Thinking I should get a new soldering station.

I've had my eye on a Weller station for the longest time, but put off getting one for one reason or another.

Looking at the selection, I think the choice comes down to the two following models, or, indeed the equivalent made by another company:

WTCPT (link)

WES51 (link)

The main difference is in the WTCPT the tip temperature is fixed, but rigorously controlled, while the WES51 has user adjustable power but the tip temperature is left unregulated.

I'm leaning towards the WES51 has being slightly more in line with my own style of work: I'm more likely to want higher or lower temperatures depending on the job at hand than I am to require "700F" exactly.

Update 1.

A little concerned the WES51 is only 50W max. That's less than the maximum I use now for the tough stuff.

Hakko has two models in the price range I'm looking...

Under windows at least, Audacity is unable to record audio at bitrates above 16 bit.

It will seem to, all right, but the data is quantized at 16 bit (30 microvolt LSB), regardless of the settings chosen.

The attached images show the same source, the first recording is made in Audacity, supposedly 24 bit, but actually only 16 bit, while the second is recorded with a program than actually supports 24 bit, exported, and imported into Audacity. The data is amplified +70dB in both cases to make the difference visible.

Audacity will happily manipulate and save high bit rate data, but as a result of licensing restrictions and on account of it being freeware, it does not support the actual recording of this data.

***

Any internet search will confirm that the Windows version of Audacity is limited to 16 bit recording. And yes, it's more of a limitation of Windows than it is of Audacity. My irritation, however, is chiefly with Audacity...

Two headphone amplifiers sharing the same basic MOSFET source follower output stage.

When the source current and source resistance are optimized for the given headphone load and similar maximum output power (~50 mW at 1% THD), the distortion pattern vs. output power is remarkably similar.

One plot below is simulation, the other measurements. The J-Mo 2 simulation closely matched the actual measurements, it wasn't worth my while to generate a full simulated data set when I already had the measurements on hand. No reason to suspect that the Szekeres sim is inaccurate, either.

The take home message is the distortion characteristic of a MOSFET follower is what it is, and unavoidable. Take it or leave it, as it were. However - and this is key - if you don't optimize the stage for the headphone impedance, the distortion for a given output power will increase significantly.

As an aside: Greg did his homework with the original circuit....

I've always enjoyed the sound of the Greg Szekeres' Headphone Driver (buffer) and derivatives sharing the MOSFET source follower output stage.

I've often wondered however, whether it's distinctive sound is because it is unusually free from noise and artifacts, or because its unusually prone to heavy second harmonic distortion.

It's not hard to set this up in LTSpice, but I haven't seen it done before. So, for your education and enlightenment, I present the harmonic distortion vs. output power data for the original "classic" circuit as uploaded to Headwize all those years ago. The LTSpice asc file is also included I you want to play along. The harmonic data is generated by hand, reading the FFT peaks for 10 or so different input voltages.

Changed the collector load on the voltage amplifier stage to a current source (Q3), as per the Marantz SR2285B circuit.

Also increased the resistance of the feedback connection, R6+R8, there seemed to be no obvious advantage in making it much smaller than the typical load (>10k). The compensation capacitor C2 is increased to match, to flatten the HF response.

The circuit can drive light loads to +20 dB. Of course that's not much of a challenge for this general class of circuit.

I'm a bit stumped as to what the next logical step is from here. Seems to me to depend on what you actually want the circuit to do.

Over the last couple of years most of my interest in audio has been with transistors. I've been slowly teaching myself to read and understand the circuits.

Circuits like this one for example. Not hard, but still a bit too complicated for me to understand without the helpful wikipedia markup attached.

Instead I've looked at primarily at the schematics I have for discrete audio preamplfiers, 1970's vintage typically. Based on what I've learnt so far, I've done up a "test mule" in LTSPICE, shown below.

It's not a circuit you should build. It's for pedagogical purposes, though it does actually work reasonably well - in simulation anyway. Its just a simple starting point to observe how the different parts interact under simulation.

It came up at the help desk, but I want to put this before diyaudio.com members generally:

I feel strongly that people who build audio equipment as a hobby should take it upon themselves to obtain a basic understanding of both the practical and theoretical aspects of electronics. Take a trip to the library and read through the first couple of chapters of electronics textbooks, that kind of thing.

It's more than just the safety aspect, I think of it as a basic necessity...

So, how many people here are familiar with the following statement?

The impedance of a capacitor is -j/([omega]C)

Familiar as dirt? Never heard of it before?

If you don't think about it too hard, you'd imagine that the signals in the phono stage would be smaller than the signals in the line level preamplifier stage that follows it. Or that the signals in the DAC/CD player would be about the same level or slightly lower than the signal in the preamplifier.

As usual, the answer is "it depends". It depends on the sensitivity of your speakers, how loud you are listening, and the voltage gain of the amplifier. It also depends on whether we are talking about a MM phono stage or low output MC.

My point is simply this: the volume control is an attenuator, and at the typical "9 o'clock" position the input signal is reduced in magnitude by about 35 dB.

That cuts it back down to being comparable to the output of a moving magnet cart, and much, much smaller than anything found in a DAC stage.

It means you absolutely, definitely, positively must spend as much effort and care...

Mid-range 1970's stereo receiver.

I was curious to find out a) what the phono circuit was, and b) how tight the RIAA response might have been.

The answer is "four transistors" and "pretty damn good", respectively.

We are impressed. These Japanese engineers knew a thing or two. I would like to see some of these old circuits resurrected as discrete phono stages with modern components to see just what they are capable of.

I did up the X-reg circuit in LTSpice.

Results shown below, together with the LTSpice .asc file you can use to play around with this yourself.

First attached image shows FFT for the rectified DC (green), reference voltage (red) and X-reg output (blue) for the designed-for 10 mA output (top) and a more punishing 100 mA (bottom).

Second image shows an LTSpice screengrab for the LT1086 with bypassed adj pin under comparable loading. Input voltage in blue, output in green. This is a reasonable approximation of a "good" IC regulator.

Last image shows a plot of the exported LTSpice FFT data for the X-reg and the LT1086-12V (Cin 1000uF, Cout 100uF) both at nominal currents of 10 mA. The LT1086-12V is a reasonable substitute for a generic LM7812, i.e. a "bad" IC regulator.

A typical op amp will have sufficient PSRR to mop of the residual noise from the bypassed LT1086. The fixed LT1086-12V, on the...