R2))

f = arctan (wC2R1R2/(R1+R2)) – arctan (wC2R1R2/(2R1+R2))

t = f/w

t(w~0) = C1 R1 R2 / ( R1 + R2 ) – C1 R1 R2 / ( 2 R1 + R2 )

The two filter frequencies are:

f1 = 1/((2R1+R2)(2pC1R1R2) = f

f2 = 1/((R1+R2)(2pC1R1R2)

Using the crossover frequency instead, the time delay at low frequencies is 1/(2pf) = 250 ms. Together with a time shift of 70 ms of the direct signal, the effective time delay is 320 ms. MS

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Excel circuit simulator.

Download circuit simulation spreadsheet (in MS Excel format)

The MS-Excel circuit simulator (above) lets you experiment with the filter’s component values for the standard crossfeed filter (see my article here for a simulator for my enhanced-bass filter). Component values can be changed in the upper left corner of the table. The pictures (the frequency response and time delay characteristics of the crossfeed filter) automatically adapt. The central branch in the spreadsheet consists of a resistor and a capacitor in series. For the standard version of the crossfeed filter, the capacitance has to be given a very high value so as to act as a short-circuit. Each side-branch has a parallel pair of a resistor and a capacitor in series. Each side branch thus consists of two resistors and two capacitors.

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Circuit model for spreadsheet simulator

For the standard version of the crossfeed, one resistance has to be set to zero and its corresponding capacitance to the regular value (440nF / 320nF / 220nF). The other resistance has the regular crossfeed value (1.1k / 1.5k / 2.2 kOhm) and its corresponding capacitance is set to a very high value. The values in the spreadsheet are for a relatively low crossfeed level.

CONSTRUCTION DETAILS

Building a headphone amplifier is like building a power amp – only the current demand is just a little bit lower (about a factor 100). Various designs can be found in the internet, and it is relatively easy to integrate the proposed crossfeed filter. I designed my own (see the picture) which is op-amp based. I know, many hi-fi enthusiasts say “yuck” to opamps, but note that even in many so-called “High-End” CD-players, opamps are found in the signal path for amplification and filtering.

The op-amps should be chosen with care. They have to be able to deliver relatively high current values and to drive low impedance loads. I decided to use the National Semiconductor LM6171. This is a wide band (100 MHz) voltage-feedback opamp that is able to deliver 10V into a 100 ohm load. To prevent difficulties when driving low-impedance headphones (32 ohms or less), I placed a 47 ohm resistor at the output of each channel. With my Beyerdynamics (DT990/DT931, 600/250 ohm) and my Sennheiser (HD600, 300 ohm) headphones, the opamps perform most adequately. Other alternatives are the LM6172/6181/6182, the OPA604/627 by Burr-Brown (used in the HeadRoom systems) or the LTC1206/1207 by Linear Technology (able to drive 30 ohm loads!).

Schematic of headphone amp with natural crossfeed network.
Figure 6a

Schematic of headphone amp with enhanced-bass natural crossfeed network.
Figure 6b

The schematics shown in figures 6a/6b/6c represent the third generation of my original design. There are two versions of the headphone amplifier: one with the standard crossfeed and one with the enhanced bass crossfeed (see An Enhanced-Bass Natural Crossfeed Filter for more information). The standard crossfeed sound is nothing for a bass-freak. One should not expect a punchy bass, only a relaxation of the sound.

The two crossfeed settings of the original bass-enhanced filter are comparable to the low and the high crossfeed levels of the standard filter in this article. The enhanced-bass filter has a frequency response very similar to the modified Linkwitz filter to compensate for any apparent loss of bass due to the crossfeed. While I hardly notice any specific loss of bass with the standard filter, I present the enhanced-bass design to give DIYers an option of which filter to build. I have added a medium crossfeed level so that both the standard and bass-enhanced filters now have 4 settings. The 4.4K resistor bridges the switch at all settings to reduce any “blops” during switching.

The output stage of each opamp is connected to one of the voltage rails by a 1.5 kOhm resistor. This forces the output stage into class-A functionality and increases soundquality considerably. Also 10pF capacitors are added to the feedback loop to increase stability at high frequencies. Careful matching of all resistors prevents offset voltages and the need of coupling capacitors and the amplifier now is DC-coupled. The power supply has a ground loop breaker, so the audio inputs and outputs MUST have floating grounds – their grounds cannot be directly connected to the enclosure. (See A Precision Preamplifier-Power Amplifier System with Natural Crossfeed Processing for more discussion about biasing opamps to function in class A and ground-breakers in power supplies).

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Figure 6c

A stable power supply to the opamps is crucial for optimal performance (figure 6c). I used a toroid transformer. Charge-reservoirs of 1000uF and two voltage regulators (L7815/7915) provide constant voltages of ±15VDC. Moreover, high frequency noise is extensively filtered by 100mH inductors. 2200uF capacitors and 100nF film capacitors further reduce any ripple and noise after the voltage regulators. I admit that the large capacitors are a little bit overdone. When I switch the device off it will still work for about 10 seconds. However, capacitors are relatively cheap, so why not.

The amplification and power supply circuitry is rather straightforward. I have just a few notes:

Both the standard and enhanced-bass filters have four crossfeed level settings: none (normal stereo mode), low, medium and high. I personally prefer the low and the medium crossfeed levels for most applications.

No opamp is perfect. To ensure a zero offset voltage at the output the impedance values in the circuitry have been carefully balanced. Do not use other resistor values than as indicated in the schematics, as this might lead to damage of your headphones.

The headamp has two sockets for connection to headphones. Both sockets will provide different sound characteristics. One socket has a very low output impedance and gives the amp tight control over the headphone action. However, many headphones have been sonically optimized to be driven by an output impedance of 120 Ohms and may sound better when connected to the other socket. Generally, the low impedance socket provides a clean sound whereas the high impedance socket yields a warmer sound. Use the one you like most. There is no risk of damage to your headphone by connecting it to either socket. You can also use the sockets to connect two headphones simultaneously. However, the volume produced by the high impedance socket will be slightly lower than that of the other socket.

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As far as the signal paths are concerned, I used WIMA MKS-film capacitors (the red ones), electrolytic capacitors of ELNA, and ¼ Watt metal-film resistors.The 7 Watt toroidal power transformer is by Nuvotem-Talema, and the 10k logarithmic potentiometer by Alps (the blue one).

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Figure 8

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Figure 9

The frequency response and the time-delay of the standard crossfeed network are presented in the figures (figures 8 and 9). Turning the switch from position 1 to position 4 activates and increases the crossfeed. I personally prefer positions 2 and 3 for most recordings. To see the response graphs of the enhanced-bass filter, see my article here.

Last warning: do yourself a favor and don’t use cheap parts. Have a decent (logarithmic) potentiometer (Alps or equal), film capacitors, gold plated sockets and a proper metal housing. Even if you don’t like and use the crossfeed-filter (which I doubt), your headphones are most likely to sound much better than they ever did using the normal headphone socket of your CD-player or amplifier. Manufacturers generally do not care about the sound quality of headphone sockets and build them cheaply. They sound accordingly.

Now, before you start building, you surely want to know how the crossfeed network performs. I have made 3 WAV-recordings to demonstrate the effect of the crossfeed filter. The first recording is made without filtering. The second and third files are made with the filter switch in positions 2 and 4 respectively. The recordings contain a simple balance test. The effect of the filter on normal music is more subtle and only can be truly appreciated (in my humble opinion) while listening for longer periods than is possible with these files.

Balance test w/o filter (1.4Mb – wav)
Balance test w/filter in position 5 (1.4Mb – wav)
Balance test w/filter in position 6 (1.4Mb – wav)

The standard crossfeed amplifier is now commercially available as the Corda HA-1 from Meier Audio. For people that are interested to build their own headphone amplifier, I’m offering a DIY-kit of the standard crossfeed amplifier for approximately DM 450 ($203 US). I know that it is cheaper to buy all the electronic parts by yourself, but please note that for the money you have a professional PC board added (with soldering mask, tinned soldering eyes, and all the holes drilled) as well as the aluminum case with a 4 mm (!) front- and a 2 mm back-plate with all the holes milled. Although construction is rather straightforward (a component plan is added), this project is not intended for real novices. If you doubt your own skills please contact the author for the possibilities to obtain a finished device.

Well, I never heard any of the HeadRoom systems, but the reviews of these systems very well describe the impressions that I have with my own amplifier. The sound is not really externalized out of the head, because pinnae-reflections and reverberation are missing. However, the soundstage in our head becomes more continuous and has more inner-logic. With some recordings the effect is hardly heard; with other recordings, the effect is rather extreme. All recordings can be listened to for much longer periods of time, without your head screaming “Take those headphones off!”

Have fun!

c. 1999, 2000 Jan Meier.
The author’s website: Meier Audio.

Addendum

3/19/99: Capacitor values in figure 1 corrected from uF to nF.

3/20/99: S2 in figure 6a now has a bypass position.

5/11/99: Updated amplifier and power supply schematics to simplify design, avoid excessive DC offsets at the amplifier outputs and provide for a more flexible bass enhancement control. Figures 6 – 9 modified.

5/27/99: Updated PC board and component layouts with 47 ohm resistor at preamp outputs (after the coupling capacitors) to prevent oscillation at RF frequencies.

9/7/99: Tomohiko Ishigami made this Pocket Headphone Amplifier (see the article by Chu Moy) which uses the acoustic simulator circuit by Jan Meier (see A DIY Headphone Amplifier With Natural Crossfeed). He reduced the gain of the amp to unity to minimize problems with noise, which he later traced to the CD player itself.
The larger case is from Radio Shack (RS 270-213).

I feel it is very good idea to use modular approach. I used separate board for crossover and the buffer itself. This way, I did not have to go crazy load all the parts on one board which will result in a hay wire. Also, this approach is useful when I was trying to achieve smaller size.

I was able to use 1uF polymer capacitor for input…. These are so tiny. It is made by Phillips and you should be able to find it in Digikey [Digikey part nos. shown below]. I used this same type for my crossover circuit allowing me to conserve a lot of space:

3019PH-HD 1uF Metal Film Box ( 10mm (H) by 7mm (W) by 6mm (L) )
3015PH-ND .22uF Metal Film Box
3011PH-ND .047uF Metal Film Box

 

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11/23/99: Added calculations for time delay of crossfeed signal.

1/7/00: Several DIYers have installed Jan Meier’s natural crossfeed filter as a front-end to the pocket amplifier by Chu Moy. Jan offers these tips re: selection and placement of a volume control for this combination:

It all depends on the specific circuitry. Generally it might be better to place the pot after the filter instead in front of it. The influence of impedance changes might be less pronounced. A 10 kOhm pot will certainly be too small. 50 kOhm will be a kind of minimum I think. However, note that with certain opamps this will result in changing offset voltages, since the DC impedance changes with volume.

2/8/00: Added calculations for time shift of direct signal.

2/16/00: Tomohiko Ishigami wrote:

Here is a picture of Mr. Meier’s stand alone acoustic simulator unit which I use with my Melos headphone amplifier. It sounds very transparent having almost nothing in the signal path. It is so small you can hide it anywhere. I just wanted to show you since it looks so cute!

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5/1/00: Jeff Medin‘s version of the Pocket Headphone Amplifier (see the article by Chu Moy) has 3 sections: a gain stage, the crossfeed filter by Jan Meier and an output buffer stage. The power supply creates a virtual ground with a Texas Instruments TLE2426 voltage reference instead of a resistor divider network. The 1uF (or less) capacitors are Philips box-type metal film; capacitors larger than 1uF are Panasonic FC/Z series. All resistors are 1/4W Yaego metal film. Medin writes:

This is the FIRST amp I built after discovering HeadWize. It is a “basic” pocket amp with the natural crossfeed circuit by Jan Meier. ALL parts are from Digikey. It has very good decoupling with 3 capacitors per opamp and 3.9uH chokes (the 4 green things that look like resistors – they are connected in series with each V+ and V- lead). The first stage (on the left side of the first picture) is an OPA2132 with a gain of 10.

This then feeds a Meier crossfeed circuit (4 caps in a row) and you can see the crossfeed resistor on TOP of the board (2.2k) with long leads. The output from the filter feeds a voltage follower (OPA2132) stage. The switches are for low and high crossfeed, power, and bypass for binaural recordings. I used Philips Box style metal poly caps. The two large caps on top & bottom of board are 1uF input caps. The output is taken from the OPA2132… with a 100 ohm resistor… which is included in the feedback loop so it will drive very low z phones and to prevent oscillation due to capacitance from long cables. I used 100 ohm resistors in BOTH stages. If the resistor is OUTSIDE the loop, the impedance WILL have an effect on the sound of the phones, sometimes more bass, sometimes MUCH less signal based on the efficiency of the phones, etc. etc. Some phones as you know are spec’d to be run from an impedance of 100-150 ohms or so. I have a 15 year old APT/HOLMAN preamp (designed by same guy that invented THX-Tom Holman) and it’s Headphone Jack is driven by a 5532 with a 120 ohm resistor OUTSIDE loop right to the jack. I would suggest people can try both (like Jan did) and see what sounds better to them. I would DEFINITELY recommend that you include this resistor in at least the last stage.

Note that I did not have any problems, I always “over-build” opamp circuits so I don’t have to worry about problems later on. It’s just habit.

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5/4/00: Added MS Excel circuit simulation application link and image.

1/16/00: Jan Meier has started a new company called Corda, which is offering a DIY headphone amp kit for approximately $250 US. This headphone amp has the latest circuit revisions as described in his article A Precision Preamplifier-Power Amplifier System with Natural Crossfeed Processing, such as opamps biased for extended operation in class A, ground-loop breaker topology, a crossfeed filter with 4 settings, and two headphone jacks with different output impedances. He writes:

The circuitry in the DIY kit represents the third generation of my original design. The output stage of each opamp is connected to one of the voltage rails by a 1.5 kOhm resistor. This forces the output stage into class-A functionality and increases sound quality considerably. Also 10pF capacitors are added to the feedback loop to increase stability at high frequencies. Careful matching of all resistors prevents offset voltages and the need of coupling capacitors. And the amplifier now is DC-coupled.

The power supply has a ground loop breaker, so the audio inputs and outputs MUST have floating grounds – their grounds cannot be directly connected to the enclosure. (See A Precision Preamplifier-Power Amplifier System with Natural Crossfeed Processing for more discussion about biasing opamps to function in class A and ground-breakers in power supplies).

The headamp has two sockets for connection of a headphone. Both sockets will provide different sound characteristics. One socket has a very low output impedance and gives the amp tight control over the headphone action. However, many headphones have been sonically optimized to be driven by an output impedance of 120 Ohms and may sound better when connected to the other socket. Generally, the low impedance socket provides a clean sound whereas the high impedance socket yields a warmer sound. Use the one you like most. There is no risk of damage to your headphone by connecting it to either socket. You can also use the sockets to connect two headphones simultaneously. However, the volume produced by the high impedance socket will be slightly lower than that of the right socket.

I know that it is cheaper to buy all the electronic parts by yourself, but please note that for the money you have a professional PC board added (with soldering mask, tinned soldering eyes, and all the holes drilled) as well as the aluminum case with a 4 mm (!) front- and a 2 mm back-plate with all the holes milled. Although construction is rather straight-forward (a component plan is added), this project is not intended for real novices. If you doubt your own skills, please contact me for the possibility of obtaining a finished device.

5/27/01: Headphone amp and power supply schematics updated to Corda HA-1 design. Added schematic for enhanced-bass crossfeed filter amplifier. Also added revisions in text discussing latest updates.