Professional radio broadcasting should aim to make high quality sound productions and transmissions.

Transcription

1 Karl Lippe, UQP1-neuformat.doc, page 1 QUALITY PARAMETERS OF AUDIO SIGNALS 1. INTRODUCTION Professional radio broadcasting should aim to make high quality sound productions and transmissions. Quality in audio broadcasting has two aspects: 1. A sound production should represent the original intention of the artist (writer, composer, interpreter etc.). It is the task of the production team to skillfully use the studio equipment so that this aim is achieved. 2. Pre-recorded material should be reproduced in such a way that it is identical to the original recording. The "sound" of the recording should not be changed. Any undesired change to the original recording is considered a loss in sound quality. The professional broadcast equipment normally used must comply with the highest sound quality standards and is therefore very expensive. This course will deal with the second item - the technical aspect of sound quality. The first item would be the focus of a course for producers. The sound quality obtained in the broadcasting studio is for the benefit and the pleasure of thousands or even millions of listeners. Some listeners make considerable investments in expensive radio sets to make full use of this sound quality.therefore all broadcasting personnel should make sure that the quality of the equipment is fully used. All broadcasting personnel is involved in maintaining the sound quality of radio productions although their roles may be different. Some of their concerns are listed as follows: Producers/Announcers: Operators: Maintenance: - operate the equipment properly, - identify poor sound quality, - report problems to the maintenance department. - operate the equipment properly, - identify poor sound quality, - report problems to the maintenance department. - communicate with the operators and producers, - identify poor sound quality, - maintain the equipment properly.

2 Karl Lippe, UQP1-neuformat.doc, page 2 Although the duties of the three groups are different, there are some very important abilities, required of all of them: - To be aware of the demand for high sound quality. - To be able to identify deficiencies in sound quality. - To have the basic knowledge to properly operate the equipment. - To be able to communicate with others about sound quality problems. It is the aim of this course to train all groups of broadcasting personnel in the skills listed above. In order to be able to communicate with others about sound quality, it is necessary to get acquainted with the terms used to describe sound quality. These are the sound quality parameters. Although these are technical or physical terms, non-technicians will also have to use them in order to have a common vocabulary to discuss sound quality problems. It is not mandatory, however, for everyone to have a fully understand the complex technical background for all of these terms. This course will therefore only deal with the basic definitions of these terms. The number of sound quality parameters used in broadcasting is limited. Some people would argue that more parameters are necessary to fully analyse and describe the quality of some equipment. In order to keep the matter understandable and also to keep the maintenance procedures within financial constraints, the following parameters should be sufficient: - Signal level - Frequency response - Distortions - Noise - Wow and Flutter Sometimes different terms are used for some of these parameters. This course will deal with each parameter in one chapter, discussing the practical and the theoretical aspects. The knowledge contained in this course should be mandatory for producer and operators. Maintenance technicians or engineers need to deal with this subject more in detail.

3 Karl Lippe, UQP1-neuformat.doc, page 3 2. THE SIGNAL LEVEL Signal level is not always considered to be a quality parameter. However, we have to realize that for the listener, the quality of the reception will depend strongly on the proper setting of the levels in the studio and at the transmitter. Therefore properly balanced levels are the first requirement for good sound quality. The signal level is the only quality parameter which can be controlled during operation in the studio. Proper control of the level is the most essential skill when being trained in studio operation The Audible Effects of the Signal Level The correct signal level must be controlled during mixing, recording, play-back and transmission of radio productions. During any of these processes the following mistakes can occur: - The relation of levels of two or more signals being mixed are not set correctly, so that the signals do not appear in proper relation to each other. - Levels between two takes or between music and announcement are not set properly, so that the listener experiences an unpleasant change in volume. - An entire programme or recording is set at a level too high or too low, so that the technical limits of the equipment are exceeded. Furthermore the signal level has an effect on other quality parameters, namely the signal-to-noise ratio and the distortions. - If the signal level is too low, the signal-to-noise ratio will be reduced, - If the signal level is too high, the distortions will increase. The reasons for these effects will be discussed in later chapters. It is especially important that the level must not only be set correctly at the output of the studio, but must be properly set for each stage of the signal chain. Therefore, correct setting of levels of each piece of equipment is essential. The correct setting of the transmitter levels has other aspects, as well: - Setting levels too low, i.e. under-modulating the transmitter, is a considerable waste of energy, as it causes a reduction of the service area.

4 Karl Lippe, UQP1-neuformat.doc, page 4 - Setting levels too high, i.e. over-modulating the transmitter, leads to clipping and distortion or even tripping of AM-transmitters. FM-transmitters will produce radiations which violate radio regulations. The human ear is an unreliable instrument to judging correct signal levels. - It cannot accurately detect sound level. It is impossible for somebody to tell if two signals gave the same level, unless they occur immediately after each other. - The human ear is not "calibrated". We cannot tell the signal level just by listening, nor can we tell the level difference between two signals, occurring at the same time or after each other. - The human ear is relatively insensitive to changes in signal level. A change of approximately 10% is necessary to detect an instantaneous change in level. From this we can make two conclusions: - We need an objective means to measure the signal level. This is the programme meter for the studio operator or the level meter for the maintenance technician. - Audio signals need not to be measured with high precision. A precision of ±10% is sufficient. The technical quantities used to describe signal level take these facts into account Different ways to define signal levels. Acoustical signals are difficult to handle. Sound is therefore converted as soon as possible to electrical signals by the microphone. Signals in the studios are therefore electrical signals. These can then be amplified, recorded and transmitted. Electrical signals are described by electrical quantities. Audio signals are always alternating signals, in other words, varying signals. To describe these signals, we have to apply special theories and mathematics. Basically, audio signals can be described by their effective or RMS voltage and their frequency. The typical signals in the studio, produced by music or speech, do not consist of one amplitude and one frequency, but of constantly varying values, and even of many voltages and frequencies at the same time.

5 Karl Lippe, UQP1-neuformat.doc, page 5 For testing the equipment and getting a clear and definite result, it is necessary to use a signal having one constant amplitude and frequency. This results in the definition of test signals. Examples of test signals: U = 1.55V, 1kHz or U = 0.1V, 3150Hz The adaptability of the human ear to different sound levels and a wide range of levels makes it useful to have a logarithmic scale for signal levels. Because the signals in the studio are available as electrical signals, a relationship has to be defined between signal voltage and signal level. This is done by defining the DECIBEL (db). The db is a relative quantity with a logarithmic characteristic. Many different forms of db are in use depending on the reference quantity. An additional letter is used to define which reference signal is used. In audio electronics we mainly use db, dbr and dbu (formerly dbm). The db without any defined reference is only used to describe the relationship of two signals. It cannot be associated with a certain signal level or voltage. db is used to express gain or attenuation: U G(dB) = 20 log Ł U out in ł dbr is used to express absolute signal levels relative to a given standard. The standard must be defined somewhere or must be commonly agreed upon (e.g. the standard studio level). L(dBr) = 20 log Ł U U ref ł Often the dbr is simply called db. (E.g. "This recording is 10dB too low.") Strictly speaking, this is incorrect. dbu is used to express absolute signal levels. The reference used is an absolute voltage. The value of this reference voltage is defined historically. It is the voltage which will produce a power of 1mW in a 600Ω resistor. (Formerly the term "dbm" was used for this purpose.) ( 1mW 600W) 0.775V U ref = =

6 Karl Lippe, UQP1-neuformat.doc, page 6 Any signal level can then be determined by the following formula: Thus L (dbu) = 20 log U 0.775V 0 dbu = 0.775V Any other signal voltage can be calculated, e.g.: -4dBu=0.49V ; -2dBu=0.62V ; +6dBu=1.55V ; +8dbu=1.95V ; +20dBu=7.75V Level meters used for maintenance and calibration of audio equipment are normally calibrated in dbu or volts/millivolts. Either scale can be used for measurement purposes, but in broadcasting electronics it is more common to use the dbu-scale. The signal level does not need to be measured with high precision. It is normally sufficient to measure with a precision of 1dB, which means a tolerance of approximately 10%. Therefore, calculations can also be done in a rather rough way. Often the use of the formulas, which would require the use of a calculator, can be avoided if a few common relationships are memorised. The following table shows the relationship between the most important values for signal ratios and db: db factor With these values most db-values or ratios can be converted by simple calculation. Examples: from db to ratio: 64dB = 60dB + 10dB - 6dB 1000*3/2 = 1500 (precise: 1585) 5dB = 17dB - 6dB - 6dB (7/2)/2 = 1.75 (precise: 1.78) or 5dB = 6dB - 1dB 2 minus 10% = = dB = 20dB + 10dB = 10 *3 = 30 (precise: 31.62)

7 Karl Lippe, UQP1-neuformat.doc, page 7 from ratio to db: ratio is 50 ~ 7 * 7 17dB + 17dB = 34dB (precise: 34dB) or 50 = 100/2 40dB - 6dB = 34dB ratio is 18 = 2 * 3 * 3 6dB + 10dB + 10dB = 26dB (precise: 25.1dB) ratio is 1.6 = 16/10 = 2*2*2*2/10 4*6dB - 20dB = 4dB (precise: 4.1dB) The Studio Level When talking of levels during production and transmission of programmes, the "normal" or maximum level is referred to as 100% or 0dB. This is a relative level. It is therefore necessary to define the reference or the standard studio level. This will then be an absolute level (in dbu) or voltage. Of cause, during normal operation of the studio, it is not necessary to know the standard studio level. The maintenance technicians must ensure that all equipment is aligned to this level. In Germany, the standard studio level is +6dBu = 1.55V. Other countries have other levels between +4dBu and +12dBu. The actual value is not important, but it is vital that only one standard studio level exists for the entire broadcasting house. Of course, it is useful to adopt some national or even international standard in this respect, and it is also useful for non-technicians to know this value Dynamic Range. The normal programme material in broadcasting produces varying levels. This means a range has to be defined in which the signal is allowed to vary. This is called the dynamic range. The dynamic range is limited towards the high levels and the low levels by restrictions given by the equipment. High signal levels are limited by the maximum signal amplitudes the equipment can handle. Exceeding this limit results in clipping and distortion of the signal. Low signal level should still stand out against the physical noise produced by all equipment. Therefore the "noise level" of the equipment is the lower limit for the signal.

8 Karl Lippe, UQP1-neuformat.doc, page 8 Fig Graphical representation of the dynamic range of a tape machine. The normal maximum signal is the standard studio level (0dB).The head room is used to prevent accidental over-modulation, which leads to clipping. The foot room is the required level difference between the lowest possible level and the noise level. It is required to prevent the noise from disturbing low volume passages of the programme. The normal dynamic range of broadcasting programme material should not exceed 40dB. During production and transmission, constant control of the signal level is essential at all stages in order to keep the signals within the optimum range of the equipment. Overmodulation particularly produces very nasty effects and is annoying for the listener Programme Meters (level Meters) Different abilities are required of level meters and different solutions have been developed to meet them. Although a meter cannot be used to judge things such as the balance of one voice against another, or speech against music, it is nevertheless an essential item of studio equipment.

9 Karl Lippe, UQP1-neuformat.doc, page 9 The demands placed on programme meters can be devided in two groups: 1. The objective demands to make optimum use of the equipment: - to provide indications of levels which would result in under- or over-modulation at the recorder or transmitter, - to check that the same signal level is used for studio, recording room, and transmitter, 2. The subjective demands for good listening: - to compare relative levels between one performance and another - to reduce the dynamics of natural sound events (about 110 db) to the dynamic range of broadcasting (40 db). There are several types of meters which can be used to line up equipment or check for over-modulation: - the peak-meter or peak programme meter (PPM) (used almost all over Europe) - the VU-meter (used in America) The peak-meter Correctly speaking, this is a quasi-peak meter. It consists of a full wave rectifier circuit followed by a logarithmic amplifier and an indicating device, calibrated in rms-values. Peak reading means: a 10ms full level sine wave results in 90% (-1dB) indication of a constant tone of equal amplitude. A pulse of 3ms results in an indication of -4dB. This is the attack time or integration time. The recovery time or fall-back time from 0 db to -20 db (or from 100% to 10%) is about 1.5s to 2s. The scales of the different indicator instruments (meters) are almost logarithmic to facilitate the reading, since one wants to stretch the upper part in the total range of - 50dB to +5dB, (or 0,3% to 180%, corresponding to an amplitude ratio of 1 to 600). In order to better recognize the overload limit during operation, the range is stretched between -6dB and +5dB, corresponding to 50% till 180%. The accuracy of the indication is ±1dB between -5db and +5dB and ±2dB between -40db and -5dB.

10 Karl Lippe, UQP1-neuformat.doc, page 10 Different types of instruments can be used for indication: - Light spot meter, formerly one of the most commonly used instruments. - Bar graph meter with neon discharge elements. Two independent measuring amplifiers and indicating bars make it useful for stereo. The bars are composed of up to 100 segments each, which give the impression of continuous bars. The overload range (0 to +5dB) will illuminate at higher intensity. At present, it is one of the most common peak meters. - LED-meter with up to 70 LED's. Amplifier and display are in one unit. - Pointer instrument with measuring amplifier and moving coil instrument in one unit. Often the instruments have its mechanical zero at the 0dB-point or at the right end of the scale. Different types of Peak Programme Meters: The display of a bar-graph PPM with discharge display. A bar-graph PPM using LED. A pointer type PPM with amplifier.

11 Karl Lippe, UQP1-neuformat.doc, page 11 Requirements for Peak Program Meters German versions: Scale: Rise time (integration time) to -1dB (90%) reading: -50dB to +5dB 10 ms Fall-back time (return time) after 100 % reading: 1,5 s to -20 db (10%) 2,5 s to -40 db (1 %) Overshoot: +0,3 db Peak programme meter - BBC-specification The BBC version of the peak programme meter is a pointer instrument. The scale has divisions from 1 to 7. Each division represents 4dB. The standard level (100 % modulation) is +8dBu at reading 6. A pulse of 4ms produces an indication of 80% of the full deflection. Fallback time is about 8,7 db/second. The VU-meter: A more simple type of meter than the PPM is the American VU-meter. It was standardized since more than 40 years ago in the USA and is composed of a very sensitive moving coil instrument with full wave rectifier and preceding damping network. Specifications: A 300ms voltage pulse results in a 90 % indication. The recovery or fall-back time and attack or integration time are equally 300ms. The meter has two scales, one is calibrated in decibels from -20VU to +3VU and is approximately logarithmically linear (more than half of the scale is occupied by a range of 3dB on either side of the nominal 100% modulation). Since little programme material will remain consistently within such a narrow range, the needle generally either registers only small deflections or flickers wildly over the full range of the scale. This scale is intended for use with steady tone for line-up. The other scale is calibrated in volume units (VU), numbered from 0 to 100 over about two thirds of its range, with the last part uncalibrated and marked in red. This scale represents percentage of full modulation and is intended for use on programme material. A damping network increases the input resistance to 7,5 kohm.

12 Karl Lippe, UQP1-neuformat.doc, page 12 Different programme material leads to different indication behaviour, requiring some experience for exact control. Although the VU-meter is cheaper and lighter and doesn't need either a power supply or maintenance, it's disadvantages compared to the PPM, cannot be overlooked: - low sensitivity, - limited range, - long attack time which underrates short peaks. VU-meters require a so-called lead. This means the VU-meter is calibrated to indicate higher signal levels than are really there, to compensate for its inability to react immediately on varying levels of normal programme material. Different programme material (speech, classical music, pop music) will require different leads, ranging from 4dB to 10dB. Some VU-meters allow adjustment to the lead according to the requirement of the programme. It should be set according to the experience of the operator. Most VU-meters have fixed lead settings, e.g. 6dB. When constant test tones are applied to VU-meters, they will indicate a value which is higher than the true signal level (higher by the lead). Therefore VU-meters cannot be used to do measurements or service alignments in the studio. Fig Scales of different VU-meters. Conclusion: The functions and characteristics of the two types of meters are very different. Rather than one being better than the other, they complement each other nicely. Indeed, a good engineer should be conversant with both types of meters and be able to use them to advantage in order to get the most out of the equipment.

13 Karl Lippe, UQP1-neuformat.doc, page 13 The VU meter displays a value which is more or less proportional to the loudness of a sound as perceived by the human ear. It is thus a kind of volume (loudness) indicator. The advantage of using a VU meter for mixing audio signals is that it gives a visual indication of the acoustic conditions being experienced by the listener. However, audio signals are neither symmetrical nor periodic (over a short period of time) and thus a VU meter is incapable of delivering reliable information as to the exact signal level present (volts or dbu). In contrast, the PPM is a more scientific instrument which is basically accurate when it displays the peak level of a signal over a given period of time (200ms). It's rise time of 10 ms is short enough to ensure the correct measurement of most transients. The main advantage of the PPM is that it allows the operator - in recording or broadcasting - to obtain the maximum performance (in terms of signal- to-noise ratio and headroom) of the equipment. The principle drawback of the PPM is that the indication has little or nothing to do with the audio volume experienced by the listener Psycho-acoustical effects of the human ear. As priviously stated, the human ear is not a calibrated measuring device. There are a number of physiological characteristics which have to be considered. Some of them will be discussed here. Equal Loudness Contours The human ear will not recognize the sound levels of different signals equally. Experiments have shown that frequencies around 4kHz are recognized with a much higher intensity than very high and very low frequencies. Fig The typical curves of equal intensity for different frequencies of the human ear.the curves will differ for different sound intensity. At low intensity, the difference in intensity is very large, at very high levels the difference is minimal. This relationship will be of some importance when discussing unwanted signals (noise).

14 Karl Lippe, UQP1-neuformat.doc, page 14 Amplitudes of Sound Fig A curve showing the amplitude statistics of natural sound. Note that these are statistical values and it can well occur that some sound has a different amplitude distribution. If we measure the amplitudes of all signal frequencies of normal sound (programme material) we will find that different frequencies occur at different level. As frequencies and levels are constantly changing, it is necessary to make statistical evaluations for different sound (music, speech, natural sound). The following curves show the resulting amplitude statistics. The curve shows that the amplitudes at frequencies below 200Hz and above 1kHz are getting smaller. At the each end of the audio band they are 20dB less than in the centre. For the audio equipment, this means that some of the signal frequencies require less amplitude or power than the centre frequencies. For example, the tweeter of a loudspeaker normally has to handle less power than the medium frequency loudspeaker. (Therefore, use caution during tape cueing: cueing produces an extraordinary amount of high frequencies, which can easily kill the tweeters of the monitor boxes. Reduce the volume during cueing!) Masking Effect The equal loudness contours are only relevant if just ONE signal is tested. If several signals are present at the same time, the stronger signals will cover or mask the weaker ones. The closer the frequencies of the two signals are together, the stronger this effect will be.

15 Karl Lippe, UQP1-neuformat.doc, page 15 Fig The curve for the audibility of a signal changes if other signals are present, masking the signal being tested. The closer the frequencies of the two signals are, the stronger is the masking of the weaker signal. The masking effect is used in modern digital signal processing for the purpose of data reduction. 3. FREQUENCY RESPONSE The frequency response of audio equipment is the parameter, which describes how well the equipment is able to reproduce the original sound. The frequency response must satisfy the requirements of the human ear, but also has to consider the technical possibilities of the studio equipment. The general requirement regarding the frequency response of equipment is that it represents all required audio frequencies equally. We say its frequency response is linear. Deviation from this ideal condition is referred to linear distortion The Audible Effects of the Frequency Response. Generally it is said that the human ear recognizes "sound" in a range from approximately 20Hz to 20kHz. But not all people can hear this entire range. Especially in the upper frequency range many people can hear less. The following graph represents people`s audible skills. Towards frequencies beyond 14kHz there is little sound information, recognized by only few people. Fig This diagram represents the audio frequency range, the density of the lines representing the number of people hearing signals at different frequencies.

16 Karl Lippe, UQP1-neuformat.doc, page 16 This shows that it is not necessary and also not useful to transmit frequencies below 40Hz and above 15kHz in an audio system. It is more important that all studio equipment can handle all signal frequencies of this range with equal gain. This will maintain the original relationship between the frequencies and will reproduce the original sound. This is then referred to as "high fidelity". The possibilities of audio equipment THE POSSIBILITIES OF AUDIO EQUIPMENT Audio signals cover approximately 8 octaves, which is a very wide range. For some audio equipment it is difficult to cover this range. The requirements placed on the equipment must therefore take into account the technical possibilities. Example: We must except that a small cassette recorder cannot produce the same sound quality as a studio tape machine. Therefore, we will have different strict requirements regarding the frequency range of different equipment. Fig Different frequency ranges for different equipment. The ranges depend on the technical capability of the equipment Definition of the frequency response In general the studio equipment should handle all signal frequencies with equal gain. The common method for representing this is to draw a graph showing the gain or output level versus the frequency. This is then called a frequency diagram of the frequency response. Logarithmic scales are normally use for the frequency and the amplitude axis.

17 Karl Lippe, UQP1-neuformat.doc, page 17 The frequency response produced by ideal equipment is represented by a straight line. We say the frequency response is linear. Practical equipment will often not be able to produce an ideal linear frequency response. Fig Example of a non-ideal frequency response from equipment. In order to judge the quality of the frequency response, we give tolerance limits to the response. If the curve remains within the tolerance limits, it is good, if it exceeds these limits, it is bad. The tolerance limits require a reference. This reference is normally the level of the output signal at some centre frequency of the total frequency range. This is normally 1kHz. The gain or the level at 1kHz is referred to 0dB. 0dB is the reference point for all other frequencies. Fig The frequency response of equipment within the tolerance limits. The response of this example is good.

18 Karl Lippe, UQP1-neuformat.doc, page 18 Often the tolerance is allowed to increase (roll off) at the lower and the upper end of the range. This is because - the lower and the upper end of the frequency range is not very important for a good sound quality, - in these ranges, the equipment may have special problems maintaining the linear response. The higher the quality of equipment, the more narrow is the tolerance scheme. In equipment data sheets, the tolerance scheme for the frequency response is often given by the intersecting points. E.g. the tolerance scheme for the above curve can be given as: 100Hz to 10kHz: ±1dB 40Hz to 15kHz: +1dB, -2dB The studio equipment most likely to produce linear distortions is the recording equipment - mainly tape machines and cassette recorders. Regular and careful service is necessary to maintain a satisfactory frequency response: - Clean the heads, the tape guides and the capstan frequently, - Service and align the machine regularly, - Use the tape lifters during spooling. 4. DISTORTION Correctly speaking, this is called "non-linear distortion", to distinguish it from the linear distortion, but the term "distortion" is commonly used. Distortions are some of the nastiest products of poor sound quality. If they are audible, they are always an indication for - grossly incorrect operation of the equipment, - totally misaligned equipment, - defective equipment. In general it can be said that

19 Karl Lippe, UQP1-neuformat.doc, page 19 The presence of audible distortions in the studio is always an indication for a dramatic error and requires immediate action. In fact, every piece of equipment in the studio produces a bit of distortion. Under normal operation conditions, this should remain at such a low level that it will not be audible. The limit where distortions become audible is between 1% and 3%, depending on the sensitivity of the listener and the kind of music. Most sensitive to distortions are pure tones such as test signals or piano tones. Orchestral, band music or speech are less sensitive to distortions. The following list shows the distortions which can be expected from different type of audio equipment: - microphones: % - mixers, amplifiers 0.1% - 0.5% - record players 1% - tape machines 0.5% - 1% - cassette recorders 1% - 5% - transmitters 1% - receivers 1% - loud speakers 1-5% The distortions produced by any of these types of equipment will increase dramatically as soon as the maximum level for the equipment is exceeded. When clipping is reached, the distortions may reach up to 30%. Therefore, most studio equipment has a safety margin for excessive levels, the so-called head room. Different equipment has different headrooms: - mixers, amplifiers 15dB - 20dB - tape machines 6dB - cassette recorders 6dB - transmitters 0dB This table shows that especially transmitters and tape recorders are especially sensitive to over-modulation. When sending signals to any of these devices (recording studio, continuity) greatest care must be taken to keep the level within the required limits. Peak programme meters should to be used to measure the signal levels sent to such equipment.

20 Karl Lippe, UQP1-neuformat.doc, page Definition of Distortion. The distortion of signals is always caused by some equipment. When distorting the signal, the equipment is adding an additional signal to it. In terms of physics, this new signal will always have frequencies, which are plain multiples of the original signal. These are called harmonics. We can therefore say that distorting a signal produces harmonics. Depending on the kind and the cause of distortions, different harmonics will be produced. Basically we distinguish between symmetrical and asymmetrical distortions. In symmetrical distortions, the positive and the negative half-waves of the signal will be distorted equally. In asymmetrical distortions, the positive and the negative half-wave of the signal will be distorted differently. - Symmetrical distortion will produce only odd harmonics, (e.g. only 3nd, 5th, 7th, etc.) - Asymmetrical distortion will produce odd and even harmonics, (e.g. 2nd, 3nd, 4th, etc.) Fig Graphical representation of a differently distorted sine wave. a.) Symmetrical distortion of the sine wave (clipping). The spectrum shows the strong amplitude of the fundamental frequency and odd harmonics. b.) Asymmetrical distortion of the sine wave. The spectrum shows the fundamental frequency and odd and even harmonics.

21 Karl Lippe, UQP1-neuformat.doc, page 21 The definition of harmonics is therefore rather simple; we just have to determine how many harmonics the equipment produces. If more than one harmonic is produced, all of them must be added geometrically. Their total amplitude is related to the amplitude of the original signal. The ratio is then given in percent or db. Distortion(%) = ( U + U U ) f2 f3 U f1 fn 100% Distortion(%) Distortion(dB) = 20 log Ł 100% ł U f1 is the voltage of the original signal with its fundamental frequency; U f2 to U fn are the voltages of the harmonics with multiples of the fundamental frequency. The distortions are measured with special distortion measurement equipment in the following way: A pure sine wave with no harmonics is applied to the input of the equipment under test. The frequency is normally 1000Hz and the level should be the normal operation level for the equipment (e.g. standard studio level). At the output, the signal is measured with a filter, separating the fundamental frequency from the harmonics. The level of both are determined and defined in relationship to each another. This is done either by means of suitable scales or, with modern equipment, by microcomputers. Fig The test set-up to measure the distortion of a piece of equipment.

22 Karl Lippe, UQP1-neuformat.doc, page NOISE Noise is defined as any signal which is added to the original signal. Noise is not entirely an electronic problem. Noise can have -acoustical causes: -mechanical causes: -magnetic causes: -environmental causes: -electronic causes: background noise during a recording floor noise rumbling of a record player dusty or scratchy records magnetization noise of the tape copy effect crackling of the receiver due to lightning, Hum due to mains interference, electromagnetic interference on cables. thermal noise in components, interference between components, hum from power supply. All kinds of noise produce different undesired effects, and different preventive measures are necessary to avoid or reduce them. Excessive noise can be caused by defective equipment, as well as by improperly installed and operated equipment. Therefore, maintenance technicians as well as operators have to do their part in preventing noise. The cause of the noise can often be concluded from its sound: hiss: hum: tape noise, amplifier noise (thermal noise), signal level too low. mains interference, grounding or screening problems, unbalanced inputs, defective power supply. crackling: dirty or scratchy record, contact problems, dirty potentiometers. It lies within the skill and the experience of the technical staff to identify the causes of noise and to decide about suitable measures. It should be considered that there is always a certain amount of "natural" or physical noise which cannot be avoided. The studio equipment must be aligned and operated in such a way that the noise remains inaudible under normal operating conditions. One way to create excessive noise during a production is to operate with inadequate levels. While levels above the standard studio level cause distortions, levels below the studio level may also allow the noise to become audible.

23 Karl Lippe, UQP1-neuformat.doc, page 23 Fig The level diagram of a studio chain. The level in the tape machine is too low, so the signal-to-noise ratio is reduced. Although the low level is boosted again in the ampliflier, the noise is amplified at the same time and the signal-to-noise ratio remains poor Different ways to define noise. In professional audio electronics noise is defined by its level. Any audio level meter with sufficient sensitivity (down to -60dBu or -80dbu) can be used. The CCIR standard requires that the meter has a quasi-peak characteristic. The noise level of equipment is determined by measuring the signal level at the output of the equipment with no signal applied to the input. As there is no useful signal at the input, the signal at the output must be "unuseful": noise. Fig Measuring set-up for measuring the noise level of equipment. The unweighted noise level measured in this way is given as unweighted noise level: db qs q: indicating the use of a quasi-peak meter, s: indicating the sound range from 20Hz to 22kHz.

24 Karl Lippe, UQP1-neuformat.doc, page 24 In order to give a figure for the noise which is representing the subjective "inconvenience" to the listener, the noise can be judged or weighted, according to the sensitivity of the human ear. In order to do so, the noise signal is passed through a socalled weighting filter. The filter characteristic is standardized. Unfortunately there are several standards. For professional use, the CCIR standard is commonly used. For consumer equipment the so-called "A" weighting filter is preferred. This normally gives better (lower) noise values. Fig The frequency characteristic of a weighting filter according to a CCIR468-3 and an "A" weighting filter. The weighted noise level measured in this way is given as weighted noise level: db qps q: indicating the use of a quasi-peak meter, p: indicating a weighted measurement (french: pondéré) s: indicating the sound range from 20Hz to 22kHz. Weighted and un-weighted noise measurement will produce values which may differ up to 10dB. Using different types of weighting filters may produce values which can differ up to 20dB. Be careful when comparing noise figures from different manufacturers. Ask for the standard used and only compare equal standards. Another quantity used to describe the "noise quality" of some equipment is the signalto-noise ratio. The signal-to-noise ratio is defined as the ratio between the noise level and the maximum operation level of equipment. Example: Noise level measured: -62dBqps maximum signal level: +6dBu signal-to-noise: S/N = 68dB

25 Karl Lippe, UQP1-neuformat.doc, page 25 The noise level is a characteristic of the equipment, while the maximum operating level is defined by the standard of the broadcasting station. Therefore, The signal-to-noise ratio is not a constant for some equipment. It is just a helpful hint to the operator when judging the dynamic range available. Furthermore, the manufacturers of some consumer equipment tend to calculate the S/N-ratio not by relating the noise level and the maximum operation level, but by relating the noise level and the clipping limit. This means they add the head room of the equipment to the S/N-ratio. Example: Noise level measured: maximum signal level: clipping level: -62dBqps +6dBu +14dBu signal (clipping level) to noise: 76dB Fig Graphical representation of different S/N-ratios for different operation levels. Note that the noise level is the same in all cases. The impressive quality parameters sometimes given on consumer cassette recorders are often calculated this way.

26 Karl Lippe, UQP1-neuformat.doc, page WOW AND FLUTTER. Wow and Flutter is a deformation of the original sound, which can only occur if mechanical recording was involved. Therefore it is a problem which only concerns tape and cassette machines and record players The Audible Effects of Wow and Flutter Wow and flutter produces some of the nastiest deformations of sound. The typical example occurs with poor cassette recorders. The listener experiences a change in frequency of the original signal. The English terms "wow" and "flutter" describe the characteristics very well. If it is a slow change (less than ten per second), we call it wow. If it is a fast change (more than ten per second), we call it flutter. Distinguishing wow and flutter is useful because they often have different causes. wow: flutter: - sticky and dirty tape guides, - incorrect tape tensions, - incorrect brake adjustments, - incorrect pinch roller pressure, - moist and sticky tapes. - dirty capstan, - dirty pinch roller, - bearing problems, - worn-out idler. The list indicates, that wow and flutter problems are normally caused by a lack of or incorrect maintenance. Often it is not caused by a defect, but can be avoided by cleanliness and proper servicing of the equipment. The operators and producers can contribute to the required cleanliness of the equipment by keeping away coffee, tea, soft drinks and cigarettes. It should be noted that all analog recording equipment will produce a small amount of wow and flutter. Under normal operating conditions this should be so little that it is inaudible. Audible wow and flutter is always an indication of a serious technical problem and requires immediate action.

27 Karl Lippe, UQP1-neuformat.doc, page Definitions of Wow and Flutter. Technically, wow and flutter are frequency modulations of the original signal. The frequency deviation defines the intensity of wow and flutter and the modulation frequency defines the frequency of wow and flutter. To measure wow and flutter, a stable frequency is recorded. The frequency deviation is then measured during play-back. The peak frequency deviation is related to the recorded frequency, and the relationship is given in per cent. The human ear is most sensitive to wow and flutter at frequencies around 3kHz. The test frequency for wow and flutter is standardized by CCIR to 3150Hz. (3000Hz for NAB). To measure wow and flutter, test tapes and test records are available with this special test frequency. Fig Wow and flutter produce a frequency deviation from the original signal. The test signal is normally 3150Hz. The value of wow or flutter is determined by the ratio of peak value of the frequency deviation to test frequency. The human ear recognizes frequency deviations differently, depending on the frequency at which it occurs. We will find a frequency deviation the most annoying if it occurs with a frequency of 4Hz. Higher or lower modulation frequencies are less irritating. Therefore, wow and flutter is weighted with a frequency response which simulates the sensitivity of the human ear for wow and flutter. Fig The weighting curve for wow and flutter according to CCIR The curve has a maximum at 4Hz and produces less sensitivity towards higher and lower frequencies.

28 Karl Lippe, UQP1-neuformat.doc, page 28 Wow and flutter can only be measured with special wow and flutter meters. The meters contain a generator, producing the test frequency. The analysing circuit consists basically of a FM de-modulator, a weighting filter and an indication device. When no wow and flutter test tape is available, a recording should be done with the test signal (3150Hz) from the wow and flutter meter. The tape is then reproduced while the output signal from the tape machine is fed to the input of the wow and flutter meter. The meter will indicate the value. If the indication fluctuates, the average value is considered. Fig The set-up for wow and flutter measurements: First, the test signal (3150Hz) is recorded on tape. Then the tape is reproduced and the signal is analyzed by the wow and flutter meter. Wow and flutter meters are also able to measure the speed drift (deviation). Drift produces not a varying, but a constant change in frequency. One can also say that drift is a wow of very low frequency. Normally, a constant speed deviation produces no audible effects. But it indicates a problem in the drive system of tape machines and record players. It is therefore useful to measure it during maintenance. Using the vary-speed of tape machines or record players will produce drift on purpose.