If the operation of an instrument is complicated enough to encourage or require the use of a computer for the analysis of the data, then it is reasonable to put as much of the functionality as possible in software, and use hardware described by some or all of these conditions:
1. already available for a different purpose;
2. general purpose and thus relatively inexpensive due to the size of the market;
3. already owned by or at least familiar to the intended users of the new instrument.
If these conditions are met, then it is possible to make an instrument that uses minimal special hardware, and as for the software, once written, it can be distributed to anyone who wants it for free in its initial and updated forms.
The instrument under discussion here is a device for measuring the impedance of a pickup across frequency, presenting basic and derived quantities in a graphical form. The hardware is the A/D recording interface, available in many forms for recording music, but requiring only two channels for this application. This discussion describes the use of an example of this hardware, while using available software to demonstrate that the required functionality can be achieved, recognizing that convenient operation would require a custom program.
The minimal required user constructed hardware is a stand for holding the PUT with connectors for its leads and cables for connecting to the recording interface. This is something that is easy to make. The version used for these tests is shown here ().
The leads of the PUT are connected two the two terminals. Underneath, one terminal is connected to a 1K resistor which goes to ground; the other is connected to a random noise signal from the recording interface, in this case an Apogee Duet 2. Each terminal also is connected to one of the input channels of the Duet. The terminal connected to the resistor goes to a high input impedance instrument channel; the other one goes to a 4 DBU line input. The voltage across the resistor is a measure of the current through the PUT. The other signal is a measure of the voltage across the series combination. The impedance calculated from the ratio is the desired impedance plus 1K real, easily subtracted in the software. Thus there is no reason to build something like an analog current to voltage converter: this is simpler.
The Duet has a stereo head phone output and two speaker outputs. In the current perverse terminology, a speaker output drives an amplifier input, while a headphone output really does connect to head phones. Either could be used to drive the pickup with random noise. The headphone output has more leakage to the inputs, and so one speaker output is used instead, with the headphone amp muted.
The two voltages are measured across frequency by Electo Acoustics Toolbox running on a Mac. EAT performs the cross spectral analysis necessary to find the transfer function relating the two. This information is transferred to another program for computing the impedance and other quantities. Also the results can be displayed as graphs. Computing low noise results requires some integration; this is partly a result of the random nature of the signal, partly from additive noise present in any system. Because of the cross spectral analysis, the random signal contributes less noise than one might expect. Although the signal level at each frequency is random, the result is a ratio of the two inputs, and that is not random. It is still subject to some noise, however.
Integration is for less than one minute; the frequency resolution is 24Hz. The frequency response of the duet has some phase shift and a small amplitude change at the bottom and top. It is only differences that matter, and so one could do better in this regard by using the preamps on both channels. However, a calibration is necessary in any case, and it is possible to use a higher level into the pickup when skipping the preamp in the signal monitor channel.
The gain of the preamp is adjusted to give levels significantly below clipping, but high enough to avoid losing signal to noise ratio. A calibration is performed, and the preamp gain must be remembered so that it can be reset to that level in the future. (Or perform a new calibration.) The calibration consists of connecting a resistor across the terminal instead of a pickup. A value is chosen that gives a similar signal level that a typical pickup does. The integration is performed for ten times longer than for the PUT so that the calibration process introduces little random noise. The calibration data are transferred from EAT to the computation program, processed, stored, and applied to any PUT data that is input. The calibration is frequency by frequency and removes both amplitude an phase errors. Of course, these are audio amplifiers that are capacitor coupled, and so we cannot get information at dc, but we can see the limits as the frequency goes down.
A later post will present some results.
1. already available for a different purpose;
2. general purpose and thus relatively inexpensive due to the size of the market;
3. already owned by or at least familiar to the intended users of the new instrument.
If these conditions are met, then it is possible to make an instrument that uses minimal special hardware, and as for the software, once written, it can be distributed to anyone who wants it for free in its initial and updated forms.
The instrument under discussion here is a device for measuring the impedance of a pickup across frequency, presenting basic and derived quantities in a graphical form. The hardware is the A/D recording interface, available in many forms for recording music, but requiring only two channels for this application. This discussion describes the use of an example of this hardware, while using available software to demonstrate that the required functionality can be achieved, recognizing that convenient operation would require a custom program.
The minimal required user constructed hardware is a stand for holding the PUT with connectors for its leads and cables for connecting to the recording interface. This is something that is easy to make. The version used for these tests is shown here ().
The leads of the PUT are connected two the two terminals. Underneath, one terminal is connected to a 1K resistor which goes to ground; the other is connected to a random noise signal from the recording interface, in this case an Apogee Duet 2. Each terminal also is connected to one of the input channels of the Duet. The terminal connected to the resistor goes to a high input impedance instrument channel; the other one goes to a 4 DBU line input. The voltage across the resistor is a measure of the current through the PUT. The other signal is a measure of the voltage across the series combination. The impedance calculated from the ratio is the desired impedance plus 1K real, easily subtracted in the software. Thus there is no reason to build something like an analog current to voltage converter: this is simpler.
The Duet has a stereo head phone output and two speaker outputs. In the current perverse terminology, a speaker output drives an amplifier input, while a headphone output really does connect to head phones. Either could be used to drive the pickup with random noise. The headphone output has more leakage to the inputs, and so one speaker output is used instead, with the headphone amp muted.
The two voltages are measured across frequency by Electo Acoustics Toolbox running on a Mac. EAT performs the cross spectral analysis necessary to find the transfer function relating the two. This information is transferred to another program for computing the impedance and other quantities. Also the results can be displayed as graphs. Computing low noise results requires some integration; this is partly a result of the random nature of the signal, partly from additive noise present in any system. Because of the cross spectral analysis, the random signal contributes less noise than one might expect. Although the signal level at each frequency is random, the result is a ratio of the two inputs, and that is not random. It is still subject to some noise, however.
Integration is for less than one minute; the frequency resolution is 24Hz. The frequency response of the duet has some phase shift and a small amplitude change at the bottom and top. It is only differences that matter, and so one could do better in this regard by using the preamps on both channels. However, a calibration is necessary in any case, and it is possible to use a higher level into the pickup when skipping the preamp in the signal monitor channel.
The gain of the preamp is adjusted to give levels significantly below clipping, but high enough to avoid losing signal to noise ratio. A calibration is performed, and the preamp gain must be remembered so that it can be reset to that level in the future. (Or perform a new calibration.) The calibration consists of connecting a resistor across the terminal instead of a pickup. A value is chosen that gives a similar signal level that a typical pickup does. The integration is performed for ten times longer than for the PUT so that the calibration process introduces little random noise. The calibration data are transferred from EAT to the computation program, processed, stored, and applied to any PUT data that is input. The calibration is frequency by frequency and removes both amplitude an phase errors. Of course, these are audio amplifiers that are capacitor coupled, and so we cannot get information at dc, but we can see the limits as the frequency goes down.
A later post will present some results.
Comment