Get here Loudspeaker Data Files for Simulation Programs
|Computer based acoustical Loudspeaker Measurements
The output of a loudspeaker is difficult to measure because it has to be done with a microphone. Measurement microphones are usually omnidirectional which means that they pick up, not only the output from the loudspeaker, but also any background noise and sound reflected from objects around the loudspeaker.
This means that echoes from the walls of the room where the measurements are done corrupt the direct output from the 'speaker. One solution to this problem is to build an anechoic chamber-- a very expensive option. Another technique is to make the measurements outdoors with the test speaker and microphone at the top of a tower. This works well, but only on sunny days when there is no background noise.
A third alternative becomes possible if we can capture the output from the loudspeaker as a digital signal. Once the acoustic signal from the 'speaker has been digitised it can be analysed mathematically. Suppose that we feed an impulse into the loudspeaker. The Fourier transform of the impulse is the frequency response, a sum that can be done easily in a computer. So, if we capture the impulse data we can find the frequency response of the loudspeaker. The trick is to choose which part of the impulse to transform. By deliberately truncating the 'tail' of the impulse we can effectively cut off the reflections since these arrive at the measurement microphone later than the direct sound. The reflections are removed by a window in time.
This windowing technique is very powerful and is used in many commercial loudspeaker measurement packages. Its chief disadvantage is that it is difficult to calibrate the microphone to show the absolute sound pressure level measured. This is because of the mathematical technique used (FFT) combined with the normal methods for mic' calibration (use of piston-phones and so on). Even so the method produces good relative measurements.
If you need to make a set of measurements for comparison you must use identical settings for each. Otherwise the relative levels that you record will not be comparable.
There are a few fundamental rules that apply to this form of measurement.
1. The time to the first reflection determines the window length. This is fixed by the size of the measuring room.
2. The window length determines the low frequency cutoff. The low frequency cutoff of the measurement is always greater than the reciprocal of the window length.
3. Because the data that is captured and processed has a constant number of points per Hz there are very few data points in the lowest octaves of the measurement. Half the data points will be in the top octave.
4. The width of the impulse determines the bandwidth of the measurement. The wider the bandwidth required the narrower the pulse must be.
5. The energy in the pulse is determined by its width and height. A wide tall pulse will contain more energy than a short narrow one.
What do you need?
- PC min. 1GHz
- min. 512Mb memory
- 60mB free Harddisk capacity
- Windows 98, Windows 2000, Windows Vista ( depends of software )
- Fullduplex Soundcard
- Measurement Microphon
- extern Microphon Pre Amplifier
- recommended: Power Amplifier (min. 15Watt @ 4ohms)
Single channel measurement setup for acoustical measurements
Dual channel measurement setup for acoustical measurement
To protect the soundcard input from high voltage that is generated by the power amplifier, it is recommended to use a
voltage probe circuit, as shown in fig. 2.
Values of resistors R1 and R2 have to be chosen for arbitrary attenuation (i.e. R1=8200 and R2=910 ohms gives probe
with -20.7dB (0.0923) attenuation if the soundcard has usual input impedance - 10kOhms).
In a single channel mode and semi dual channel mode this probe is not connected.
Semi Dual channel measurement setup for acoustical measurement
To make an on axis response for the loudspeaker you will need to set your speaker up on a stand as farfrom all reflecting surfaces as possible. You should also try to find the quietest place to do the measurements as noise added to the response will degrade the results. The measurement microphone should be placed about 1 m in front of the speaker on your chosen measurement axis. When you have set up the mic and speaker measure the distances to the nearest reflecting surface with a tape measure. Normally this will be either the floor or the ceiling.
Low frequency cutoff = 343 /( (2 * (x2 + h2)0.5)
Where 343 m.s-1 is the speed of sound. For a room with a ceiling height of 2.4 m and a measuring distance of 1 m this gives:
Low frequency cutoff = 343 /( (2 * ((0.52 + 1.22)0.5) = 132 Hz
So, for this size room you know that any data shown below 132 Hz is rubbish! Bear in mind that you should put an extra margin on this because there are very few data points collected in the lower octaves of the measurement.
The next step is to connect the equipment.
Dealing with Noise
Where possible you should try to use a quiet room for your measurements, maybe using the room outside normal working hours. However, you may find that the background noise in your test environment still makes it difficult to get good results. If this happens there are three things that may help. First, use a steep high pass filter on the microphone to remove low frequency noise. You can tailor this filter to the practical low frequency limit of your measuring room.
|USB 1.1/2.0 Soundkarte Audigy2 NX MP3+
|Soundblaster Live Platinium
|Soundblaster X-Fi Extreme
|One commercial microphone that is suitable for speaker measurements is Behringer ECM8000. It requires preamplifier with phantom power.
||Building a microhone adequate for speaker measurements is rather simple. One suitable electret microphone capsule is Panasonic WM-60AY. Except small peak at 15kHz, frequency response is straight and may be used without correction.
||Professionel microphon and very expensive!
Recommended Microphone Preamps
| Frequency range: . . . . . . 20–20 000Hz
MIC INPUT: . . . . . . . . 20–70 dB, switchable
STEREO LINE: . . . . . 0dB
for 1V at the output: . . . . 0.16 –100mV, switchable
MIC INPUT: . . . . . . . . 2.2 kOhms
STEREO LINE: . . . . . 10 kOhms
Phantom power: . . . . . . . +24V
PREAMP OUT: . . . . . 1V/12 V max., 100Ohms
STEREO LINE . . . . . . 1V/ 6V max., 100Ohms
High-pass filter: . . . . . . . 100Hz/-3dB, 12dB/oct.
Low-pass filter: . . . . . . . . 12kHz/-3dB, 12dB/oct.
Mic: . . . . . . . . . . . . . . > 66dB
Line: . . . . . . . . . . . . . . 80dB
Power supply:. . . . . . . . . 15 V~ via supplied AC/AC
adaptor (230V~/50Hz /10VA)
or four 9V transistor batteries
|Behringer MIC 100 Tube ULTRAGAIN
|Frequency response: <10Hz - 43Khz (3dB)
Gain: variable (+26dB to +60dB)
20dB Pad: Level attenuation 20dB
+48V: activatesthe phantom power
Phase reverse: 180degree
Input Level: 8-segment Led meter (-24, -18, -12, -6, 0, +6, +12dB)
|DIY Recording and Measurement Microphones
||The purpose of this article and small group of projects is firstly to introduce the electret microphone into the ESP projects lineup, and secondly to allow the reader to build a microphone that although uncalibrated, can be used to great effect as a measurement mic with any loudspeaker project, or for very high quality recordings.
Rod Elliott (ESP)
|DIY Microphone PreAmp
||Eric Wallin Microphon Amplifier with PCB.
Very good performance, easy to build (PCB) and 9 VDC powered.