A critical measure for all audio components is distortion, typically specified as total harmonic distortion (THD) or total harmonic distortion + noise (THD+N). THD is defined as:
where :
f1 = fundamental frequency amplitude f2 = second harmonic amplitude f3 = third harmonic amplitude f4 = fourth harmonic amplitude fn = nth harmonic amplitude (< 20 kHz)
THD+N is the rms sum of all frequency components (up to 20 kHz) divided by the amplitude of the fundamental.
However, measuring distortion has long been a difficult task, especially when figures of greater than 90 dB or more are needed. The frequency source must be of a purity exceeding the desired measurement. Plus, the analyzer's noise and dynamic range can limit the achievable resolution.
The conventional approach to this problem is described in Figure 1. A 1-kHz sine wave is passed through a low-pass filter to attenuate source harmonics greater than 1 kHz. The amplitude of the sine-wave fundamental is measured and stored as the reference level for calculations. The filtered signal is applied to the device under test, whose output is passed through a band-reject (notch) filter, removing the fundamental-frequency component. Thus, the remaining distortion components have a much smaller total dynamic range. Finally, the signal is amplified and measured with a spectrum analyzer or digitizer. After correcting for gain, the THD can be measured and calculated.
While this is an effective method, it suffers from some inherent weaknesses. The final THD can only be as good as the filtered source. Also, the filters must not introduce their own harmonic distortion and noise components. Furthermore, the notch filter must avoid attenuating the second harmonic or be calibrated to correct for this error.
Figure 2 shows a different, simpler approach. This method takes advantage of the characteristics of a low-distortion instrumentation amplifier. These devices are differential-input, single-ended output components that subtract and amplify the difference between the inputs. Monolithic INAs have excellent resistor matching, hence good gain error and common-mode rejection. Plus, a wide range of gains are easily achieved using an external resistor. Low noise and distortion are features of products specifically designed for audio applications, such as the INA217.
The unfiltered 1-kHz reference signal is applied to the INA noninverting input, while the same signal is applied to the DUT, whose output is connected to the remaining INA input. Because the instrumentation amplifier possesses good common-mode rejection at 1 kHz (typically >80 dB), the distortion components common to the source and the DUT output are attenuated accordingly.
Because most of the signal amplitude at 1 kHz is removed, the remaining difference signal, due only to the DUT, can now be gained as needed (commonly about 40 dB) and output to the spectrum analyzer. This raises the signal above the system's noise floor by the INA gain of 40 dB.
Resolution is also enhanced by the same 40-dB gain. Moreover, the results aren't affected by the reference sine-wave purity, as source distortion components are common to both inputs and rejected by the INA. On top of that, no filters are needed.
I think that the ultimate solution is to use both differential cancellation AND severe signal filtering (two Twin-T's !) before FFT analysis.
With very carefull calibration, -150 dB limit can be reached at 1 kHz (22050 Hz sampling rate, 65536 FFT size, infinite averaging, no smoothing at all) in few minutes. The whole system must be battery powered, temperature stabilzed and perfectly shielded.
Pierre Lacombe -November 09, 2007
I'm really fond of articles like this one. I like this article too much.
Panchagalle Ganesh -October 14, 2004
With proposed circuit the attenuations of source harmonics (1, 2, 3, and 4 kHz) will be defined mainly with phase differences of input signals due to freqency of DUT! For illustration a table is given: Cutoff frequency-- Attenuation of source harmonic components in dB 1 kHz 2 kHz 3 kHz 4 kHz 5 kHz 14 9 6 4 10 kHz 20 14 11 9 40 kHz 32 26 22 20 100 kHz 40 34 30 28 1 MHz 60 54 50 48 10 MHz 80 74 70 68
Savo Leonardis -August 31, 2004
The article was intresting, useful, and easy to understand. However, how do you ensure that the DUT has a phase angle of zero? If it does not, how can you correct for this?
*** The delays are there, but contribute little error. As it turns out, the devices I was measuring (small signal amps) have BW of 1 to 6 MHz and slight delays. The delay causes the biggest error in the fundamental, which is not important. The harmonics exhibit error also, but as they are so far down in amplitude (with respect to the fundamental), the errors are second order and have little effect on the total distortion calculation. This was not intuitive to me so I tried it on the bench with excellent results. We have compared this method with Audio Precision System Two audio testers with quite favorable correlation. I have yet to look at some audio power amps with a more limited bandwith, but plan to in the near future. If you try it, please let me know how it goes. Thanks. --Jerry Riddick
Trevor Jones -August 02, 2004
As long as the PHASE response of the DUT is exactly 0 degrees, this will work well. If it is not, then sadly this entire idea will give erroneous results. A 1 degree phase difference in the DUT at the frequency of interest results in about -35 dB cancellation. Mind you that as long as you look carefully at a scope trace of the output of the canceller with the input signal as reference, it's not hard to see the fundamental coming through to warn you that you are not measuring what you think you are measuring.
RB -June 30, 2004
Measuring distortion is the matter of my life. I've tried to reach the ultimate by building my VK-1 oscillator and VK-2 distortion meter. Their full pdf-documentation can be found in the Test Equipment directory on the site www.amplifier.cd.
Vladimir Katkov -June 29, 2004
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