Preamp

[Sean_Goddard]

Fair enough. I just realise that sometimes simulations don't necessarily translate into real life.

I tried changing the pulse width on your sim to incorporate the energy saving system I designed which measures the coil current and when there is no more di/dt it shuts the pulse off (as maximum I coil has been achieved). But is stops working. I know my circuit works so I took it out and just left the changes pulse width. Still Nada!

I still love this preamp design though, it's simple, cheap and very effective. Maybe it's why the late Great Andy Flind favoured it for some of his designs.

[Teleno]

What is VN ?

It's a dummy voltage source for noise analysis. You can ignore it (short circuit).

[Teleno]

Nice work Teleno, but I have to ask, what happens to this massive pulse that D1 "absorbs"

The pulse is exactly the same as the one absorbed by the diode in the traditional R/D design.

I'm assuming sufficienty dimensioned electrolytic capacitors are already connected to the power supply, just like in the normal design.

The .ZIP file contains the .asc simulation file for LTSpice. Feel free to experiment with modifications.

[Sean_Goddard]

Nice work Teleno, but I have to ask, what happens to this massive pulse that D1 "absorbs" surely that is relying on a power supply with 0R impedance ? Would it not be better to put another diode with the cathode to D!/D3 junction and a nice hefty electrolytic to ground from that point or else I feel that the supply quality will suffer.

Please advise on your thoughts regarding this.

I intended my suggestion to be a method for capturing the Rx signal for analysis, NOT for use in the final design. Sorry I didn't make that clearer. My Bad.

[6666]

What is VN ?

[Teleno]

Has somebody bread-boarded Teleno's front-end? It definitely should be tried.

Here's my theoretical preamp design for wohever wants to try it out.


Characteristics (theoretical):

Gain: 23
Input noise: 3.7nV/sqr(Hz) (2.3 uV total in the band 1KHz - 700KHz)
Current consumption: 3mA.
Output with no signal: 2.23V

Explanation:

D1: Absorbs flyback, dimension according to expected current.
D2: prevents the emitter-base of Q1 from being damaged by reversed voltage.
D3/R3: polarizes Q1 working point at Ic = 100 uA.
R1: damping resistor, a bit lower than its theoretical value.
R2: gain resistor. Gain is aprox. R2/R1.

Avatantages realtive to traditional R/D:

High gain,
Damping and preamplifying in one single step for less noise.
Lower than theoretical damping resistor contributes less noise.

Disadvantages:

3 mA consumption.
Baseline not zero (still achievable by negative voltage rail on R2's end.)

[Monolith]

Reply #6?

Yes, that is what I had in mind. Sorry, my bad memory has cheated me again. This is only a damping circuit.

Teleno, why don't you show us your pre-amp circuit? We could breadboard it together with your damping circuit.

[green]

Has somebody bread-boarded Teleno's front-end? It definitely should be tried.

Reply #6?

[Monolith]

So what about the preamp?

Has somebody bread-boarded Teleno's front-end? It definitely should be tried.

[Old cart]

So what about the preamp?

Good point, but everything here is at least about the preamp measurement, selection and optimization!

[Teleno]

So what about the preamp?

[Old cart]

Have a look at this; http://www.cirrus.com/en/pubs/proDat...2-34-BS_F3.pdf

With two you would have a sample rate of 96KHz X 4 = 384KHz and 23Bit resolution. What if each of the samples was taken in the following manner Pulse 1 = samples 1,2,3,4, pulse 2 = samples 5,6,7,8 etc like is done with a fast scope?

Any milegage in this or do we need faster? If so what sample rate do we need. Can we sample the decay curve at 4 points and use them as a reference by using each ADC channel per point?

In a project I was saw a long time ago, radio signals were being decoded from all over the world. The signal average level was -123dB and these were certainly readable in terms of clarity of audio after processing. Considering "Earth Noise" is at around -127dB I think that is pretty good going. Just shows you what is possible with the correct hardware, unfortunately I can't post any schematics here as the project methods used were classified.

I can only sit back and watch now as you guys have this in hand and are teaching me a few things.

Does anyone else think it would be a good idea to go DUAL COIL? At least we could eliminate a lot of crud from the Tx from getting into the Rx signal (hopefully).

This technique is called interleaving. It can certainly be done but is quite tricky, I would think, at the hobbiest level. The samples have to be placed very accurately, within a few pS of the right place to avoid errors when the waveform is spliced together. Also some sort of calibration is necessary between the individual A to D converters so they all match perfectly. Otherwise the number of effective bits ( the measure of how much waveform distortion is introduced) will be decline. This is particularly important at higher frequencies. I think direct conversion,using the A to D to replace the integrators and beyond, is a great idea on a high performance software defined detector. It has not been used in commercial designs because the high performance converters are quite expensive. Also software cost are high. I like the idea of using audio converters as there has been a lot of development work done on them. The particular device you mention have serval neat features like programmable gain and high impedance differential inputs. It also has integrated digital filters. I like separate Tx and RX coils.

[Monolith]

that is nice software. It will also let you try out various filters and see the effect on the noise.

|Thanks Old cart, for the tips and explanation for FFT. Very useful.

[Sean_Goddard]

Have a look at this; http://www.cirrus.com/en/pubs/proDat...2-34-BS_F3.pdf

With two you would have a sample rate of 96KHz X 4 = 384KHz and 23Bit resolution. What if each of the samples was taken in the following manner Pulse 1 = samples 1,2,3,4, pulse 2 = samples 5,6,7,8 etc like is done with a fast scope?

Any milegage in this or do we need faster? If so what sample rate do we need. Can we sample the decay curve at 4 points and use them as a reference by using each ADC channel per point?

In a project I was saw a long time ago, radio signals were being decoded from all over the world. The signal average level was -123dB and these were certainly readable in terms of clarity of audio after processing. Considering "Earth Noise" is at around -127dB I think that is pretty good going. Just shows you what is possible with the correct hardware, unfortunately I can't post any schematics here as the project methods used were classified.

I can only sit back and watch now as you guys have this in hand and are teaching me a few things.

Does anyone else think it would be a good idea to go DUAL COIL? At least we could eliminate a lot of crud from the Tx from getting into the Rx signal (hopefully).

[Old cart]

Thanks for reply #81. Shows me how to get a good FFT. What I'm missing in the PI example is how to use the FFT plots. It still looks easier to see if the peak to peak signal is grater than the peak to peak noise. The FFT could show high 60 Hz noise when it could have a low 60 Hz peak to peak because it's continuous.

Got it. Since I have no plots it is hard to give you a precise example. However I think you make a good point. For a simple quick test you can simply measure the ratio of the P-P noise and compare that to the P-P signal. That is not likely to be very accurate as the ear is not sensitive to peak but rather more likely to respond to the sound power or voltage squared power. More importantly the FFT on most scopes is a power spectrum so more closely responds to what you would hear. Lastly the FFT will give you reasonably accurate frequencies which can be very important when you are trying to figure out where the noise is coming from. For example if there is a peak at 10KHz it is likely to be coming from the charge pump. If there is a rising noise spectrum below a couple of hundred Hz it is likely caused by an op amp near the input that has excessive 1/f noise. Knowing this makes it much easer to isolate the problem.

One process could go like this.

1. Disable the transmitter and measure the preamp output noise with the coil replaced by a resistor of equivalent value.
2. Exchange op amps or adjust the restor values around the op amp to maintain constant gain but reduce noise. Generally to lower Johnson noise you lower the value of the resistors. Unfortunately if you are using protection diodes you can not lower the input resistor much below 1000 ohms. FFT's are very useful at this point. If noise at specific frequencies is present try to figure out where it is coming from, for example via the supply lines or perhaps due to board contamination or maybe even RFI being rectified in the input diodes.
3. Turn the transmitter back on and connect the coil. Measure the noise using the same set up as above. By using the scope trigger it should be possible to only capture data when the transmitter is off and the signal has decayed. In this way you are only measuring noise, but at more of a system level. If your scope has the ability to measure the RMS value of the spectrum you can measure the effective noise power broadband.
4. Repeat the noise reduction efforts but this time you wil be seeing the effects of recovery of the big current pulse drawn by the coil, power supply instability and other added noise.
5. Move the trigger forward so that the coil flyback is captured to measure MOSFET Avalanche noise on the preamp output.
6. Record and think about all noise effects and try to get them as low as possible.
7. Move to the output of the integrator and repeat, once again trying to get the lowest noise. You may find that the noise level varies as you move the sample points and widths. There is lots of room for tuning here.
8. Keep moving forward in the signal path one stage at a time. Since the bandwidth is going down in each stage you may have to slow down the scope sweep speed to capture the frequency range you are interested in. Since you are reducing the noise at each step in the process the noise should be going down BUT each stage typically has gain so noise is likely to go up. Absolute value is NOT the important thing, what you are trying to do is to lower the value from where you started. It is also important to note that you always get to see the time domain waveform at every step which a help to ensure none of you noise lowering efforts causes other problems.
9. Somewhere after the integrator you can introduce a target into the equation. By noting the target response you may find you can reduce the stage bandwidth. Knowing that noise goes down with the sq root of the bandwidth helps. Note also that with a target you should adjust threshold so that you have normal audio response. This lets you detremine if some of the noise is coming back from the audio stage, which are often pretty crude and can introduce noise.

NOTE, some more advanced scopes allow you to average the frequency domain, on the FFT. This is very useful to determine if the noise is synchronous withe the transmitter ( noise wil not go down with averaging) or asynchronous ( noise will go down ).

One last point it is good dea to practice using the FFT's on a KNOWN signal, like the calibrator signal. This 1 KHz square wave, which has all odd harmonics ( 1,3,5,7,9,11KHz) out to maybe 11 KHz, should be sampled at around 50 to 100 KS/s. You wil be able to learn about how to best use the FFT on your scope as well as what effect adjusting the controls have.