Hi Guys

In the past, analog signal generators fell into two categories: low-distortion oscillators (LDO), and function generators.

LDOs were sinewave generators intended for audio testing and more specifically , to be used along with a distortion analyser to test THD (total harmonic distortion). If you were lucky, the LDO distortion was around 0.002% (20ppm) and these would be fairly expensive and require some time to settle. More affordable sinewave generators exhibit 0.3% THD or so, maybe down to 0.1%.

Function generators produce sine, square and triangle wave outputs. THD for the sine is as described for affordable sinewave-only units.

In our digital age, test signals can be generated using computers using data tables and DACs to generate various wave shapes. The resolution of a sinewave depends upon the number of data points per cycle, similar to the sampling rate of an analog-to-digital conversion, just in the opposite direction. The DAC output is typically a current fed into an analog integrator stage (a virtual-earth stage with the virtual-earth node fed directly from the DAC). THD can potentially be quite low.

The digital version of a function generator is called an "arbitrary waveform generator" , or AWG. The AWG is a cheap and dirty signal generator that can produce, sine, square, triangle, pulse and other wave shapes. Those other shapes can be practically any shape you wish to program, limited only by the interface provided by the manufacturer.

For audio testing, the AWG should have a THD spec listed for its sinewave output. Generally, this will be in the 0.3% to 0.1% range. More expensive AWGs will offer sine THD down to 0.03% or 0.02% - the lowest I've seen for affordable AWGs. Cheapo AWGs are often around $60cdn where the units with the lowest-THD sine output are $5-600cdn.

DSPs can have AWGs built in, with sine THD listed (or not) down to 0.05% to 0.03%. It is certainly convenient to have both devices in one box as it makes frequency response tests easier to do.

Over the years I've installed a number of mixing FX loops based on the TUT Best All Tube FX Loops with a lot of success and find that design to be transparent and more useful than the serial loops in many amps.  I like the idea that the dry path isn't going through all the buffers, mixing circuits, etc. in the pedals as well. 

In general, I've only thought an FX loop is needed in higher gain amps but more and more I'm seeing them lower gain amps (Bassmans, Plexis, etc) and get requests to install them in.  At what level of gain is there enough benefit to warrant the effort?  I was hoping I could get some guidance  and opinions on the matter.  Thanks everyone for the help!

Hi Guys

Stereo playback systems are only really defined or limited to the fact that there are two audio channels. Apart from that, however you wish to approach things is up to you.

There are some guide lines that are helpful.

The room should be one where a Human conversation sounds natural. Not too reflective or echoey. Not too dead. Just right. Something Goldilocks would approve of were she interested in audio matters.

The two acoustic sources should be full range, covering the range of Human hearing from 20Hz up to 20kHz. If you are interested in pipe organ music, you might want to add a subwoofer to each side to extend the range below 16Hz - the lowest note Bach demanded.

However the speakers are arranged on each side with respect to whether it is one box or several with narrow ranges, they should be clustered to appear as much like a point source as possible. There are many speaker types that arrange the woofer lowest, then the midrange above, and tweeters above that. There may be intermediate ranges of drivers, or multiple drivers per range. In the case of planar (flat) speakers, sections of the panel may be limited in range and the horizontal plane of the audio image will tilt with frequency and with seating height of the listener.

One aspect of the speaker selection is with regard to the width of the "sweet spot" for the listener. This is the optimal seating position to experience the best audio image.  Wider speaker separation can widen the sweet spot, as will a more diffuse presentation of the sound. Bose made their name using acoustic diffusion, essentially bouncing the sound off the walls so the image is "wider than the room". There is a natural effect in the result, but new distortions are added by the room itself. Other decisions made by the designer further impair the accuracy of the sound. For example, in the 901 model there are eight drivers facing the rear and one facing the front (towards the listener). All the drivers are identical with just under 1-ohm voice coils, then all wired in series to present 8-ohms to the amplifier. Every driver is undamped with respect to its reverse current generation, which distorts the sound as the amplifier has no control over the driver cone position. This was a super high-profit model for Bose and they sold a lot of units. Obviously a lot of people found the sound to be acceptable.

The choice of amplifying equipment and signal sources is up to each individual. What do you like? Do you want integrated equipment, or separates? do you listen to vinyl LPs? 78s? Edison rolls? to tape? to CDs? to the radio? or do you just need TV sound? Do you prefer all solid-state? or tubes? or hybrid? Do you need this to tie in with home automation systems? Should the stereo be small? the size of a bread box? the size of a refrigerator? maybe the latter are your speakers?

There is usually a distinction between "accuracy" and "fulfilling". For some people, these are the same, but for everyone there is a nostalgic goal of finding what first excited us about listening to music that we want to recapture. Once you are there, the system should disappear and you are inside the music.

Hi Guys

The every-day working man's headphone has always been the dynamic type, where the transducers are simply small dynamic loud speakers. These will be 8R up to 600R.

Traditional headphone outputs from power amplifiers were simply a set of 100R-1W dropping resistors feeding a stereo jack that had closed contacts to disable the main speakers. This meant that whatever the big PA was, it would be idling however hot it normally would, providing the usual voltage swing at much reduced current.

Preamps might have a headphone output that was fed from the main output, again with closed switches on the headphone jack, or from an auxiliary output dedicated to the task. If the latter, there would generally be a headphone Level control.

Dedicated headphone circuits were often just a couple of transistors before ICs became dominant, in which case a generic opamp circuit would drive the headphones. In all cases, there would be a low-value resistor added in series with each cup. This actually reduces THD and stabilises the driver from the cable impedance. At the lowest impedance, parallel opamps or a buffer help for high-SPL listening. 1mW at 8R is only 90mV at 9mA. 1mW at 600R is 770mV at 1m3A. These are RMS; peak values are 40% higher. Nested feedback circuits or discretes can push THD to inaudibility.

Most opamps can drive 1k loads, with more now capable and rated for 600R. The ubiquitous 5532 is rated for 600R loads yet it can drive 30R cups through 50R resistors quite well. For most people who only casually use headphones, this is adequate. The nice thing about opamps and most similar discrete circuits is the high power supply rejection ratio (PSRR), which is the ability to ignore supply noise. As a headphone driver with modest or unity gain, the PSU can be quite "cheezy" and still be hum-free.

Of course, with the low power requirement of headphones, we can be inventive and creative  and use tubes, transformers, or hybrid circuits as we see fit. Whatever your aesthetic is, just go for it.

Hi Guys

Planar magnetic headphones offer very high SPLs at pretty low distortion. The load is almost purely resistive, therefore it is easy to drive. The voltage and power required is also quite low.

Remember: The sensitivity rating  for headphones is at 1mW input, and this often produces 100dB of sound.

The leader in the planar magnetic headphone camp is Audeze. Their models have become rather diffuse inasmuch as every model is "the best" but they are styled for specific applications and music types. The element to be driven is a resistive track laid down on thin film. The resistance is usually 20R to 110R. At 1mW, the drive required is 140mV up to 330mV. at 7mA to 3mA, respectively. These are RMS values. You can see that driving the magnetic planar headphones is not very difficult.

If you want to push the drivers to their claimed 130dB limit, a bit more drive is needed, at 4V5 to 10V5 respectively for 20R and 110R. These voltage are why you see a recommendation to use a 5W amp, or a 10W amp. It is not that the headphones need the power; rather, that they need the voltage if driven single-ended. Differential drive cuts the voltage swing in half as far as each driver is concerned, except now you need two drivers per cup with symmetric signals.

Whether single-ended drive is used or differential, standard opamp circuits with or without buffers added will work reasonably well. The ultimate performance is attained using discrete fully complementary circuits. Wherever possible, we use inverting gain stages to eliminate common-mode distortion, but we have to be careful to preserve overall phase coherence from input to output. Because this is all low-voltage +/-24V solid-state circuitry, distortions of all types can be pushed to -140dB.

Hi Guys

The lowest-distortion headphones available are electrostatic types. The most famous brand offering these, Stax, from Japan, lovingly calls the headphones "ear speakers".

Electrostatic headphones require a high driving voltage, plus a high fixed bias voltage, none of which can be provided by a standard audio power amplifier. The bias voltage is between 200V and 600V and is current-limited by 5-10Megohms. The reference drive is 50Vrms + 50Vrms for 100dB of acoustic output. The listing of two voltages means there is differential drive, two inputs of opposite phase, which could also be listed as 100Vrms differential drive.

The headphone element has two fixed elements to which the audio signals are applied. Between these is a metalised film of thin plastic that has the fixed voltage applied to it. As the signal voltages move positive and negative, the positive-charged middle element is pulled towards one stator while being pushed by the other, then drawn and pushed to the opposite stator over the audio signal cycle. Since there is electrostatic force applied evenly over the movable film, it can move very quickly and as a rigid plane with very low distortion.

The stators appear capacitive to the amplifier, typically around 100pF or so each. This is not a particularly difficult load except that the charging and discharging currents can be high if there is no current limiting during fast transitions.

Stax offered many types of ear speaker drivers ranging from passive to exotic, setting the standard for others like Koch to follow. The passive driver uses a step-up transformer to raise the voltage needed for 8R speakers up to the requirement for the electrostatic panels. The secondary of the transformer is center-tapped, with the CT grounded, creating the symmetric anti-phase drive signals. A small transformer provides mains isolation and feeds a voltage multiplier to generate the bias voltage. Compared to the distortion room loudspeakers add, the distortion of the transformer and the very-low-THD of the ear speakers combined is much lower, and usually lower than a conventional dynamic headphone driven through the usual 100R dropping resistor from the amplifier output.

Dedicated headphone amplifiers for ES were introduced which allowed the user to turn off his large amp and not waste all that power, driven by the system preamplifier / source selector. The dedicated driver is always made as a differential amplifier from input to output, with dual feedback loops. Internal voltages are typically +/-400Vdc, providing an incredible signal swing approaching 280Vrms x2. However, considering that a combined 100Vrms produces 100dB of output, these rails would allow 5.6 times the total voltage swing, assuming no losses, which corresponds to only 15dB more sound. We are already way above the Human Scale of Loudness, so all this headroom gets us is freedom from clipping on signal peaks. We would also expect there to be better linearity when not using so much of the available swing.

The first diff-amp offerings had open-collector outputs, where there is active pull-down and passive pull-up. This was because the highest-voltage semiconductors were NPN. Even with this, cascoding was used throughout to distribute voltage stress and to improve high-frequency response. The next iteration added buffering with current-source loading for active up/down load control. This reduced THD slightly. A further variant was like the first with the high-voltage output transistors replaced with a dual-triode tube.

Stax offered a lower-output battery-powered driver that unfortunately used a switch-mode power supply. The hash from the SMPS was all over the output and the THD was poor and fidelity low. On a good analogue scope or any DSO, the SMPS noise at the output is terrifyingly ugly.

Personally, I have built drivers for Stax headphones using linear +/-100V rails (same as the Stax unit above) and have zero issue with clipping. THD is unmeasurable. The topology is a typical form using only inverting gain blocks and fully-symmetric circuits. Now that we have highly complementary semiconductor pairs, circuits like this are very common. As an experiment, I switched to 40V rails and still had no evidence of clipping when listening at sane loudness levels.

Many headphone enthusiasts have designed all-tube, hybrid and solid-state drive circuits for ES cups and report their findings. Some of the listening tests seem a bit scary to me, when they have super-high-voltage drive circuits and can hear clipping !

Hi Guys

Headphones can be a bit tricky to select as wearing them is still a bit unusual despite being ubiquitous and despite how much many individuals use them on a daily basis. Compared to listening to music through loudspeakers distant from us, the headphone experience can be "tinny" and "tiny".

A room mix is exactly that: each ear receives sound from both of the stereo speakers and there is a blend of sounds. The electrical signals to each speaker may be highly distinct, but our brain perceives the musical picture with both directionality and uniformity. This means that headphones are already at a disadvantage compared to room presentation, in that hard-left and hard-right sounds will be separated to an extreme.

The ambience of the room is added to the ambience of the recording, and this too is lost with headphones.

The room imposes no aural or spatial restriction on our physiology, where a headphone is extremely close to our ear canal, or inside it in the case of "ear buds" or "in-ear monitors". Both of the latter are very imperfect and must be customised to the individual. Typical headphones cup the outer ear and can be ventilated or not. Too tight a fit can give a claustrophobic feeling. Too open or loose may disturb others when the music level is too high. The coupling of the cup to the ear is preferred to be tighter if a good bass response is to be attained. It is the absence of bass that dominates the feeling of a "thin sound".

High frequency sounds can impact as harsh very easily with many headphones. This is partly due to the direct coupling of the transducer with the ear canal, compared to the transit through a larger space for room sound, which softens the attack. The proximity to the transducer allows the listener to hear the normal speaker breakup modes as distortinos that cause intermodulation effects, with IM being much harsher to our ear than THD.

In our modern time, the use of small ear-buds and headphones was popularised by the Sony Walkman, then the Sony Discman, then all the copies. This portable sound deteriorated with the advent of digital sources, such as the ipod, where data reduction compounded the fidelity loss imposed by data compression. Unfortunately, there is a generation of music listeners who have never experienced proper fidelity. Live sound is too loud to be decipherable. Background music in bars and restaurants is too loud and is also distorted by the poor acoustic environments. Music mixing and production values are ever-changing and not always for the better. Environmental noise impacts our hearing and is almost entirely responsible for hearing impairment as we age.

With all of this, we try to listen carefully in the music store as we audition headphones.

Headphones are extremely sensitive, with a reference power of 1mW used. Most headphones produce 100dB of sound at 1mW input, which is enough to cause serious hearing damage very quickly. So, having more reasonable sound pressure levels does not take much power at all.

Hi Guys

When we listen to recorded music, we hope to experience the intended presentation of the performance. The original recording might be a live performance, or it might be a multi-track mix of the band. However it began, that is what we hope to experience.

Our listening room should be acoustically balanced, ideal to have a conversation in. This means it is not entirely "dead" or echo-free, and not too "live" with a noticeable echo or "zing". Hotel rooms are often of a shape that has a zing, which is a short echo, and lack enough absorptive materials to tame that echo.. Every natural environment has some reverberation, so we are accustomed to and need that ambient information when we are deciphering sounds.

Because we have binaural hearing, the best electronic playback system will have two acoustic sources for left and right. Provided the recording has the musical voices mixed correctly, with ambient information intact, this stereo presentation will feel three-dimensional to us. Even a monaural recording can have a natural feel provided the ambient portion is present.

The left and right acoustic sources MUST be full-range, with a frequency response covering 20Hz to 20kHz.

These full-range acoustic sources can be single drivers, multiple drivers, or sub-sat systems. The sub-sat term came originally from the use of adding a sub-woofer to a full-range cabinet, which then morphed into what was more accurately a separate bass cabinet and mid-treble cabinet. However the frequencies are divided between drivers or boxes, the group should be tight and positioned left or right. The listener needs there to be cohesive left and right signal sources

Theatre sound is a bit gimmicky and presents sound in an inaccurate format for proper music enjoyment. The focus is on the video screen, and thus, the acoustic playback is skewed to center-screen. A single subwoofer or woofer is placed on the center-line, with mids-treble splayed out to left and right. That such a layout is now commonplace in most homes is a travesty to musical enjoyment. The main reason for such a trend is economic rather than qualitative.

I use the example of a helicopter flying by. You hear the thumps of the propeller coming from the left and moving to the right. Along the way, you hear the high-frequency part of the engine noise follow from left to right. If you play this  sound back on a true stereo system, it sounds exactly as it did live, except you have control over its loudness If we play this back over a home theatre system, the thumps are in front of us. Is the helicopter to the left? to the right? in front of us? We do not know until we hear the high-frequency portion move from left to right but all the while the thumps are dead center.

Bass is directional from the point of view of the observer. So, any playback system that has a single acoustic source for bass is inherently flawed.

Do not be fooled by the newer home theatre systems, the point-2, which has two subs. These are driven by the same signal and represent the same mixed information. The only benefit of the second woofer box is that having two such boxes allows placement that can break up or minimise room resonance activation by low-frequency tones.

So, the ultimate music playback system is good old-fashioned stereo. Movies sound better through it as well.

Hi Guys

Anyone who has read my books knows that my preference for music playback is proper stereo. There are several aspects involved regarding how to achieve the same experience in the listening room as existed on stage during a performance. That stage might also have been a recording studio, where the recording process can alter the performance quite radically depending on how each instrument sound is captured.

The simplest approach to recording with an eye (ear?) to playing back in stereo, is to use a two-microphone  setup to capture the room mix. This mimics how a listener would experience the performance directly, and hopefully this transfers correctly to the listening room experience. There are thousands of recordings made this way from the earliest days of electronics up to the digital age. Some are great, some are poor, and most are good.

In our stereo recording approach, we can use two microphones on their own stands separated however the recording engineer determines, or two mics can be "crossed" at close proximity to each other, or a binaural mic-head can be used.

The separate microphones method is the easiest to implement as it requires no special equipment and actually the minimum of skill to achieve a pretty good result. The microphones should be in positions where listeners would be, far enough away that each microphone picks up all instruments. Their left-right positioning should be modest but distinctly not coincident; rather, tending to extreme separation. This provides a natural blending of the sounds with room ambience for a normal feel. If the microphones are widely spaced, the distinction of instrument placement across the stage may be enhanced, with caveats: Each instrument must have its own single speaker cabinet or "zone" where its sound is dominant; the PA sound ideally carries only vocals or has a proper mix corresponding to the stage positions of the instruments; or the PA sound is minimised as much as possible in the stereo microphone placement.

There are a lot of details in the above descriptions to be expended upon as we go along.

The binaural head-mic is designed to mimic the placement of Human ears within our head, and thus, the mount is shaped like a Human head, with ear canals that have microphones at the place where our ear drum would be. The exterior contouring is either highly simplified or shaped to represent an average shape of a Human ear. Ears are unique and no two people hear exactly the same thing. besides, our brains make an interpretation of the information coming from our ears as to what it is we are hearing.

The stereo signals captured by the microphones only need two tracks on tape or whatever form the recording system takes. Playing it back on a distortionless home listening system is the ideal case, especially if that system is properly positioned in a"Human conversation rated" room.

I have some power transformers that are made specifically for full wave bridge rectifiers. In some cases, I may want to use a 5V filament transformer and hybrid tube/ss diode bridge rectifier. I was wondering what to expect as far as a sag and voltage drop is concerned. Will it drop as much as voltage as the tube alone?  How would I approach calculating it? 

Thank for the help!