Audio system measurements

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Audio system measurements are made for several purposes. Designers take measurements so that they can specify the performance of a piece of equipment. Maintenance engineers make them to ensure that equipment is still working to specification, or to ensure that the cumulative defects of an audio path are within limits considered acceptable. Some aspects of measurement and specification relate only to intended usage. For example, magnetic tape speeds and types, interface specifications, or power output.

Others are intended to signify the quality or 'fidelity' of reproduction that is perceivable by a human. It is important that such measurements be based on psychoacoustic principles, so that they truly measure the system in a way that is 'subjectively valid'.

Subjectivity and frequency weighting

Measurements based on psychoachoustics, such as the measurement of noise, often use a weighting filter, . It is well-established that human hearing is more sensitive to some frequencies than others, as demonstrated by equal-loudness contours, but it is not so well known that these contours vary depending on the type of sound. The measured curves for pure tones, for instance, are different from those for random noise. The ear also responds less to short bursts, below 100 to 200 ms, than to continuous sounds <ref> Moore, Brian C. J., An Introduction to the Psychology of Hearing, 2004, 5th ed. p137, Elsevier Press</ref> such that a quasi-peak detector has been found to give the most representative results when noise contains click or bursts, as is often the case for noise in digital systems. <ref> BBC Research Report EL17, The Assessment of Noise in Audio Frequency Circuits, 1968.</ref> For these reasons a set of subjectively valid measurement techniques have been devised and incorporated into BS, IEC, EBU and ITU standards. These newer methods of audio quality measurement are universally used by broadcast engineers throughout most of the world, as well as by some audio professionals, though the older A-weighting standard is very commonly used elsewhere.[1] Subjectively valid methods came to prominence in consumer audio in the UK and Europe in the 1970s, when the introduction of compact cassette tape and Dolby noise reduction techniques revealed the unsatisfactory nature of many basic engineering measurements. The specification of weighted CCIR-468 quasi-peak noise, and weighted quasi-peak wow and flutter in particular became common; and attempts were made to find more valid methods for distortion measurement.

No single measurement can assess audio quality. Instead, it is usual to apply a series of measurements to test for the various types of degradation that can reduce fidelity. Thus, when testing an analogue tape machine it is necessary to test for wow and flutter and tape speed as well as for distortion and noise. When testing a digital system, testing for speed variations is normally considered unnecessary, but testing for aliasing and timing jitter can be desirable in case these aspects are causing audible degradation. The claim often made that different methods of measuring noise, or distortion, are better suited to different items of equipment is not upheld by professional audio engineers. Once subjectively valid methods have been shown to correlate well with listening tests over a wide range of conditions, then such methods become the preferred ones for all purposes, and it is important to realise that engineering methods are not even valid when comparing like with like. One CD player, for example, might appear to have more noise than another CD player when measured RMS, or even A-weighted RMS, yet sound quieter and measure lower when 468-weighting is used. This could be because it has more noise at high frequencies, or even at frequencies beyond 20 kHz, which is not important since this is not heard. (See noise shaping.) This was what happened in fact when Dolby B was introduced. The noise sounded 10 dB quieter, but failed to measure much better unless 468-weighting was used rather than A-weighting.

Measurable performance

Analog electrical

Frequency response 

The signal should be passed at least over the audible range (usually quoted as 20 Hz to 20 kHz) with no significant peaks or troughs. The human ear can discern differences in level of about 3 dB, so peaks and troughs must be less than this. Modern equipment is capable of less than ±1 dB variation over the quoted frequency range. Rapid variations over a small frequency range (ripple), or very steep rolloffs are considered undesirable as they can correspond to resonances associated with energy storage which produce delayed echoes and hence colouration on the sound.

Total harmonic distortion (THD) 

For high fidelity, this is usually expected to be < 1% for electronic devices; mechanical elements such as loudspeakers usually have higher levels. Nowadays very low distortion is easy to achieve in electronics with use of negative feedback, but the use of heavy feedback in this manner has been the topic of much controversy among audiophiles — for more on this see electronic amplifier. Loudspeakers typically produce more distortion than electronics, and 1–5% distortion is not unheard of at moderately loud listening levels. Human ears are less sensitive to distortion in the bass frequencies, and levels are usually expected to be under 10% at loud playback. Distortion which creates only even-order harmonics for a sine wave input is sometimes considered less bothersome than odd-order distortion.

Output power

A genuine measurement quotes the maximum sinewave power output per channel, which by convention is considered the most meaningful measure of power available on music signals, though it should be noted that real, non-clipping music has a high peak-to-average ratio, and is usually of a lower level. The common figure of PMPO (peak music power out) is meaningless and generally used in disingenuous marketing literature. The correct measurement is average power, usually estimated from an RMS measurement of voltage or current, which gives it the common, but incorrect, designation of "RMS power". See also Audio power.

Power specifications require the load impedance to be specified, and in some cases two figures will be given (for instance, a power amplifier for loudspeakers will be typically measured at 4 and 8 ohms). Any amplifier will drive more current to a lower impedance load. For example, it will deliver more power into a 4-ohm load, as compared to 8-ohm, but it must not be assumed that it is capable of sustaining the extra current unless it is specified as able to.

Intermodulation distortion (IMD) 

A measure of the spurious signals resulting from unwanted multiplication of different input signals. This effect is contributed by non-linearities in the system. Again, heavy negative feedback can tame this effect, but many believe it is better to design to minimise it arising in the first place.

Noise 

The level of unwanted noise generated by the system itself, or by interference from external sources. Hum usually refers to noise only at power line frequencies (as opposed to broadband white noise), which is introduced through interference or inadequately regulated power supplies.

Crosstalk 

Caused by stray inductances or capacitances between components or lines, crosstalk results in things such as unintentional mixing of stereo signals or mixer channels. This is given in dB relative to a nominal level of signal in the path receiving interference.

Common-mode rejection ratio (CMRR) 

In balanced audio systems, equal and opposite signals (difference-mode) are used, which are subtracted, canceling out interference which affects both signals equally (common-mode). CMRR is a measure of a systems ability to ignore any interference and especially hum which arises at its input. It is only important on long lines, or when ground loop problems exist.

Dynamic range and Signal-to-noise ratio (SNR) 

A measurement of the range of signal levels the device is capable of.

Dynamic range refers to the ratio of maximum to mimimum loudness in a given piece of music or programme, and this measurement quantifies the maximum dynamic range an audio system can carry. This is the ratio (usually expressed in dB) between the noise floor of the device with no signal and the maximum signal (usually a sine wave) that can be output without distortion.

Signal-to-noise ratio (SNR), however, is the ratio between the noise floor and an arbitrary reference level or alignment level. In "professional" recording equipment, this reference level is usually +4 dBu (IEC 60268-17), though sometimes 0 dBu (UK and Europe - EBU standard Alignment level). 'Test level', 'measurement level' and 'line-up level' mean different things, often leading to confusion. In "consumer" equipment no standard exists, though −10 dBV and −6 dBu are common.

Different media exhibit different amounts of noise and headroom. Though the values of course vary widely between units, typical analogue cassette might give 60 dB, a CD almost 100. Most modern amplifiers have >110 dB dynamic range, which approaches that of the human ear; 160 dB. See Programme levels.

Phase distortion, Group delay, and Phase delay 

A good system will maintain the phase coherency of a signal over the full range of frequencies. Phase distortion can be extremely difficult to reduce and eliminate. The human ear is actually largely insensitive to phase distortion, and so for many this figure lacks importance; however, there are always those who will argue the opposite.

Transient distortion 

A system may have low distortion for a steady-state signal, but distort sudden transients. This is often due to a lack of power delivery fast enough to supply the system during the transient. Related measurements are slew rate and rise time. Transient distortion can be hard to measure. Many otherwise good power amplifier designs have been let down by having an inadequately responsive power supply. Most typical loudspeakers generate significant amounts of transient distortion, though some exotic designs are less prone to this (e.g. electrostatic loudspeakers and plasma arc loudspeakers).

Damping factor 

A higher number is better. This is a measure of how well a power amplifier can control the reactive load of a loudspeaker. The amplifier must be able to damp out resonances caused by the mechanical inertia of the moving parts of the speaker. Essentially this involves ensuring that the output impedance of the amplifier is as close to zero as it can be made. Damping factor is actually just a different way of specifying the output impedance of an amplifier. It is significantly affected by the cables used to connect the speakers to the amplifier, as poor quality cables can have a large resistance compared to a typical amplifier output.

Mechanical

Wow and flutter 

This pertains to the drive mechanism of analogue media, such as vinyl records and magnetic tape. "Wow" is slow speed variations, caused by longer term drift of the drive motor speed, whereas "flutter" is faster speed variations, usually caused by mechanical defects such as out-of-roundness of the capstan of a tape transport mechanism. A lower number is better.

Rumble 

The measure of the low frequency noise contributed by the turntable of an analogue playback system. A lower number is better.

Digital

Note that digital systems do not suffer from many of these effects, even though the same processes occur in the circuitry, since the data being sent is symbolic. As long as the symbol survives the transfer between components, the data itself is perfectly maintained. The data is buffered by a memory, and is clocked out by a very precise crystal oscillator. The data usually does not degenerate as it passes through many stages, because each stage regenerates new symbols for transmission.

Digital systems have their own problems, however. Digitizing adds quantization noise (random data) which is measurable, depending on the resolution of the system. Clock timing errors (jitter) result in non-linear distortion of the signal. The quality measurement for a digital system basically revolves around the probability of an error in transmission. Otherwise the quality of the system is defined more by specifications than measurements, such as the sample rate and bit depth. In general, digital systems are much less prone to error than analog systems. The analog systems invariably at the inputs and outputs of the digital system can suffer analog effects, however.

Jitter 

A measurement of the variation in period between clock cycles, which should theoretically be exactly the same period. Less jitter is better.

Sample rate 

A specification of the rate at which measurements are taken of the analog signal. This is measured in samples per second, or hertz. A higher sampling rate allows a greater total bandwidth or flatband frequency response. It can also reduce the effects of jitter.

Bit depth 

A specification of the accuracy of each measurement. For example, a 3-bit system would be able to measure 2³ = 8 different levels, so it would round the actual level at each point to the nearest representable. Typical values for audio are 8-bit, 16-bit, 24-bit, and 32-bit. The bit depth determines the theoretical maximum signal-to-noise ratio or dynamic range for the system. It is common for devices to create more noise than the minimum possible noise floor, however. Sometimes this is done intentionally; dither noise is added to decrease the negative effects of quantization noise by converting it into a higher level of uncorrelated noise.

To calculate the maximum theoretical dynamic range of a digital system, find the total number of levels in the system. Dynamic Range = 20·log(# of different levels). Note: the log function has a base of 10. Example: An 8-bit system has 256 different possibilities, from 0 – 255. The smallest signal is 1 and the largest is 255. Dynamic Range = 20·log(255) = 48 dB.

Sample accuracy/synchronization 

Not as much a specification as an ability. Since independent digital audio devices are each run by their own crystal oscillator, and no two crystals are exactly the same, the sample rate will be slightly different. This will cause the devices to drift apart over time. The effects of this can vary. If one digital device is used to monitor another digital device, this will cause dropouts in the audio, as one device will be producing more or less data than the other per unit time. If two independent devices record at the same time, one will lag the other more and more over time. This effect can be circumvented with a wordclock synchronization.

Linearity 

Differential non-linearity and integral non-linearity are two measurements of the accuracy of an analog-to-digital converter. Basically, they measure how close the threshold levels for each bit are to the theoretical equally-spaced levels.

Unquantifiable?

Some audiophiles have postulated that the present set of audio measurements, as exemplified by the above list, does not fully represent all that is significant in accurate music reproduction, and instead represents only those aspects which are relatively easy and cost-effective to measure with our current technology. Given the complexity and sophistication of human hearing and perception, it is felt that some consideration should be given to the possibility that there may be aspects of music reproduction that have yet to be identified.

All of the above measurements are quantitative, not qualitative. Subjectivists claim that listening tests are more appropriate for appraising the quality of an audio system than measuring the accuracy with which it can reproduce a waveform.

See also

• Audio quality measurement
• High fidelity, especially the section on double blind tests
• Audiophile
• Physics of music

References

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External links

• Explanation of INL/DNL measurements