Sound System Reference Manual

Public Address

Sound Reinforcement

Bosch Security Systems

Sound System Reference Manual Introduction
Sound systems are used for amplification of speech or music to enhance intelligibility
or loudness by electro-acoustic means in order to serve an audience with a higher degree of listening comfort. A public address distribution system is designed primarily to carry live and recorded messages, signal tones and background music (if required), from several different sources, to a number of selectable remote areas. Common applications would be: hotels, restaurants, railway stations, airports, factories, oil platforms, office buildings, schools, shopping areas, ships, exhibition areas, etc. A sound reinforcement system would normally be used to reproduce live voice (and often music) to a number of people who are generally located in the same room or area as the signal source. Typical applications are churches, lecture halls, political gatherings, conferences, etc. Sound system design is a comprehensive subject combining a chain of devices: microphone, sound processing equipment, amplifier and loudspeaker, together with the acoustic environment, into a single system. The microphone converts the acoustical vibrations, caused by an audio source, into an electrical signal. The processing equipment modifies the signal to compensate for deficiencies in the source or environment. The amplifier increases the level of the signal to one adequate for driving loudspeakers. The loudspeakers convert the electrical signal back into vibrations, which are greatly influenced by the acoustic environment, and in turn received by the ear of the listener. This manual is intended to give readers with a technical background a reference to the various aspects of audio engineering and sound system design.

TABLE OF CONTENTS
SOUND - THE THEORY...................................................................................................................................................... 1 1.0 BASICS............................................................................................................................................................................ 1 1.1 Speech........................................................................................................................................................................ 3
1.1.1 Dynamic Range ..................................................................................................................................................................... 3

1.2 Music ......................................................................................................................................................................... 4
1.2.1 Dynamic Range ..................................................................................................................................................................... 4 1.2.2 Musical Range versus Frequency .......................................................................................................................................... 5

1.3 Sound......................................................................................................................................................................... 6
1.3.1 Ear-characteristics.................................................................................................................................................................. 6 1.3.2 Weighting .............................................................................................................................................................................. 7 1.3.3 Sound Pressure Level............................................................................................................................................................. 7

1.4 Sound Propagation in Air........................................................................................................................................... 8 2.0 DECIBEL NOTATION........................................................................................................................................................ 9 2.1 Definition................................................................................................................................................................... 9
2.1.1 2.1.2 2.1.3 2.1.4 Logarithmic characteristics of the ear.................................................................................................................................... 9 Power ratios ........................................................................................................................................................................... 9 Voltage ratios....................................................................................................................................................................... 10 dB references ....................................................................................................................................................................... 10

2.2 Calculations............................................................................................................................................................. 11
2.2.1 Addition and subtraction...................................................................................................................................................... 11

THE SOUND SYSTEM ....................................................................................................................................................... 12 3.0 AN INTRODUCTION ....................................................................................................................................................... 12 3.1 Functional requirements ......................................................................................................................................... 12 MICROPHONES ................................................................................................................................................................. 13 4.0 MICROPHONES .............................................................................................................................................................. 13 4.1 Considerations when Selecting a Microphone ........................................................................................................ 13 4.2 Microphone Types ................................................................................................................................................... 13
4.2.1 4.2.2 4.2.3 4.2.4 4.2.5 Electrodynamic .................................................................................................................................................................... 13 Condenser ............................................................................................................................................................................ 13 Back Plate Electret............................................................................................................................................................... 13 Electret................................................................................................................................................................................. 13 Choices ................................................................................................................................................................................ 13

4.3 Pick-Up Response Patterns ..................................................................................................................................... 14
4.3.1 Omnidirectional ................................................................................................................................................................... 15 4.3.2 Cardioid ............................................................................................................................................................................... 15 4.3.3 Hyper-cardioid..................................................................................................................................................................... 15

4.4 Special Microphones ............................................................................................................................................... 16
4.4.1 The Lavalier and Lapel microphone .................................................................................................................................... 16 4.4.2 Noise cancelling microphone............................................................................................................................................... 16 4.4.3 Radio (Wireless) microphone .............................................................................................................................................. 16

5.0 TECHNICAL PRINCIPLES ................................................................................................................................................ 17 5.1 Directivity................................................................................................................................................................ 17 5.2 Sensitivity................................................................................................................................................................. 17 5.3 Installation Considerations ..................................................................................................................................... 17
5.3.1 Potential problems and causes ............................................................................................................................................. 17 5.3.2 Solutions .............................................................................................................................................................................. 17

6.0 MICROPHONE TECHNIQUE ............................................................................................................................................ 18

..............................................2 Tone controls ......................................36 14....................................................................................................................................................................................................................................... 21 8............................................................1..2......................................... 45 Full range high power loudspeakers............................................................................................1 13................3 Amplifier/Loudspeaker Interface.........................................................................................2 9.......36 14.........................................1 Earthing (grounding).................1...........................................33 13................................................................................................................................... 42 Ceiling loudspeakers ....................................33 13...............................................................0 COMPRESSOR/LIMITER .......................................................................................................... 25 Band-pass filters.............................................................................................................................0 GROUNDING AND SCREENING ...........................................5 13......................................................................................2 14....................................................................................................................................................................................................................0 AUTOMATIC VOLUME CONTROL..........6 9....21 8.................................................................... 43 Sound columns................... 48 Efficiency..................................................................................................................... 29 Loudspeaker equalisation...... 33 Power bandwidth........................2............................................ 34 13........21 8..................................................................................... 25 Parametric equaliser....................................................1............................................................................................................................. 36 14............................................................................................................................. 29 9.................................4 15...........................1 9.................................. 26 Parametric triple Q-filter......................................1................1 15.................................................................................................................................................................................................3 Nineteen inch rack units ......................................................................................2 15...1..2 Detailed Considerations ........................................................................2 Radio and Mains Born Interference ....41 LOUDSPEAKERS .................. 27 The acoustic feedback loop...............................................1 Equaliser Types .3 13......2 13.......................2.............................................. 36 14..................4 13....................................................1...................................................................................................................................1.......................................25 9...................................35 HARDWARE INSTALLATION ..........................................................................................................................................................................................................................................................1.........................................................................2 The Power Amplifier...2....................................................................................................................................................................1 Loudspeaker Types ..........................................................2 Earth (ground) loops ......................................................1..................................................................................................................................0 TIME DELAY ........................................................................ 49 ..............................1 Specifications................ 39 Interference introduced via cables........................................................................................................7 Basic tone controls ................2 16...................1...................................................................................................................................................................................................1............................2 Adjusting signal levels in a system chain.................................................................................................................................. 34 Temperature Limited Output Power (TLOP) ......1.............................................5 Standard loudspeaker cabinets ......................4 Prevention of Interference..............................................1............ 29 Loudspeaker equalisation & Loop equalisation ........30 11...................................... 38 14................................................................................................................................................................1.........................................................................0 MIXING CONSOLES ......................................................................................................... .........................................................1.............................. 44 Horn loudspeakers...............36 14..................................................................4 Resonant frequency.............................................................................1 16.................... 21 8...........................4 9...............2.........................................................3 16............................................1........................................................2 9...............................................................6 Frequency Response .............. 33 Non linear distortion or clipping (THD) ..47 16.......................................................................47 16.........................................................2..........1....... 33 Linear distortion................................................................................. 33 Rated Output Power .42 15........................ 40 Interference introduced inside rack unit......1 Basic Principles...................................................................................................................0 TECHNICAL CONSIDERATIONS ..........................................42 15.............1 Safety and system earth’s...............................................................................................................................................................................................................................39 14...........................................................................................................................19 7........................42 15...........................................1 The Pre-amplifier .........0 LOUDSPEAKERS ....................................................1 14....................... 48 Sensitivity ............ 27 Resonant acoustic feedback ....................................2....................................25 9........................................................................................2.......................................2....1..........................................................0 AMPLIFIERS AND PREAMPLIFIERS ............................................................................................... 26 Introduction.......................................1................................................................................. 28 Loop equalisation....................................................................3 Microphone Earth Loops....22 8.....................................................................................................................................................................................2......................................48 16................................................................................................................................... 26 Graphic equaliser ...................................................1...............................................................................................31 12............................................. 40 14.........2 Equalisation.................................................................................................................................................................................................................1 9.....................................................................................2..............................................................................................................2................2.............................3 9............4 9.......................................................................................0 EQUALISERS.......................3 15................................................................................................ 27 Principles of equalisation ................................................................................................................................................................................AMPLIFICATION AND PROCESSING .................................................. 40 Interference induced from 100 V loudspeaker wiring............................................2.....................................................................1 Inputs ....22 9.................................................................19 8...........2 Matching Loudspeakers to Amplifiers ........................46 16.............................2.................................................1.3 9..............................3 14................................... 45 15...........................5 9.........0 TECHNICAL PRINCIPLES ...............................................5 9...........................27 10.................................................................................................................................................................................................................................32 13..................................................................................... 48 Directivity (Q)....................................

.............................................. 54 Refraction ..........................................................................0 DESIGNING FOR THE ACOUSTIC ENVIRONMENT .................................................................4 17.......................... 82 22.............................. 55 18..........................3 19....................1................................................................................................................. 67 Small low ceiling auditorium....................................................... 63 Converting RASTI to %ALcons .........................................................................................................................2 Sound Reinforcement System Calculation & EASE................................ 55 18........3........1 Technical Considerations.................................. 54 Reflection.............. 63 19................................................................1................................................................................. 56 Calculation of Direct and Indirect Sound Fields............................. 68 19................................1.................... 66 19..................................................................1................................. 69 19.................................. 59 Speech Transmission Index (STI & RASTI) ...................................................................0 OUTDOORS ..... Large exhibition hall............................................................................................................................... 82 GOLD LINE AUDIO SPECTRUM ANALYZER DSP30........................................................................... 52 Directivity ......................................................3 18.........................................................................3.................. 54 18...................................... 73 20.....................................5 17..............................1.............................................7 Small reverberant traditional church building................................................. 64 19......................................................0 MEASURING EQUIPMENT ..........................................................................................1.................................................................6 17......1 High ceiling area e............................................................................................................................................0 APPENDIX ................................................................................................................................3.........................................................1 20..................................................................................................................................................................................TRANSFER ANALYZER .1 Loudspeaker Placement and Coverage .... 67 The total (church) sound system chain...........................................7 18.................. 67 Wide low ceiling auditorium........................................................................................................................................................................1............................g.................................5............... 54 Ambient Noise ................................................................................................. 80 22...............................................................................................2 22................................................................ 74 EQUATIONS......................1......................................................................................................................................................................................................................................................................3 20............................ 59 Calculation of the early / late ratio............0 INDOORS ...........................1............................. 53 Variations of both distance and power........................ 69 19..................... 55 Reverberation..........6 19..................................................4 22...............5 18................................................................................................0 E A S E SOFTWARE................ 68 Predicting & calculating the performance of the church system...........4 System Calculation ........................... 75 SURFACE MATERIAL LIST WITH ABSORPTION COEFFICIENTS ..........4 18......................................................... 61 Subjective %ALcons and RASTI requirements........5 19.....2 Summary of the Loudspeaker-design ......... 56 Reverberation time.................................................................................................................................... 77 21...............................................................................................2 17..............................2 19.2 18......................... 52 17...................3 22................................................. 51 17.................3 17..3..... 51 17..................................................................................................................................................................... 66 Large reverberant monumental cathedral........................................................................................................................................1.................................. 64 19.......................................................1 17...............................9 Reflection & Absorption................................................1 22.............................................................3........................................................................................................................................... 82 B & K SPEECH TRANSMISSION METER 3361 .....................................................................................3...................................... 70 20........1..................................................1...........................................1.................................................. 58 Calculation of Reverberant Sound Fields ..............................................................................3 Speech intelligibility in churches & community halls ................................................................................................................................................................................................................................................6 18.............1..............4 19..........................4................................................................................................................................................................................................1...................................................................2 20................5 MLSSA ...................................................................................................3.....7 Power ...TRANSFER ANALYZER ..........1 18........................................... 83 ................................. 82 MLSSA ...............................................THE ACOUSTIC ENVIRONMENT.................................................................................... 65 19.............4 DEFINITIONS ........................1................................ 67 Large high ceiling auditorium....................................................1 Technical Considerations.................................................................................................................1 19......................................... 82 NEUTRIK AUDIOGRAPH 3300.......... 53 Attenuation due to Distance............ 73 SYMBOLS AND UNITS ...........................................................8 18.................................................................

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Once the particles start moving. the hammer hitting a piano wire or a vibrating cone of a loudspeaker. 3. Plane waves originated from a plane source. transmitted. processed. Sound waves are generated by air particles being set in motion by physical movement (such as the bow being drawn across a violin. and perceived by the listener. Spherical waves originated from an in all directions radiating point source.Sound . Attenuation due to distance 2 acc.). they begin a chain reaction with other particles next to them. effective and useful audio system.0 Basics In order to design an efficient. So sound is energy that is transmitted by pressure waves in air. 2. it is helpful to have a grasp of the way sound is received. a movement of air is transmitted in all directions. etc. square of distance = r 1 acc. by expanding and compressing air. In this way.The Theory 1. distance =r no attenuation = r0 Spherical Cylindrical Plane ( Sphere) Single Speaker (Cylinder) Array of speakers 1 (Plane) Multiple speakers . We can make a distinction in the waveforms: 1. Cylindrical waves originated from a line source.

The total sound power (W) is in all cases the same but the sound pressure (N/m2) or sound intensity (W/m2) is decreasing with r2. frequency response. 2. we apply generally the spherical wave attenuation r2 but regard the waveform as a plane wave. the quietest sound level to the loudest one.) we can analyse the signal. time 0 ms = Oscilloscope with memory screen Peak = = Integr. on amplifiers) Slow = = Integr. Sound. level meter. In music. but dependent on the applied instrument’s characteristics (integration time) different levels will be shown. when converted into electrical signals.On relative long distances from the sound source (r >> source dimensions >> wavelength). as we refer to in this manual. Any kind of transmission or registration of sound. time 30 s = Used for Amplifier cooling design 2 . In acoustics. time 4 s = Sound level meter (SLM) LTA = Long Time Average = Integr. In sound engineering we prefer to express it as the difference between the maximum incidental peak value and the minimum value of the converted electrical signal. on mixing desks) Fast = Short Time Average = Integr. alarm signals or attention tones. The justification can be seen in above picture where the sound is moving in jumps of e. 3. music. etc. Sparks = = Integr. time < 5 ms = Peak level meter (e. time 270 ms = VU-meter (e. generally consists of speech. Dependent on our terms of reference dynamic range has different meanings: 1. VU-meter. With different measuring instruments (oscilloscope. imposes a limitation on the dynamic range.g. The limitation of the dynamic range is the most prominent and the most important. time 125 ms = Sound level meter (SLM) VU = = Integr. intelligibility and natural quality.g. the difference between pianissimo and fortissimo.g. 1m.

s. 3 . because of this. Contribution to Intelligibility 5% 13% 20% 31% 26% 5% 7% 22% 46% 20% 3% 2% Contribution to Speech Power 1. integration time 30 sec LTA. Speech has loudness variation and frequency variation. versus time. Dependent on the voice strength the frequency spectrum (which is the lowest bass sound through to the highest treble one) is changing according the diagram. decay time 2. provide the articulation.1 Dynamic Range The accompanying graphs show the speech pattern. and they create the impression of loudness. The human mouth produces these sounds with a narrow opening angle and. having a frequency spectrum above 1000 Hz. The lines in the graph represent the average level per 1/3 octave.7 sec. 100 dB 90 80 Speech (male) 1 3 70 2 60 10 20 30 40 50 60 70 80 90 100 secs.m. Because of this certain techniques are employed to compensate for these phenomena in order to make the speech intelligible. play their part in altering the speech spectrum. The vowels in a sentence have a frequency spectrum below 1000Hz. of a trained announcer speaking at a fixed distance from the microphone. Curve 3: r.m. possible.1.1. 1k 2k 4k 8k 16kHz Our principle aim is to deliver this complete speech spectrum to the listener’s ears as unchanged as The loudness of speech as it relates to frequency. Curve 1: peak value. value. at times making it impossible for listeners to understand what is being said. which are discussed throughout this book. rise time 1 msec. Curve 2: r.s. so easily causing reverberation. value. Words contain both vowels and consonants. measured using different instruments. integration time 270 msec. is rather directional. Unfortunately various acoustical phenomena. Male Speech Spectrum 80 dB SHOUT Sound Pressure Level 70 LOUD 60 RAISED 50 40 30 20 63 125 250 500 CASUAL NORMAL Let's look at loudness first.1 SPEECH Speech consists of words and pauses. The human mouth producing these sounds does so with a wide opening angle and in indoor situations can hit hard surfaces within range like walls and ceiling etc. The consonants of the words in a sentence.

value.2 MUSIC As with any sound transmission. 100 Symphony no. integration time 30 s LTA. value. Curve 3: r. 3. the music will lack the depth to reproduce bass instruments fully. If the frequency response is limited at the lower frequencies.m. VU meter level. measured simultaneously.s. Peak meter level. lacking both subtlety and excitement. 1.2. LTA level.1 Dynamic Range The accompanying graphs show comparative dynamic ranges of different styles of music and speech. the music will appear emotionally flat. integration time 270 ms. Therefore 0 VU or100% on a VU meter should correspond with a headroom of 8 dB under the distortion limit of the equipment. Military Band 1 dB 90 80 2 3 70 60 10 20 30 40 50 60 70 80 90 100 secs. If it is limited at the higher frequencies. showing: 1. 5 (Beethoven) 1 dB 90 2 80 3 70 60 10 100 20 30 40 50 60 70 80 90 100 secs. which are vital for instrument recognition. 2. so in practice 6 dB peak clipping is permissible. Distortion during very short peaks is almost inaudible. 4 . decay time 2. Curve 1: peak value.1. rise time 1 ms. Speech (male) 1 dB 90 80 3 70 2 60 10 100 20 30 40 50 60 70 80 90 100 secs.7 s. It can be seen that there is an average of 14 dB difference between the peak and VU level. harmonics.s. If the dynamic range is limited. the two most prominent features of music reproduction are the dynamic range and the frequency response. will not be fully present. causing the music to sound dull.m. Curve 2: r.

36 0.2 Musical Range versus Frequency Octave bands used for sound measurement Third-octave centres 31 40 50 63 63 80 100 125 160 200 250 315 400 500 630 800 1k 125 250 500 1k 1K 1K6 2K 2K5 3K2 4k 25 2K 4k 5K 6K3 8k 8k 10k 0.1.68 0.2.0 Note Piano keyboard (equal temperament) FREQUENCY VOCAL Soprano Contralto Baritone Bass WOODWIND Piccolo Flute Oboe Clarinet (B flat or A) Clarinet (E flat) Bass Clarinet Basset Horn Cor Anglais Bassoon Double Bassoon BRASS Soprano Saxophone Alto Saxophone Tenor Saxophone Baritone Saxophone Bass Saxophone Trumpet (C) Trumpet (F) Alto Trombone Tenor Trombone Bass Trombone Tuba Valve Horn STRINGS Violin Viola Cello Double Bass Guitar KEYBOARDS Pianoforte Organ PERCUSSION Celeste Timpani Glockenspeil Xylophone FREQUENCY 25 28 31 33 37 41 44 49 55 62 65 73 82 87 98 110 123 131 147 165 175 196 220 247 262 294 330 349 392 440 494 523 587 659 698 784 880 987 1047 1175 1318 1397 1568 1760 1974 2093 2350 3637 2794 3136 3520 3949 4186 A B C D E F GA B C D E F GA B C D E F GA B C D E F GA B C D E F GA B C D E F GA B C D E F GA B C 31 40 50 63 5.34 25 31 40 50 63 80 100 125 160 200 250 315 400 500 630 800 1k 1K 1K6 2K 2K5 3K2 4k 25 5 0.8 1.4 80 100 125 160 200 250 315 400 500 630 800 2.7 Wavelength (m) 1k 1K 1K6 2K 2K5 3K2 4k 25 5K 6K3 8k 0.043 10k 5K 6K3 8k 10k 10.17 .085 Octave centres Frequency (Hz) to nearest 1.

Since the 0 dB (SPL) absolute reference is 20 µPa.1 Ear-characteristics As is shown in the accompanying graph. the listener will have an impression which corres10 ponds with the mirror image of the Threshold of Hearing 20 dB equal loudness contour. corresponds to 20 Pa. using the 1 kHz reference frequency. which. 0 If the noise level is now raised a different impression of the same 20 63 125 250 500 1k 2k 4k 8k 16k 20kHz noise is received. when compared with a 2 or 3 kHz signal 50 an important consideration in any system used to reproduce music. In studying the graph. 20 dB 20 (SPL) is reproduced.3. This is due to the Frequency ear responding to the noise Equal loudness contours according to the changing curves of equal loudness.3 SOUND Sound is a series of vibrations compressing and rarefying the air. having a broad frequency spectrum. 20 Pa ≅ 20 20 log _______ 2 x 10-5 ≅ 120 dB (SPL) 1. very much more energy is needed to produce a bass 60 signal of a given loudness. As the sound pressure level is increased. In other words. called the 'threshold of pain'. being the quietest sound pressure that an average person can hear. 70 Firstly. 100 The threshold of hearing represents the bottom limit of a series of ‘equal 90 loudness’ contours. This is called the 'threshold of hearing'. just short of being painful to the ear. a point is finally reached. the Sound Pressure Level at 130 dB the threshold of hearing varies with (SPL) Threshold of Pain 120 frequency. (Pa = Pascal = N/m2). with a level of. 6 . which at 1 kHz is: 20 µN/m2 = 20 µPa = 2 x 10-5 Pa. which are also shown. when we measure sound. we 80 notice two important factors. Since. all frequencies are present in the signal but depending on the level. say. a reference related to pressure must be used. they will be heard in different relationships. Sound Pressure related to this reference level is expressed in dB (SPL). Because of this it would require 60 dB (SPL) at 30 Hz to 110 produce the same impression of loudness as 0 dB (SPL) at 1 kHz. Loudness is the subjective experience of sound level. which is barely perceptible to people with normal hearing. The reference used is the level of sound at 1 kHz. we refer to changes in air pressure.1. 40 The second point is that if 30 noise.

Many simple sound level meters though are equipped with an A-curve filter only.3 Sound Pressure Level The accompanying chart shows the sound pressures in dB(SPL). This is designated dBA (SPL). threshold of hearing woods library office traffic pneumatic drill threshold of pain dB 0 10 20 30 40 50 60 70 80 90 100 110 120 130 140 ticking clocks tearing paper conversation shouting rock band jet aircraft (100m) 7 . B-.6 2k 2.1.5 A-curve This weighting should theoretically be used only for measurements below 40 dB (SPL). and nowadays the majority of acoustic measurements are taken solely with A-weighting.2 Weighting In order to imitate this characteristic of the ear. 1.3.5 4k 3. dB 10 0 -10 -20 -30 -40 80 A-WEIGHTING 125 100 160 200 250 315 500 400 630 800 1k 1.weighting. and C.3 10 Hz 12.15 5 8k 6. There are three types of curves internationally standardised and they are called A-.25 1. a sound level meter often incorporates different filter curves which corresponds with this subjective hearing.3. for several common sound sources.

λ = wavelength (m).3 dB per 100 metre. which influence the propagation of sound in air are: f = frequency (Hz). Kuttruff.37 || 2. breaking window. Cyril Harris.14 1000 0. loudspeaker.340m λ = 1000 λ = v/f Air Absorption Listening to sound on distance makes us aware of a frequency dependant attenuation due to air-absorption. For normal conditions. sound relies on a vibrating object (vocal chords.07 0.161 V/(αS + 4mV) m = attenuation constant (m-1 ) Room : 100x100x10 m α = 0.94 || 0.45 2000 1.92 4000 4. Because the humidity effects the amount of water molecules in the air.4 SOUND PROPAGATION IN AIR Sound could most simply be defined as a series of vibrations compressing and thinning the air.51 204 6.25 4.1. wavelength and velocity is given by: Using these equations.76 ------------------------------------------------------------------------------------------------------------------------------------- Example: References: [1] Room Acoustics (1991) .000 m3 S = 24000 m2 0 Relative Humidity = 60% Temperature 20 C Freq.96 8000 16 6400 1. it is seen that at 1 kHz at 20°C.78 312 5.02 dB/meter.12 48 6. etc. The humidity level should certainly not be discounted since its effect can be quite dramatic.46 500 0.60 250 0.83 || 16.62 || 0.18 || 0.56 6. To be transmitted.1 V = 100.15 0.41 || 0. the velocity may be calculated by: __ v = 20 √ T where T is the temperature in Kelvin (0°C = 273 K) This means that at 20°C ___ v = 20 √293 = 342. p = pressure (Pa). 8 . m [1-2] 4mV RT60 || m [3] m[3] RT60 (m2) (s) || RH=60% RH=20% RH=20% (Hz) (10-3 m-1) ------------------------------------------------------------------------------------------------------------------------------------125 0.23 6.28 112 6.89 || 5.37 0.34 1736 3.1 0.39 5. This attenuation for a frequency of 500 Hz equals 0.) which imparts its motion to surrounding molecules or particles. also the attenuation of a sound signal is effected. whilst a relative humidity of 80% attenuates a 4 kHz signal by 0. the wavelength is 340 _____ = 0.6 14. T = temperature (K). v = velocity (m/s). Velocity of sound The velocity of sound is determined mainly by the temperature.09 dB/meter. Reverberation time The effect of reverberation time (RT60 ) in a room with volume (V) and surface (S): with: α = average absorption coefficient RT60 = 0.10 6. for 2000 Hz equals 1 dB per 100 metre and for 8000 Hz equals 7 dB per 100 metre (RH=70%).3 ≅ 340 m/s The relationship between frequency.49 596 5. Important physical parameters.5 1.91 1. [2] Handbook of Chemistry and Physics (1973) [3] Absorption of Sound in Air versus RH and T (1967). This means that a relative humidity (RH) of 20% attenuates a 4 kHz signal by 0.28 3.2 47. the higher the frequency the more attenuation. in air.

so the decibel (dB) is defined as: 10 Log P1/P2.1.1. Our human senses . 2. whilst continuing to listen to them one at a time. 126 mW is being fed to one loudspeaker and 100 mW to the other. Only when one loudspeaker receives 26% more power it will sound noticeable louder.2 Power ratios The Bel is defined as: Log P1/P2 . and that the ear hears the same difference between 1 W and 2 W as between 100 W and 200 W. and because of this both their signals are of equal loudness.1 Logarithmic characteristics of the ear To evaluate the ear’s behaviour in respect to sensitivity for level differences.all function logarithmically. hearing. with each power increase of 26% being one tenth of a Bel and called a decibel (dB). for instance. e. Initially the same power is supplied to each loudspeaker.1 DEFINITION The use of the decibel (dB) notation system is common in sound and communications work. .g. 9 .0 Decibel Notation 2. 100 mW. the noticeable increase in loudness is obtained by raising the level in a given ratio. is given in the accompanying table. A power increase of a factor 10 is one Bel. Sine wave generator Stereo amplifier mW mW As the power to one of the loudspeakers is slightly increased no difference in loudness will be heard. This is a logarithm increase not a linear increase. no noticeable difference will be heard until it receives 26% more power (26% of 126 mW = 32 mW). expressed in dB. which brings the higher loudspeaker output to 126 + 32 = 158 mW. This shows. 2.2. At this point e. that 3 dB amplification doubles the power. we can experiment as follows: The diagram shows two identical amplifiers and loudspeakers with a signal generator switched to one then to the other alternately. and that a 100 times increase in power gives 20 dB amplification.touch. Increasing power in ten stages of 26% brings it to ten times its original level. sight. in the presence of a stimulus the least perceptible change is proportional to the already existing stimulus (Weber-Fechner law). This system allows meaningful scale compression or expansion as required and greatly simplifies computations involving large quantities. It must be appreciated that the dB is only a ratio. That is.g. the intensities will again be equal. In this way. If the power to the first loudspeaker is once again increased. etc. not by adding specific amounts of power. If the power of the other loudspeaker is also increased to 126 mW. 1 2 3 10 4 5 6 7 89 2 power ratio 3 100 2 3 1000 4 5 6 7 89 4 5 6 7 89 1 2 3 4 0 5 6 7 8 9 10 11 12 13 14 15 dB 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 The amplification power ratio. sense of weight.

a point is finally reached. If. (1 mW across 600 ohms = 775 mV. As the sound pressure level is increased. which is used in the measurement of sound pressure levels. being the quietest sound pressure that an average person can hear at 1 kHz.4 dB references Though the decibel is only a ratio. when we measure sound. using the 1 kHz reference frequency. a reference of 1 W is chosen. in this case. The reference is 1 Volt regardless of the impedance. At this point the threshold of hearing is very low: 20 µN/m2 = 20 µPa = 2 x 10-5 Pa. and 1 Pa ≅ 94 dB (SPL) 10 . it can be used to express absolute values if there is a given reference. In fact the dBu is referred to the 775 mV regardless of the impedance and is still commonly in use in studio engineering. 1 2 3 10 4 5 6 7 89 voltage ratio 2 3 100 2 3 1000 4 5 6 7 89 4 5 6 7 89 2 4 6 8 0 10 12 14 16 16 20 22 24 26 28 30 dB 32 34 36 38 40 42 44 46 48 50 52 54 56 58 60 From this it can be seen that an amplifier (or attenuator) having a particular gain expressed in dB has a different multiplicative effect dependent on whether power gain or voltage gain is being considered. which. due to its frequent application in Telecommunications is 1 milliwatt (mW) across 600 Ohms. Corresponding dB values are measured in dBV (e.) In practice however the resistance value is frequently ignored when dBm is quoted and the reference is 775 mV only.2. Since power is dissipated in the same resistor: U12 U12/R Ratio (in dB) = 10 Log P1/P2 = 10 Log _____ = 10 Log ____ = 20 Log U1/U2 U22 U22/R Because 10 Log power ratio = 20 Log voltage ratio. Since. or to put it another way. just short of being painful to the ear. this makes this reference incorrect. 4 times increase in power. dBV This is the favourite and common reference for electrical engineering. expressed as "dBm". doubling the voltage results in a quadrupled power. called the 'threshold of pain'. 20 dBV = 10V). which is barely perceptible to people with normal hearing.e. (Pa = Pascal = N/m2) So Sound Pressure related to this reference level is expressed in dB (SPL).1 Pa ≅ 74 dB (SPL). This shows that. dBm .1. a 6 dB gain results in a 4 x power gain but only a 2 x voltage gain. 2. 20 20 Pa ≅ 20 Log _______ ≅ 120 dB (SPL) 2 x 10-5 Other important levels are: 0. The reference used is the level of sound . Since the 0 dB (SPL) absolute reference is 20 µPa.g.3 Voltage ratios When 10V is connected to a 10 Ω resistor: I = U/R = 1A Power dissipated (P) = U x I = 10 W. a gain of 3 dB gives a 2 x power gain. corresponds to 20 Pa. for example. and 6 dB to 4 W and so on. This is called the 'threshold of hearing'. we refer to changes in air pressure. dB(SPL) There is another reference. i. a reference related to pressure (the Sound Pressure Level) must be used. When the voltage is doubled and still connected to the 10 Ω resistor: I = U / R = 2A P = U x I = 40W. but only a 1.4 x voltage gain. then 3 dB corresponds to 2 W. a doubling of the power (3 dB increase) will not result in a doubling of voltage.dBu One of the common references used in the past. In the same way.1. As we know sound is basically a series of vibrations compressing and rarefying the air.

at the intersection with the curved line. then move left. This means that the total value is 80 + 1. or 81. 11 . this leaves 89. follow the line down to find the numerical difference between the total and the smaller level. The difference between these figures is 5 dB. Subtract this value from the total to determine the unknown level. then follow the line to the left. The figure shown on the vertical scale at the left of the chart is the numerical difference between the total and the unknown (the larger) level. enter the chart from the bottom.1 Addition and subtraction When adding two unrelated sound sources. only their intensities (energy) should be subtracted: Ls = 10 Log [10 L1/10 . Deducted from 90 dB total. enter the chart from the left side. Follow the line corresponding to this value until it meets the curved line.2 1 0.2. Then. Using the numerical difference.10 L2/10] The following graph shows how to add or subtract levels in dB's for non-related signals. If the numerical difference between the total and the larger of the two signal levels is less than 3 dB. Example: Subtract 81 dB from the 90 dB total. Add this value to the larger signal level to determine the total. Enter the chart using the numerical difference between the two signal levels being added (top right of chart). follow the line corresponding to this value until it intersects the curved line.2. Ls = 10 Log [10 L1/10 + 10 L2/10] Two different noise sources both producing 90 dB (SPL) would be experienced as: Ls = 10 Log [109 + 109] = 93 dB (SPL) When subtracting two unrelated sound sources. The 9 dB vertical line intersects the curved line at 0.2.6 de 10 d11 de cib els 13 12 0 3 4 5 6 7 8 9 10 11 12 13 To add levels of non-related signals. The 5 dB line intersects the curved line at 1.6.decibels To subtract levels of non-related signals. If the numerical difference between the total and the smaller of the two levels is between 3 and 14 dB. only their intensities (energy) should be added together.2 dB. 0 3 1 Nu 2 me ric 3 al dif fer 4 en ce 5 Numerical difference between total and large levels .2 CALCULATIONS 2. The figure shown on the vertical scale at the left of the chart is the numerical difference between the total and larger of the two signal levels.4 dB. Numerical difference between total and smaller levels . The difference is 9 dB.decibels be tw 6 ee 2 nt wo 7 lev 8 els be 9 ing ad 1. Example: Combine a 75 dB signal with one of 80 dB.2 dB on the vertical scale.

0 An Introduction When assessing the requirements of any sound system it is important to have a firm grasp of what tasks the system will need to perform. every time the amplifier is switched on. and a complete fail-safe redundancy backup system. for example an oil platform. The individual microphone volume levels would be controlled on the amplifier. Other situations. Obviously. the acoustic environment will determine. speech & music or music alone? • Is the system required for announcements and/or for emergency purposes? • How many calls must be made. along with some technical specifications and. 3. Once carefully set up. at the same time. for example a small church needing only speech amplification. to different destinations? • How many different music sources must be routed? • What are the maximum and minimum ambient noise levels? • What is the requirement in respect to loudness? • What is the requirement in respect to speech intelligibility? • What is the requirement in respect to annoyance due to excessive loudness? • What is the requirement in respect to frequency response? • What is the requirement in respect to sound orientation? 12 . Along with this. to a great degree. what equipment should be specified. which also allows tone-control of the loudspeakers. we can reduce the equipment needed to a few microphones. the complexity of calculating the type and quantity of equipment required depends upon the installation's requirements. It is vital therefore to clearly understand the characteristics of the equipment available to meet these various needs. such a system should work without intervention. at times. require both sophisticated routing and switching systems.1 FUNCTIONAL REQUIREMENTS Before starting to design a sound system it is vital to answer the following questions: • Is the system required for speech alone. even though the sound quality should always be adequate.The Sound System 3. one mixing amplifier and a few loudspeaker columns. advice on the techniques involved in installing and using the equipment. In certain applications. This section contains a description of the basic components of the sound system.

output Sound pressure causes the diaphragm to respond in rhythm with sound vibrations.2. 4. temperature independent. Three types of element are generally encountered in microphones used in a professional audio installation.5 Choices Because the microphone is such a fundamental part of the amplification chain. but it is wiser to economise on other equipment than on microphones. and is called Phantom Powering. the louder the sound .2 MICROPHONE TYPES 4. is carried on the microphone's standard two core screened signal cable. As the diaphragm moves.Microphone s 4. whilst a solid metal plate forms the other. polarising charge is maintained across the diaphragm and the plate. 13 .2 Condenser The basic elements of the Condenser microphone are a thin metal flexible diaphragm. and Electret. the first link is often the microphone. 4. supplied by the mixing console or pre-amplifier unit. this causes the capacitance to vary. This provides the polarising charge and also power for the microphone's FET amplifier. A DC voltage.f. which forms one plate of a capacitor. The way an element is mounted in the microphone body determines the microphone's pick-up response pattern.4 Electret Similar in operation to a condenser microphone.2. induces a voltage in the coil. which converts acoustic vibrations into voltage variations. This.the faster the coil moves. causing the subsequent current flow to vary. diaphragm and a permanently charged back plate electrode (which is achieved by sealing electret material onto a metal back plate). A steady D.0 Microphones 4. As the sound varies. so that the coil moves inside the air gap of a permanent magnetic field.2.the further the coil moves. 4. which causes the capacitance to change accordingly. Condenser.C.1 CONSIDERATIONS WHEN SELECTING A MICROPHONE In any sound amplification chain. The capacitance depends on the distance between the diaphragm and the plate.2. permanent magnet diaphragm moving coil a. the diaphragm of the Electret microphone comprises a high polymer plastic film with a permanent electrostatic charge. great care should be taken when making a choice. This means that the higher the frequency . Electrodynamic.1 Electrodynamic The Dynamic microphone is based on the principle of a coil moving in a magnetic field. Normally a compromise must be made between reproduction quality and price. the distance between the diaphragm and the plate varies. the Back Plate Electret (BPE) range of microphones feature a unique design. It is a combination of an uncharged.2. 4. in turn. which in turn causes the voltage to vary. The pitch and intensity of the original vibrations determine the frequency and amplitude of this voltage.3 Back Plate Electret Though operating in a similar way to condenser microphones. 4.

240° 120° If the opening at the rear is adjusted in size and character by means of an acoustic filter. but at a much reduced level. A response approximately halfway between these two is known as a Cardioid (heart shaped) response. BPE microphones have very good speech reproduction qualities. BPE microphones require a supply voltage. the current consumption is so low that up to four microphones can be powered by a single IEC268-15A (DIN4559-6) standard phantom powered input. Other sounds. Having excellent reproductive qualities. being generally very rugged. The current drain is so small that battery life is usually several thousand hours. Because the back of the element is totally sealed. The pattern known as a Hyper. the pressure due to sound entering the back. responding to a voice from the front in just the same way as to the sound from the audience at the rear. FRONT diaphragm 0° 330° 30° Where the rear of the microphone is opened and the diaphragm is exposed to sound waves from the back as well as from the front. (in the case of less expensive models) low reproduction quality. mean that particular care should be taken when selecting dynamic microphones. vibrations and hum fields. which is greater than. Until recently Dynamic microphones were the most popular for general use. The lower sensitivity and.3 PICK-UP RESPONSE PATTERNS The microphone shown in the accompanying illustration is sensitive to sound from any direction. the sound pressure variation leads directly to movement of the diaphragm. generated at the sides and back of the microphone are also picked up. the polar plot is not omni-directional as before. A sound source situated to the side however. the somewhat lower price makes them a viable alternative to dynamic microphones. Though reproduction quality is lower than BPE microphones. but results in a figure-of-eight directional pattern. irrespective of which direction the microphone is facing. Like condenser microphones. 4. and normally the least expensive. but because they do not need a polarising charge. 210° 150° 14 .Until recently condenser microphones have been used primarily in recording and broadcast studios. The difference will generate a maximum signal. the polar response can be varied between the extremes of omni-directional and 180° figure-of-eight. requiring no phantom powering. puts the diaphragm under equal pressure from both sides and will tend to cancel itself out.Cardioid response is particularly sensitive to sounds which are generated at the front. and have low sensitivity to case noise. and phantom powered in professional models. in some cases fragile. and generally require a fairly powerful phantom power supply. Because it is responsive to sound from all directions it has what is called an "Omni-directional" response pattern. The small FET amplifier contained within Electret microphones is often battery driven in consumer quality models. condenser microphones tend to be comparatively expensive. are rugged. and out of phase with. and rarely in public address systems. 60° 300° 270° -10 -20 dB dB -20 -10 90° Sound entering from the front will produce a frontal pressure. and on axis with the microphone body. The force on the diaphragm is determined by the difference in pressure on its front and rear surfaces.

care should be taken in positioning to ensure that the operator is consistently speaking directly at front of the microphone.2 Cardioid Unidirectional microphones with a Cardioid (heart shaped) directivity pattern are normally preferred in general public address distribution applications. and where either: a) the microphone is totally isolated from the loudspeakers.3. Hyper-cardioid microphone characteristics present difficulties to the designers of Lavalier (Lapel) microphones. 4 or the front to random sensitivity ratio 10 Log4 = 6 dB. 4. but to a more extreme degree. to a lesser degree. 240° 120° 210° 180° 150° Both hyper-cardioid and.8 dB. They are used in situations where sound coming from several directions must be reproduced. or b) the microphone is in close proximity to the sound source. so that the comparative level of any amplified signal it picks up is very small. For hyper-cardioid microphones the directivity factor is max. irrespective of source distance. 60° 300° 270° -10 -20 dB dB -20 -10 90° Because of the high directivity of hyper-cardioid microphones.3. Careful tuning of the microphone ensures that whilst only a small amount of extraneous noise is picked up from the rear and sides of the microphone. cardioid microphones have a strongly increased sensitivity to low tones when the sound source is generated close to the microphone. 15 .4. due to their sensitivity to local noise generated by contact with the user’s clothing. This allows a certain amount of freedom of movement for the speaker. at times making the message unintelligible.3 Hyper-cardioid 0° 330° 30° The hyper-cardioid microphone operates in the same way as the cardioid microphone. The directivity factor is the power ratio of the 0° transformed frontal sound when compared to an omni330° 30° directional microphone with the same sensitivity for diffused sound. without large drops in volume level. Because of their normally flat frequency response.3. the Omni-directional microphone is sensitive to sound from any direction. For cardioid microphones the directivity factor is max. their voice will become unnaturally bass in character.1 Omnidirectional Because of its construction. 270° -10 -20 dB dB -20 -10 90° 240° 120° 210° 180° 150° 4. 3 or the front to random sensitivity ratio 10 300° 60° Log3 = 4. This means that if an operator speaks very close to the microphone. It responds to a voice from the front in just the same way as to the sound from the audience at the rear. omni-directional microphones are often used for recording and measurement. the pick up pattern is wide enough to pick up sound from a fairly wide area at the front.

and supermarket floors.2 Noise cancelling microphone This is essentially a hyper-cardioid microphone having an optimum speech characteristic. so filters have been built in to ensure that the frequency response is flat when the sound source is close to the microphone. care should be taken to ensure that they each operate on a different transmission frequency.1 The Lavalier and Lapel microphone These microphones have been specially designed to reproduce speech. they are particularly sensitive to high frequencies in order to compensate for the losses due to absorption by the user’s clothing and made insensitive to the low toned noise caused when the microphone rubs against the clothing. and also that the bass content of the random noise is reduced. otherwise conflicts will occur. ranging from broadcast. they are also suitable for use in such applications where a wide area needs to be monitored. 4. and are small. 16 . Being omni-directional microphones. In the field of sound reinforcement and public address there are again several different types of microphone likely to be encountered for specialist applications. 4. or (b) clipped to a neck tie or jacket lapel (Lapel Microphone) without causing discomfort.4. With this in mind. through to the individual requirements of different musical instruments.4. and is designed for extremely noisy environments such as touring buses.4 SPECIAL MICROPHONES A large number of special microphones are available. The hand held microphone has a built-in transmitter. 4.3 Radio (Wireless) microphone Great freedom of movement is provided for the microphone user by the use of a transmitter/receiver system.4. factories. and designed to be worn (a) around the neck (Lavalier Microphone). A FM signal provides a link between either a hand-held or lavalier/lapel microphone and a receiver connected to the sound system input.4. When two or more radio(wireless) microphones are used in the same location. light. allowing full hands-free use. while the lavalier model is connected to a small pocket transmitter. The microphone capsules themselves are specially mounted in order to absorb shocks and therefore reduce noise being transmitted though the microphone due to movement on the speaker's clothes. such as in a conference recording system. This type of microphone must be held very close to the mouth.

8 dB. described in chapter 9. more than any other type. Though the front-to-rear ratio is only 14 dB it is far more suitable for use in a very noisy environment.1 Potential problems and causes Problem hum oscillation crosstalk Cause Mains power cables 100 V line output cables other microphone cables 5. This is expressed in terms of what is called the front-to-random index where: Fr = 20 log Sf/Sd dB and Sd = average diffuse field sensitivity where Sf = free field sensitivity at 0° The cardioid microphone typically has a front-to-random index of about 4. This is called the front-to-rear ratio. In 4. on the 180° line. in V/Pa.2 Solutions The following steps help avoid these problems: 1. 17 .2 SENSITIVITY The sensitivity of a microphone is the output voltage for a given Sound Pressure Level at 1 kHz.3.50 dB rel 1V/Pa) ( . In installations with long microphone cables.56 dB rel 1V/Pa) ( . Use only two-core screened (shielded) cable for individual microphone signal cables and extensions. The reason is that the most ambient noise does not only come from the rear.6 mV/Pa 1 to 2.8 dB and the hypercardioid microphone has a front-to-random index of 5.5.2.1 DIRECTIVITY There is at times confusion between two terms of reference when microphones are being chosen for use in difficult acoustic environments where the risk of feedback must be reduced. but from the reverberant or diffuse field which is picked up at the sides of the microphone.3.3 INSTALLATION CONSIDERATIONS 5. indicates that the response at the rear. and it is this field that the hyper-cardioid microphone. as shown in 4. attenuates. try to ensure that they cross at 90°.3. The response of a typical cardioid microphone at 500 Hz.60 dB to .3. 3.2. 2. Keep microphone cables away from mains power and loudspeaker cables. use a cable transformer or line amplifier.5 mV/Pa ( . Sensitivities vary considerably dependent on the type of design: Studio Condenser BPE Electret Dynamic 10 mV/Pa 3 mV /Pa 1. 5. is some 23 dB less than that at the front.0 Technical Principles 5. rather than running along side each other. If it is necessary for the cables to cross. as this could cause acoustic feedback (howl around). Also:Never position a microphone in the direct field of a loudspeaker.40 dB rel 1V/Pa) ( .3 the response of a hyper-cardioid microphone is illustrated.52 dB rel 1V/Pa) 5.

are of advanced design. The operator should then reduce the amplifier volume slightly. This is to pick-up full spectrum sound including high frequencies and avoiding air blowing frontal on the microphones diaphragm and causing “plops”. or any microphone. known as 'proximity effect' will occur. This characteristic gives them a high front to random response index. are very sensitive. being particularly sensitive to sounds. and on axis with the microphone body. Speak at a consistent volume level. 1. but at a much reduced level. due to the fact that the amplified signal could be picked up by the microphone and amplified again. 2. which are generated at the front. a phenomenon. the disturbing phenomenon known as acoustic feedback. the speaker's mouth. The best distance from which to speak into a microphone is approximately 15 to 40 centimetres. 18 . effecting the overall clarity.0 Microphone Technique Microphones in the Bosch product range. Other sounds. or 'howl around'.6. or use a tone control or equaliser to attenuate the offending frequency somewhat. hyper-cadioid microphones operate particularly well in difficult acoustic environments and in areas with high background noise. but placed a little below. it is important to be aware of certain operating techniques. do not cover the microphone with your hand. the microphone would also pick up other sounds in the room. 4. and reproduce the human voice with great clarity. 3. Due to the fact that they are so directional. generated at the sides and back of the microphone are also picked up. This is a very noticeable increase in the bass content of the signal. will occur. If feedback does occur. If that distance is reduced greatly. 5. This is particularly unfortunate when the microphone is in the same room as the loudspeakers. this makes the situation worse. especially common to (hyper)cardioid microphones. Many of these microphones have a hypercardioid response pattern. If you are very close to the microphone. The microphone should be pointing directly at. If the operator were to speak from a much greater distance than that recommended. making the voice muffled. and at times unintelligible. If the amplification in this loop is allowed to continue. In order to optimise these. moving backwards sometimes helps eliminate feedback.

each having very sophisticated equalisation.0 Mixing Consoles Certain installations involve a number of microphones. More elaborate installations involving a larger number of microphones a Mixing Console (or Mixing Desk) is the heart of this type of audio system. and is a device which takes the place of a simple pre-amplifier. feeding a large number of sub groups. being the control unit where all the microphones. tape recorder and/or monitor loudspeaker(s). (for instance the stage or platform of an auditorium). In order to give the mixing engineer an undistorted judgement of the total sound. Audio Mixer PHILIPS 1 2 3 4 5 6 7 8 AUX1 AUX2 AUX3 LEFT RIGHT Mixing consoles range from simple units which accept 4 microphone inputs. have basic tone controls. the favourite place for a mixing desk is in the middle of the auditorium. The latter type tends to be accompanied by several banks of audio processing equipment and is very much the domain of the professional mixing engineer. For simple speech reinforcement systems a mixing pre-amplifier is fully adequate to fulfil the requirements. come together. located in the same area. which in turn feed a selection of main outputs. mixed. The final. which need to be amplified at the same time. and provide a mono output. to huge consoles having more than 60 input channels. 19 . cassette players. etc.Amplification and Processing 7. sound is then sent to the input of power amplifiers. On the next page a sound reinforcement system for an auditorium is shown. It accepts these various inputs and blends them together into one balanced whole.

Sound reinforcement system Cassette Recorder Multi cable 12x 1mV 4x 1V Passive Speaker 2 1 2 3 4 5 6 7 8 A A A L R Monitor 3 4 1 5 Monitor 6 Amplifier Cassette Player Speech & music in small auditorium with recording & play back facilities Passive Speaker 20 .

distribution system.250 mV.1 for information regarding the speech spectrum).5 mV. from its 0 ('flat') position. often available as a separate unit. and master volume controls are usually built into the pre-amplifier.8. +20 +10 dB 0 -10 -20 63 125 250 500 1k frequency 2k 4k 8k 16kHz Please note that some lower quality pre-amplifiers provide only attenuation. They operate as follows. which operate over specific frequency bands.1 Inputs The pre-amplifier is normally used for matching and amplifying small voltages. From this range of input requirements two inputs are often chosen: a microphone input with a sensitivity of 0. giving no amplification of either bass or treble frequencies. Typical inputs to the pre-amplifier may be: moving coil (dynamic) microphone . and a music input of 100 mV to 1. The power amplifier. If necessary it is possible to link power amplifier inputs together so that a single input signal can feed a large number of amplifiers. cassette. domestic source (tuner.5V. which is suitable for driving the power amplifier.1 THE PRE-AMPLIFIER 8. the gain is increased.2 Tone controls Tone control circuits vary the frequency characteristics of an amplifier. to provide a voltage level.1. 2. to provide a voltage level suitable for driving the power amplifier.1. dynamic pick-up . the public address amplifier must be considered as two separate sections: the pre-amplifier (voltage gain) and the output amplifier (power gain). The pre-amplifier matches and amplifies the outputs of microphones..1 mV condenser microphone . are basically amplification and attenuation circuits. 1.5 V. CD and cassette players. giving an increase in volume of the respective bass or treble frequencies. where long reverberation times at low frequencies cause problems.5 mV.25 mV. 8. 21 . Bass lift could be used when amplifying music in a heavily damped room.0 Amplifiers and Preamplifiers Although quite often presented as a single unit.1. The 'lifting' of the treble frequencies is particularly useful when it is desired to give speech greater clarity. etc. professional tape recorder . If the bass or treble potentiometer is turned to the right. all Bosch' professional preamplifiers provide both amplification and attenuation (see example next page). Bass lift and treble attenuation is rarely required. In contrast to this. 3. Bass attenuation is particularly useful in large rooms.3 mV. input sensitivity adjustments. and the frequencies within its band of influence are amplified. Tone controls.5 mV to 1. input sensitivity adjustments. usually 500 mV or 1 V. Care should be taken though not to overload the loudspeakers when amplifying the bass content of a signal. with which most people are familiar. or mixing console to a level that will feed the loudspeakers properly. 8. The bass and treble tone control circuits. electret or BPE microphone . tuners. is used to amplify the output power of a pre-amplifier. and master volume controls. the respective bass or treble frequencies are attenuated. CD. helping it to 'cut through' noisy environments (see chapter 1. The pre-amplifier also normally incorporates the tone controls.0. DCC etc) . where the bass frequencies would require reinforcement to give the music more depth. If the potentiometer is turned to the left.

transformers. 100/70/50V 22 . This type of amplifier is favourable if long loudspeaker distances are involved. to a level that will feed the loudspeakers properly. If the load is always constant. So long as the total amount of watts drawn by the loudspeakers is not greater than the rated output power of the amplifier. If using the latter.3 AMPLIFIER/LOUDSPEAKER INTERFACE As stated in 8. (This principle is discussed in the following section) Other power amplifiers. All loudspeakers may be simply connected in parallel. extremely long cable lengths are possible. Due to these low line losses. the 100 Volt line matching system is used. or if the quantity of loudspeakers changes. Many power amplifiers used in public address systems. often used in sound reinforcement systems. so that the amplifier is not able to overload the loudspeakers. the input required to feed the amplifier at nominal full power can range from 100 mV to 10 V. the amount of current (measured in amps) involved is reduced significantly. are tapped to step up the output voltage of the amplifiers from a low voltage to 100. 70 or 50 Volts. This means that even when high power amplifiers are used. which are mounted in the power amplifiers. 3. provide a direct low impedance 2. By increasing the output power voltage of an amplifier. line losses are kept low. it is very difficult indeed to match them to the power amplifier. This is a very important factor in a public address installation. or mixing desk. 2.2 THE POWER AMPLIFIER The power amplifier is used to amplify the output voltage of the pre-amplifier. and all amplifiers in the Bosch product range use what is known as the 100 Volt line principle. In the 100 Volt line matching system. it does not matter whether there is 1 loudspeaker or 150 loudspeakers connected to it at any time. acceptable to the loudspeakers. 4 or 8 ohm output. all Bosch amplifiers utilised what is known as the 100 Volt line matching principle. make sure that the impedance of the loudspeakers matches that of the amplifier. or in an application requiring long loudspeaker cable lengths. then reduce this again to the original low voltage. in order to interface loudspeakers with power amplifiers. However if the loudspeakers differ in power and impedance.2. and that the amplifier power is always lower than the loudspeaker power.8. mounted on the loudspeakers. whilst certain amplifiers in the range also incorporate low impedance outputs. 8. Depending on the design philosophy of the manufacturers. and heavy duty cabling is not required. the loudspeakers can be connected in a series/parallel arrangement to exactly match the amplifier's low output impedance. Transformers. distribution system. This system gives great flexibility in the design and use of public address systems for the following reasons: 1. In this type of situation.

3W. to reduce the power drawn by only a quantity of the loudspeakers. This means that the 70V tap enables the amplifier to power twice as many loudspeakers.The 100V line principle can be compared to a normal domestic mains electricity power supply.g. is printed beside the "power" (+) tap. When it is desired to reduce the power drawn by all of the loudspeakers. It is possible though. 70V P 1/2 power P 50V 1/4 power 100V 1/2P 1/2 power 100V 1/4P 1/4 power Note: When using the 100 Volt line matching system. The total rated power required should be calculated. whereas if they are connected to the 70V tap. P1/4. so that the amplifier is able to power 4 times more loudspeakers. The amount of appliances plugged into a supply is irrelevant. or 6W. with each producing 1/4 of its potential power. 70V 1/2P 1/4 power 23 . P. only 1/2 of their rated power is drawn. The transformers fitted to the loudspeakers have similar taps. a constant supply voltage is present. These loudspeaker transformer taps are used in the same way as the amplifier transformer taps. In a mains supply. 1. but in this case the actual power which the loudspeaker will draw (e. taking into account the reduction in power drawn when using the loudspeaker power taps. instead of the voltage. It is important that this total should not exceed the rated power of the amplifier. the 50V tap allows loudspeakers to draw 1/4 of their rated power. so long as the total amount of power (wattage) drawn is not greater than that available. P1/2. their full power is drawn. matching the power drawn (in this case by each loudspeaker) to the amplifier power available. by using the loudspeaker transformer taps. the Rated Power of the amplifier corresponds to the Rated Load Impedance of the loudspeaker network. 100V 70V 50V 0V amplifier 100V P 0V P 1/2P 1/4P loudspeaker full power When loudspeakers are connected to the 100V amplifier tap. with each loudspeaker producing 1/2 of its potential power. while the remainder draw full power. and it is necessary only to plug an appliance into the mains socket for it to become operational. it is of course simpler and more efficient to utilise the amplifier transformer taps. Similarly. by simply adding the Rated Power of the connected loudspeakers together.5W).

5 dB would require 14. W Example: Assuming an amplifier of 100 W tapped at 100 V and using a cable of 2x0.Cable lengths.5 mm2 1000 m 10000 24 . 10 W is required at the loudspeaker terminals. certain losses must occur. The lengths can be doubled when the load is distributed evenly along the cable. The values refer to a 10% voltage drop. Maximum Permissible Cable Lengths 1000 The maximum permissible cable lengths per size of cable are shown in the accompanying graph. The length of the cable should not exceed 250 m. W 10 0 V 70 100 50 35 25 10 10 100 2 x 1. which of course has an adverse effect on the overall system frequency response. Whilst considering the many advantages of the 100 V line matching system.75 mm2 . 10 0 100 V 70 50 35 25 10 10 100 2 x 0. it is important to realise that by inserting transformers into the signal chain. especially when reproducing bass frequencies. The impedance of transformers also varies with frequency. If for example. Any transformer has an insertion loss.13 W output from the amplifier.5 mm2 1000 1000 m 10000 W 10 0 V 70 100 50 35 25 10 10 100 2 x 2. using a transformer with an insertion loss of 1. and the demands placed upon the amplifiers. with the entire load concentrated at one end of the cable.75 mm2 1000 1000 m 10000 Transformers.

operating over wide frequency bands. It can even equalise the complete audio chain. helping it to 'cut through' noisy environments 2. 9. are basically amplification and attenuation circuits which operate over a specific (though fairly broad) frequency band. giving an increase in the volume of the respective bass or treble frequencies. Their purpose is to severely attenuate all signals below or above a fixed (normally very low or very high) frequency. Bass attenuation is particularly useful in large rooms. a variety of equalisers are available see next page: 25 .1 Basic tone controls The bass and treble tone control circuits. +20 +10 dB 0 -10 -20 63 125 250 500 1k frequency 2k 4k 8k 16kHz These treble and bass tone control circuits are very basic units. where long reverberation times at low frequencies cause problems.9. Bass lift and treble attenuation are rarely required. 9. the gain is increased. Bass lift could be used when amplifying music in a heavily damped room.0 Equalisers An Equaliser gives extensive control over the whole audio frequency spectrum by means of presence (gain) and absence (attenuation) filters and can be used for optimising the frequency response of the sound system. and the frequencies within its band of influence are amplified. They operate as follows: 1.2 Band-pass filters +20 +10 dB 0 -10 -20 63 125 250 500 1k frequency 2k 4k 8k 16kHz Bass and treble "Hi-Pass" and "Lo-Pass" (or "cut-off") filters are intended to restrict the frequency band. If the potentiometer is turned to the left. from microphone to ear. at the same time combating the problem of acoustic feedback by reducing the level of frequencies which cause it. The 'lifting' of the treble frequencies is particularly useful when it is desired to give speech greater clarity. Care should be taken though not to overload the loudspeakers. this would guarantee maximum amplification for the whole frequency spectrum. If the bass or treble potentiometer is turned to the right. raising or attenuating all of the bass or treble frequencies. where the bass frequencies would require reinforcement to give the music more depth.1. In situations requiring control over specific frequency bands. 3. the respective bass or treble frequencies are attenuated. with which most people are familiar.1. Used with care. from its 0 ('flat') position.1 EQUALISER TYPES 9.

which is +20 dB +10 0 -10 -20 63 125 250 500 1k 2k 4k 8k 16kHz Frequency often a sliding potentiometer or "fader". (See 9. 1-2-4 kHz.1. The level at these frequencies should be kept as near to 0 dB as possible to avoid distortion due to a general level increase. Each control.3 Parametric equaliser A parametric equaliser is a unit with 3 or 4 filters.4 Parametric triple Q-filter Basically a parametric equaliser but with pre-set fixed (speech) centre frequencies e.9. To avoid excessive phase shifting. The maximum level of both speech and music is in the 250-500 Hz frequency range. +20 +10 dB 0 -10 -20 63 125 250 500 1k frequency 2k 4k 8k 16kHz The unit is ideal for optimising the amplification of that part of the frequency-band that is responsible for speech intelligibility.and a selection of the width (Q) of the frequency band.2.1. effects a narrow frequency band (third octave). allowing the signal to be sculptured at several specific frequency bands as desired. Because only a few filters are used. This filter allows the operator to select the width & slope of the frequency band (Q) and presence or absence (Gain). The total frequency spectrum is covered. for example.5 Graphic equaliser A Fixed Frequency or "Graphic" Equaliser often consists of 30 individual filter sections. without affecting the neighbouring frequencies. if necessary. or maximum effect is at the centre of each band. it adds clarity and compensates for air absorption. one control at full attenuation and its neighbour at full amplification. speaking too close to a cardioid microphone. This makes it possible to alter. the overall response tends to be quite smooth. +20 +10 gain Q + dB 0 -10 -20 63 125 250 500 1k frequency 2k 4k 8k 16kHz The processing consists of gain correction (+ . with the surrounding frequencies being effected to a proportionately lesser degree.). and the possibility to adjust the frequency to be processed. 26 . 9. .1. care should be taken to avoid extremes of variation between adjacent controls with. The "peak". An adjustable bass cut filter provides smooth roll-off of the bass content in the signal caused by e. a very small frequency band.g.g.4) 9.

The use of sound equalisation to reduce acoustic feedback contributes toward the comfort of both performers and audience and will enhance the acoustic quality and increase the overall system gain.microphone .loudspeakers . feedback will prolong the signal components at this critical frequency.acoustic transmission link between loudspeaker(s) and microphone The acoustic transmission link consists of one (or two) direct path between the loudspeaker(s) and the microphone which is maintained by what is called the “direct sound field”.3 Resonant acoustic feedback Acoustic feedback is spontaneous oscillation caused by the transmission of sound radiated by the loudspeaker(system output) back to the microphone (system Spontaneous oscillations input). 27 .amplifiers with volume control and possibly a tone control (e. amplifiers.9.or resonant acoustic feedback can occur at any frequency for which: a) the phase angle of the transmission through the acoustic feedback loop equals zero. To avoid ringing sounds during speech or musical performances. 9. mixer) . a point will be reached where spontaneous oscillations (howling) start to occur. it is assumed that the sound system in question has already been optimised prior to conducting any equalisation measurements. the gain has to be reduced to approximately 6 dB below the level at which spontaneous oscillation begins. when the performance requires a long microphone distance or with a somewhat noisy audience.2.g. and many other paths caused by reflections and multi-reflections which are maintained by what is called the “diffuse sound field”. 9. When the gain of the sound system is gradually increased. and loudspeaker types is vital when creating a system with smooth response. spurious ringing sounds. and b) the sound from the loudspeaker re-enters the microphone louder than the original sound (loop gain ≥ 1) This cycle repeats itself. e.2. producing ringing or howling sounds. this is called Feedback Stability Margin (FSM). the feedback limit prevents an adequate sound level being produced for comfortable listening. with increased amplification until the sound reaches the system’s maximum loudness or until someone turns down the volume! Even though a sound system is adjusted just below its critical gain.g.2.2 The acoustic feedback loop The total system loop contains basically a sequence of the following elements: .1 Introduction The increase in sound level which a sound amplification system can give to a performance in an auditorium is limited by acoustic feedback. This situation can be annoying to both performers and audience because of an insufficient sound level or when the system amplification is increased. In many cases.2 EQUALISATION 9. Though the selection of microphones. This effect can also be corrected by equalisation. Room conditions can also reduce intelligibility as they “colour” the sound by changing the frequency response.

28 .2.g. This can be done manually by means of a graphic or parametric equaliser or automatically by a so called intelligent feedback exterminator which work with a number of narrow band filters adjusted dynamically at the critical frequencies and maintaining a FSM of 6dB. When considering the sound system equipment alone. 50 dB 40 30 20 10 0 63 50 125 250 500 1k 2k 4k 8k 16kHz Frequency By equalising the loop response now with an “mirror imaged” filter response. as well as reducing the resonant peaks. with its associated acoustic link. 50 dB 40 30 20 10 0 63 125 250 500 1k 2k 4k 8k 16kHz Frequency It is impossible to lay down hard and fast rules as to which equalisation method should be used. possible gain for all frequencies and preserve the signal to noise ratio. a flat frequency response can be achieved within very fine limits. but when taking the sound system as a whole. a high frequency roll off is sometimes required ( 3 dB/octave > 1 kHz). is called “Equalisation”.4 Principles of equalisation The ideal in any audio system is to obtain a flat frequency response over the complete audio frequency band. the overall gain can be increased. In order to maintain the optimum signal to noise ratio. The cancellation of these changes in the frequency response.4 kHz. A listening test after equalisation is important because a flat loop response is not always a flat listening result.2.5) we can obtain e. Using a measuring set-up as explained on the next page (9. as the requirements will be vary from one auditorium to another. whether they be peaks or dips. The prime objective is to obtain a flat frequency response of the loop to obtain max. increasing the gain at dips in the frequency response.9. should be considered. changes are introduced to the feedback frequency response by the very nature of the auditorium. the following loop response: 50 dB 40 30 20 10 0 63 125 250 500 1k 2k 4k 8k 16kHz Frequency The dominant frequency where the acoustic feedback is likely to occur is 160 Hz and secondly 3.

reflected by the room surfaces. 29 . 9. we equalise the whole loop. A 1/3 octave graphic equalizer is the easiest to adjust but difficult to hide for unauthorised tempering. which glides from 20 Hz to 20 kHz. loudspeaker(s). 9.6 Loudspeaker equalisation Power Amplifier Test Unit In a system used for playing pre-recorded music.9. The most convenient method is to inject pink noise in the sound system’s line input and measure with a 1/3 octave Real Time Analyzer on the audience position. This is a good method for adjusting a 1/3 octave graphic equalizer in the system.2. and equalise only the power amplifier.7 Loudspeaker equalisation & Loop equalisation For sound systems used for music reproduction and sound reinforcement.loudspeaker(s) and room. we concentrate our measurements on the loudspeaker reproduction.5 Loop equalisation Power Amplifier PreAmplifier Test Unit In a speech reinforcement system facing the problem of acoustic feedback.amplification . and room.2. The corresponding output. where there is a need to optimise both. is received by the system microphone and plotted on the test unit recorder. Another method is to inject pink noise in the sound system and measure with a 1/3 octave Real Time Analyzer. which consists of the system microphone(s) . In this case we use a calibrated measuring microphone at the audience position (averaged). and is fed into the power amplifier.2. the loudspeaker equalisation should be carried out first. and secondly an additional equaliser should be used in the system microphone channel. The test unit produces a 1/3 octave “warbled” tone.

the sound will be synchronised with the furthest loudspeakers and will benefit the intelligibility considerably. The loudspeakers should be selected carefully and angled for a minimum of backward radiation. but is then followed by arrival of sound from the other loudspeakers. a problem of timing becomes apparent. with loudspeakers located at the left and right hand side of the stage and dispersed at intervals along the length of the auditorium. When all loudspeakers produce their sound at the same time. To overcome this disturbing effect. If the timing is set properly. causing echoes and reverberation. the sound from each (group of) loudspeaker(s) must be delayed using time delay equipment. To overcome this disturbing effect. where the speaker is located. should be delayed proportionally so that the sound appears to come from this centre position. If the timing is set properly. This conflict between the visual and audible experience is rather uncomfortable. Another problem occurs at railway stations. instead of from the stage.10.0 Time Delay When a sound reinforcement system in an large auditorium. The other loudspeakers which should be pointing away from this centre position. the sound from each (group of) loudspeaker(s) must be delayed using time delay equipment. the listener hears the speaker's voice coming from the direction of the closest loudspeaker. Railway platform without delayed loudspeaker signals Railway platform with correctly delayed loudspeaker signals 30 . The most effective way of doing this is to use the loudspeakers located in the middle of the platform as the starting point. (based on the speed of sound travelling at 5 meters per 15 milliseconds). where the aural announcement origin is located at the closest loudspeaker. the sound will appear to originate from the front of the auditorium or area. and the sound of the loudspeakers arrives later (5-15 ms) and not more than 10 dB louder than the original speakers voice.

A limiter is ideal for mounting in call stations to guarantee a fixed maximum output level. but the dynamics of levels above are reduced.3V 1V 3V 10V 30V 0.3V 1V 30dB 3V 10V 30V Log scales Input Log scales Input A compressor reduces input signal variations above the threshold level to about one third (in dB’s) without introducing distortion. A compressor is ideal for background music applications to reduce the (often unwanted) large dynamic range of recordings or broadcastings.0 Compressor/Limiter A compressor and a limiter are input signal dependent attenuators. The attack time is 1 ms. The release time should then be set on >1s to avoid music sounding unnatural (pumping).g. independent of the person speaking (male/female/distance/loudness). 1V for all input levels above the threshold level without introducing distortion. (30 dB input variation gives only 10 dB output variation). The dynamics of input levels below the threshold are not affected. short for speech (100 ms).11. A limiter effectively restricts the output level to e. long for music (>1s). while the adjustable release time is dictated by the application. Output 3V COMPRESSOR 1 : 3 Output 3V LIMITER 1 : 30 1V 1V . 31 .3V . it is necessary to align the rest of the chain in such a way that also the maximum undistorted output level of the amplifier is reached. To utilise this maximum peak level with the full capability of the sound system.3V 30dB 0.

A. If a microphone is located inside the loudspeaker zone to which it is addressed. automatic attenuation(AVC). 32 . which uses this measurement to set the attenuation of the signal path. ensuring that the announcement itself is not measured by the unit as ambient noise. The system should be checked during periods of high ambient noise and low level talking into the microphone in order to ensure that no acoustic feedback. If the loudspeaker system is set up so that a maximum of 89 dB SPL can be achieved. 80 dB(SPL) is regarded as a comfortable maximum listening level. the PA system gain is reduced by the AVC-unit.0 Automatic Volume Control Automatic Volume Control (AVC) regulates the loudness of a P. then a control range of 9 dB would be the right choice. or limiting (Callstation) occurs. depends on the maximum loudness of the sound system. the PA-system gain is restored to its nominal maximum. A blocking circuit ‘freezes’ the input sensor while an announcement is being made. During periods of low ambient noise. and during periods of high ambient noise. therefore only the sensing input microphone gain in the corresponding loudspeaker-zone and the reset time (blocking) needs to be adjusted. with attenuation values from 6 to 21 dB. The AVC unit is factory pre-set. announcement relative to the ambient noise level.12. The ambient noise level is continuously measured by a microphone connected to the sensor input of the AVC unit. An AVC unit with 21 dB control range would only be used in PA systems which can produce a maximum level of 101 dB(SPL). the gain should be carefully set to avoid acoustic feedback. The control range of the AVC. This guarantees maximum intelligibility and minimum annoyance. being 21 dB above the comfortable listening level of 80 dB(SPL).

the normal curve of the signal wave is squared off.1 SPECIFICATIONS 13.3 Linear distortion If an amplifier is not capable of amplifying the full frequency spectrum equally. This unwanted modification of the signal is called linear distortion. producing extra harmonics of the fundamental. causing too much energy to be fed into the loudspeakers (beyond their Power Handling Capacity (PHC) limits). If the amplifier is overdriven. 33 . The written specification of this type of frequency response should state the frequencies at the points where the curves have dropped by 3 dB. the frequency response is from 63 Hz to 16 kHz. which in its extreme could give rise to a guitar input producing a 'piano' sound output. Specifications should be read carefully. the amplified waveform will be altered in a similar way as when tone controls are used.1.13.4 Non linear distortion or clipping (THD) clipping This graph shows an amplifier with too much input signal. dynamic range clipping Another problem occurs when the current continues to rise. The result is an audible change.0 Technical Considerations 13.2 Power bandwidth The Power bandwidth is the frequency range in which the amplifier can deliver its rated power (-3dB) with a maximum distortion level (THD) as stated by the manufacturer (0. making the sound uncomfortably raw.1. he is able to quote a frequency response range which extends much wider than more ethical competitors. a clipping of the output voltage is likely to occur. In our example.1. happens when the input signal exceeds the dynamic range of the amplifier. This effect. 13.1. When the voltage is clipped. When this specification relates to power amplifiers the level at which it is measured should be 10 dB below the rated output power. which could cause them to be damaged.1 Frequency Response +20 +10 dB 0 -3 -10 -20 63 125 250 500 1k frequency 2k 4k 8k 16kHz This graph illustrates the typical flat response of an amplifier suitable for music reproduction. This is commonly referred to as Total Harmonic Distortion (THD). If a manufacturer chooses -6 dB points as reference. 13. 13.5% for PA amplifiers). called non-linear distortion.

6 Temperature Limited Output Power (TLOP) The IEC 65 standard states that an amplifier. Emotional speech. 10 dB. Such instantaneous features of speech and music have to be reproduced without distortion.5 W average power without overheating. This may be expressed as 20 Log 3 = 10 dB.1. in a 19 inch rack frame. A 100 W amplifier. having a input sensitivity of 100 mV. This 100 W is the maximum output power which the amplifier can produce whilst still keeping distortion below its specified limit. stacked on top of each other. means that the peak power is roughly 10 times that of the average power. can cause pronounced audio signal peaks. This means that.1. however. as a power ratio. This is defined in publication IEC 268-3. 34 .5 Rated Output Power Rated Distortion Limited Output Power is the power which the amplifier is capable of dissipating in the rated load impedance. Under normal conditions. This is called the 'rated power' of an amplifier. located in ambient temperature of 45° C. allowance must be made for speech attaining voltage peak values of approximately three times its average. will produce 100 W output when the input voltage reaches 100 mV. without exceeding the rated Total Harmonic Distortion (THD). V 1/3V average 13. an amplifier normally operates at only one tenth of its rated (or peak) value.13. on average. should be able to run continuously for 24 hours per day at 12. for instance. Generally. with + 10% mains over-voltage. running under worse case conditions. at a given frequency or frequency band (1 kHz). the average input voltage will only be 33 mV (allowing up to 100 mV for peaks) and the average output power will only be 10 W (allowing up to 100 W for peaks). or certain passages of music. This means that 100 W amplifiers. In our example this will be 10 W. should at least be able to run continuously at 12½% of its Rated Output Power without any components overheating.

The attention and alarm signals are separately adjustable to an average level of -8 dBV (can be checked as 0 VU on the amplifier). Generally the Power Handling Capacity acc. The Temperature Limited Output Power acc. 4 Loudspeakers are (for reasons of electrical power transport and installer requirements) generally connected via a 100 Volt line system. For this rated outputlevel we specify THD . The 9 dB controlrange is chosen to assure a good performance for ambient noise levels upto 75 dB(SPL). 3 Amplifier needs 0 dBV at the input in order to deliver 100 Volt to the rated load impedance. If signal processing is applied (tone controlling. Rated Power of the amplifier corresponds via 100 Volt to the Rated Load Impedance of the loudspeaker network. IEC268-5 will be greater than the Rated Power of the loudspeaker in order to avoid damaging of the loudspeaker during excessive signal overload (acoustic feedback!). IEC 268-3 DIN45500 FTC etc.Power bandwidth .S/N ratio etc.IEC 65 is specified as 9 dB below the rated output power under extreme working conditions and is a measure for the cooling capacity of the amplifier power stages (heat-sinks and/or ventilators). time delay etc. If not.A. equalising. Therefore the total rated power of the connected loudspeakers (taking the powertapping into account) should not exceed the rated power of the amplifier. 2 Routing controller (SM30 or SM40) has 0 dBV input sensitivity for 0 dBV output.2 ADJUSTING SIGNAL LEVELS IN A SYSTEM CHAIN. output signal is due to the limiter restricted to 0 dBV (=1V). therefore in practice. 35 . The AVC-unit should be by-passed for announcement microphones placed in the addressed loudspeakerzone for acoustic feedback stability. the system gain will be reduced by the control range of the AVC during low ambient noise. The max. The AVC-unit with 21 dB control range is only for those PA-systems which can produce a maximum level of 101 dB(SPL) being 21 dB above comfortable listening level of 80 dB(SPL). Acoustic feedback stability should then be checked during high ambient noise levels and low level talking in the microphone in order to avoid any automatic attenuation or limiting. being 9 dB above comfortable listening level of 80 dB(SPL).13. 5 Automatic Volume Control (AVC) regulates the loudness of the P. acc. During low ambient noise level the PA-system gain is gradually reduced with 9 dB by the AVC-unit. The input adjusters should always be in maximum position and only be changed in the seldom situation that you do not want the full power out of the system for this corresponding microphone input. therefore only the gain of the sensing input should be adjusted to the applied microphone(s) in the corresponding loud-speakerzone and the resettime (blocking). The loudspeakersystem should be set-up such that a calculated SPLtotal of 89 dB can be achieved. The AVC-unit is factory pre-set. speech and/or music should give not more than 0 to +3 VU readings as maximum in order to guarantee that short peaks in the signal (exceeding 100V) do not cause unacceptable audible distortion. The VU-meter has an integration time of 240 ms and adjusted so that 40 Volt (sinewave rms) reads 0 VU (=8 dB below 100 Volt). 1 Microphone in a Callstation is often combined with a pre-amplifier and limiter to optimise the signal in the transport cable. announcement in relation to the ambient noise. The potmeter affecting the gain before the limiter should be adjusted to the announcer and/or acoustic feedback. This guarantees maximum intelligibility and minimum annoyance.) take care of their gain settings to avoid unwanted gain or attenuation for speech/music (can be checked with pink noise). Matching of the loudspeaker requirements to the available amplifier power is done via the tapping-down possibilities (1/2P-1/4P)(70-50V). The limiter is activated by the peaks in the signal therefore the average level of speech will be around -8 dBV but the peak level is close to 0 dBV.

The main reason for the occurrence of these problems is the inadvertent introduction of earth (ground) loops in the earth wiring.Hardware Installation 14. In a system.0 Grounding and Screening 14. distortion. The earth path provided by the mains cable is the 'protective'. and taking all of the interference. 'collected' by the screening. Earth loops exist where multiple connections to earth are made from any one part of the system.2 Earth (ground) loops Incorrect earth wiring in a public address distribution system can cause malfunction of the equipment by introducing hum. where several units are powered directly from the mains supply. The best way to ensure this is to make a separate path to earth by driving a long copper pole into the ground. Because of this. If possible. Special measures must be taken to remove the earth loop. it is often very difficult to trace the source of the problem.1. but at the same time. • via the signal cable screen. PLAY a &^% $ connection to rack frame signal cable screen mains power cable In this case three different paths to earth will have been established: • via the mechanical connection of the chassis to the rack unit. and connecting this to the amplifier or system rack(s) with an adequate earth wire. An earth loop would be the result. Once the system is installed and these problems become apparent. It is therefore vital that this is a 'clean' or 'noiseless' earth. • via the mains earth wire. an alternative clean earth should be established. It can even result in an overload condition. or 'safety' earth.1 EARTHING (GROUNDING) 14.1 Safety and system earth’s In order for a sound system to operate satisfactorily and safely. care must be taken to ensure that it is adequately earthed (grounded). 14. down to ground. Unfortunately the mains earth is often contaminated with interference. the earth connection to the units must be maintained. mounted in the 19 inch rack frame. it is vitally important to design the system so that an earth loop is not built in. A typical example of this would be a tape recorder or professional DCC or CD player. or instability.1. which may cause complete electrical breakdown of components. caused by other types of equipment which use this common earth. for safety reasons. With professional audio equipment this must also act as a 'system earth'. being connected to the system's screening (shielding) network. earth loops can be caused by the mains wiring. which takes any potentially hazardous positive voltage down to ground if an electrical fault occurs. 36 .

Even though many domestic. To avoid intermittent contact. These earth’s should not be wired together. On amplifiers which are stacked. or mounted in a 19 inch rack. a 1:1 isolating (or "galvanic separation") transformer should be fitted in the signal cable. and guarantee maximum security. but an individual wire should be run from each amplifier to the earth connection point. 3. When mains powered domestic music sources (CD players. mains power 5.) are used in a system. and some professional. 4b. two earth’s are present. The other is the 'mechanical earth'. for instance. and all the mechanical earth’s connected at that point. 37 .To ensure that each unit within the system has only one path to earth: 1.). there is often the possibility of a mechanical connection when. Where a separate clean earth is available. but not at the output of the signal source (CD player. music sources and auxiliary equipment have no electrical connection to the mains earth. In instances where the only available earth is the mains safety earth. This should be connected at the amplifier or preamplifier input. In any amplification equipment. One is the 'electrical' earth. 4a. these electrical earth’s should be wired to this ground connection point. 4. connected to the unit's chassis. the chassis of the unit makes metal to metal contact with the rack frame. etc. all electrical earth’s should be connected to the mechanical earth at the same point. An alternative to this is to connect the signal screen to earth at one end of the cable only. cassette machine etc. 6. The earth of the power cable carrying the electricity supply from the mains should be securely connected to the rack. each amplifier and music source chassis should be securely earthed (if necessary with short lengths of wire) to the rack. connected to the 0V side of the circuit. all the electrical earth’s should be connected at the point where the power cable carrying the electricity supply from the mains is connected to the rack. 2.

the screen is used as the return connection for the microphone. these should only be used for very short microphone cables in noise-free environments. The screen.3 Microphone Earth Loops Earth loops are a common source of hum in installations with long microphone and connection cables. For 100% security. with intense magnetic fields. it is possible that noise or hum will be induced. this is to avoid ground loops ( hum). no problems with hum pick-up will occur if the microphone is wired in this way. Example how to connect electrical equipment in a chain. 2. but also in the cable cores. which is connected to earth at the preamplifier input. 4. any noise or hum induced in the cable is included in the microphone signal and amplified. the screening is only on the receiving side of the equipment connected to electrical earth. a twin core screened cable. Do not connect the cable screening wire to the metal body of the microphone connecting plug. If it is necessary to use single screened cables. This is a pre-amplifier fitted with a separating transformer at its input. 3.14. Normally twin core screened cable should be used. not only in the cable screening. In most cases. Such a transformer gives a common mode rejection greater than 30 dB to the microphone signal lines and therefore cancels out any hum present on those lines. with its screen connected to the pre-amplifier’s earth. 38 . should be connected to a balanced pre-amplifier input. With such cables. is not used as the return for the microphone. and because the screen is then earthed at the pre-amplifier input. To prevent earth loops it is wise to follow the next considerations: 1.1. In severe environments.

5 W citizens band transmitter (e. rather than one large opening. within which the distribution system. 27 MHz). extraordinary radio-born and mains electricity supply conditions may cause problems which have to be solved individually. The rack can form a more efficient screen when all of the component parts (covers. and these should be in the form of louvers or small holes. Frequencies above 200 MHz (like radar or relay connections >1 GHz) seldom cause problems. 1 kW radio-therapy unit.2. but only feasible when the offending cause is located 'in house'. c. Within a 100m radius of a 0. The only holes in the outside surfaces should be for ventilation. makes an ideal screen for the electronic circuitry. This would be the case when the system is installed: a. and an all metal door. The rack used must have a top and bottom plate. and/or amplifiers of a sound system would be mounted. The equipment and the patient would have to be located inside a Faraday cage. Screening the source of radiation: Generally the most effective method. A factor which normally decreases the interference influence. 14. Screening the effected equipment: The 19 inch rack unit. especially when metal construction materials or reinforced concrete are used. is the screening property of the building. d. A variety of special application versions exist for this kind of situation. as would be the case with medical equipment etc. Within a 20 km radius of a 1 MW medium wave radio transmitter. within which the 'radiation area' would be confined.g. in normal circumstances. • When voltage spikes on the mains electricity supply exceed 800 V. However. construction bars. For instance. top plate earth bonding wires metal door cable inlet ventilator louvres base bottom plate rear 39 .2 RADIO AND MAINS BORN INTERFERENCE Bosch amplifiers and distribution systems contain extensive protection against external interference sources and. etc. within 100 m of a 27 MHz. depending on the directivity of its antenna. Within a 5 km radius of a 100 kW FM or television transmitter. Problems may be expected when: • An electrical field strength exceeds 1 V/m.14. will not be effected by them. b. Near medical equipment.) are electrically connected. 2.1 Prevention of Interference There are two basic methods of preventing radio born interference: 1. This can occur when highly inductive or capacitive loads are switched on and off on the mains network. Problems of this kind can normally be solved by installing a good mains filter.

or ferrite rod equipment interior 14. Where feasible. and in external cable ducts.2. until the offending cable(s) is (are) found. or mains. causing the system to oscillate. is a potential antenna for radio born interference. See accompanying illustrations for examples.2 Interference introduced via cables Any cable. If this is not done. where possible.2. 14. both inside the rack. In extreme cases it may be necessary to remove paint.3 Interference introduced inside rack unit In some cases. such as cover panels.g. mains/loudspeakers antenna/audio keep short ferrite ring toroid core type 4C6 The simple modification illustrated can effectively cancel this problem. Care should be taken when planning the internal wiring of the rack unit. to keep input wiring. whether signal.4 Interference induced from 100 V loudspeaker wiring Signal wiring. 40 . inductive & capacitive coupling might occur. and use self-tapping screws every 5-10 cm to make the cabinet 100% RF immune. disconnect and reconnect each cable in turn.2. should be kept separate from 100 V loudspeaker wiring. 14. 6 wires on side/rear covers). away from mains wiring and transformers. hum can be induced into a signal line from the radiation effects of mains electricity voltage cables and transformers. On large surfaces.This is done by using short lengths of wire to join each part to its neighbour. these connections should be made in several locations ( e. loudspeaker. Experiment with the amount of windings to find the optimum RF damping.

) can be modified. like distribution controllers amplifiers. requiring 133.75 inches) has been chosen. To simplify calculation of 19 inch rack space required. are 3 HE in height. etc. a fan unit should be mounted in the bottom of the rack to ensure adequate ventilation. are designed to be mounted in a cabinet with a standard front panel width. radio tuners. If power amplifiers are mounted beneath rack frames containing microprocessor controlled distribution units. When several high power amplifiers are used. which could otherwise cause instability in the microprocessor units.65 mm of rack space. This is necessary to deflect hot air currents.55 mm (1. a standard height 'HE' equal to 44. to fit also into these 19 inch racks. The use of this height standard eases the problem of calculating the number of amplifiers. Most power amplifiers for instance. CD players. Cassette front loaders.3 NINETEEN INCH RACK UNITS Bosch public address equipment. or panels that will fit into a given rack. monitor panels and many auxiliary equipment. tuner scales. a heat shield should be installed above them. Certain rules should be observed when planning the equipment layout of the rack. called a "Nineteen Inch Rack".14. Some music source equipment (background music players. and other frequently used equipment should be mounted at a height which makes the front panel clearly visible to the operator. 2. modular distribution units. 3. using bolt on accessories. 15HE 15 10 5 41 . 39HE 39 35 32HE 30 25HE 25 20 1.

The following units are based around cone loudspeakers: 15. Their shape makes them convenient for mounting on walls or pillars. or suspending vertically from the ceiling. a large sealed enclosure will provide better bass response than a small one. Because if this. Bosch offer a wide range of loudspeakers in their product range. which in order to function properly must be mounted in correctly designed enclosures (cabinets or boxes). The bass response of the sealed enclosure is very much dependent on its inside volume. compared with diaphragm (horn) type loudspeakers. limits their use in areas of high ambient noise.Loudspeakers The loudspeakers used in the audio reproduction chain are a vital factor in determining the overall quality and success of a sound system. or where the loudspeaker must be mounted a great distance from the listeners. and do not produce a high SPL. Loudspeakers with larger cone diameters generally give better low frequency reproduction. to give a wide beam of sound.0 Loudspeakers 15. and their particular strengths and weaknesses. In high quality reinforcement and Hi-Fi installations. Normally. often by building in a bass opening or elongated port having very critical dimensions. 15. The fact that they are less efficient. Dependent on the enclosures in which they are mounted. it is vital to understand the different types of loudspeakers available.1. 42 .1 Standard loudspeaker cabinets Standard (infinite baffle) loudspeaker cabinets. and to provide a very high level of reliability. are in principle a sealed box containing 1 cone loudspeaker. and have a typically wide dispersion pattern. enclosures are "tuned" to the resonant frequency of the (bass) loudspeaker. but all are designed and rigorously tested to reproduce speech clearly. the characteristic of handling a wide frequency range makes them particularly suitable for the reproduction of music and speech.1 LOUDSPEAKER TYPES Cone loudspeakers are the most commonly used units.

5 dB 3.5 5 5.5 Mutual Distance D in m 3 4 5 6 7 8 Covered Area in m2 9 16 25 36 49 64 6 9 81 Opening Angle at 4 kHz = 120 0 Level variation = 7 dB Ceiling Height in m 3 3.2 Ceiling loudspeakers Loudspeakers placed in a square pattern Ceiling Lsp A ceiling loudspeaker is a cone loudspeaker. and can be set for 6-5-4-3-2-1 dB variation.3 8.9dB 0dB -1 -2 -3 -4 -5 -6 -7 -8 -9 -10 -11 -12 α -6dB 0dB -6dB -8dB The level variation for wide opening angles however is due to the extra distance attenuation more than for small opening angles. Opening Angle at 4 kHz = 60 0 Ceiling Height in m 3 Mutual Distance D in m 1. A special ceiling speaker program CSP calculates the actual levels under and between the speakers.3 12 16 21 27 Opening Angle at 4 kHz = 90 0 Level variation = 5 dB Ceiling Height in m 3 3.5 4 4.5 4 4.2 5.5 6 2. and the limited maximum SPL available from the units.5 12 14 16 Covered Area in m2 30 49 81 110 144 196 256 Note: a: Caution should be taken when mounting these units in particularly high ceilings (> 5 meters) and in noisy environments.9 3.5 7 9 10.15. which may be recessed into a ceiling or hollow wall. 43 . The level of sound reaching the listeners may be unacceptably low.3 2. They can be spaced at regular intervals to give a fairly even coverage of sound.5 5 5.5 6 Mutual Distance D in m 5. H -6dB X C A common used calculation-method leads to the mutual distance between the speakers: D = 2 H tan (α/2) ( H = Ceiling height to Ear height and α = opening angle at 4 kHz ) And the total number of the speakers: n = Area / D2 D D = 2 H tan (α/2) n = Area / D2 The accompanying tabel shows the level variations which can be expected for different opening angles. due to the distance involved. b: It is difficult to obtain good results from a ceiling loudspeaker system in rooms with a reverberation time of more than 2 seconds (see chapter 18 for indoor acoustics).5 4 4.6 5. mounted on a front panel. α/2 D -6dB -6dB H -12dB -12dB .5 5 5.1.7 Covered Area in m2 3 Level variation = 4. For every Bosch ceiling speaker type the required number of speakers and mutual distance is calculated.5 4 4.

15.1.3 Sound columns
Sound columns are a group of (usually 4 to 10) loudspeakers mounted close together in a vertical array. Due to an interesting acoustical phenomenon, though the beam of sound emitted horizontally is approximately the same as a normal cone loudspeaker, the beam of sound emitted vertically is narrow (10-150) and therefore very directional, especially at higher frequencies. Column loudspeakers are particularly useful in situations where a great degree of control is required over the vertical spread of sound, and no spill of sound is acoustically allowed. A typical instance would be in reverberant environments (e.g. churches) where it is desirable to beam the sound down onto the listeners, without it reflecting off hard walls and ceilings. Unfortunately the bass frequencies are less directional than the higher frequencies, and spread much wider than the useful loudspeaker opening angle. In reverberant environments this wide spread of low frequencies can excite a reverberant field, causing great problems with intelligibility. In situations where the microphone is in the same room as the loudspeakers, this can also cause acoustic feedback. This can be overcome by the use of equalisation (described in chapter 10), reducing the volume of the bass frequencies in the signal. Though this is acceptable for speech purposes, it would have an adverse affect on the quality of music reproduction, so care should be taken not to completely eliminate the bass content of the signal if music is to be amplified.

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15.1.4 Horn loudspeakers
Horn (or 'diaphragm') loudspeakers, are different to cone loudspeakers in that the sound produced is generated by a small, thin metal diaphragm, and amplified by the shape and size of a folded horn. They produce a very powerful, concentrated, beam of sound enabling them to reach listeners at a great distance. Because the diaphragms are normally mounted in moulded plastic or metal folded horn enclosures, they can be easily rendered weatherproof, which allows them to be used outdoors and in dusty and humid environments. They may be mounted on masts or higher buildings and/or arrayed in a column to produce a directional vertical beam. The diaphragm loudspeakers used in public address installations have the limitation of having a fairly restricted frequency range, giving a diminished output at low frequencies due to the diameter of the horn and at high frequencies due to folding of the horn. This makes them generally unsuitable for satisfactory music reproduction, but can to some degree be compensated for by combining them with cone loudspeakers.

15.1.5 Full range high power loudspeakers
Loudspeakers with diaphragms mounted directly onto the mouth of an exponential horn are often used as the treble component of a "full range" multiple loudspeaker enclosure. The audio signal is fed through a suitable crossover filter which eliminates the bass content, which could damage the diaphragm. These enclosures, often grouped together in a cluster, are used in installations to produce full range high power sound.

Combining the horn in the centre of the woofer loudspeaker has the advantage of a compact stackable or arrayable unit.

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15.2 MATCHING LOUDSPEAKERS TO AMPLIFIERS
Two systems are available for connecting loudspeakers to amplifiers: -The direct low impedance system and -The 100 Volt Line Matching System (which is normally used in public address emergency & announcement systems). The loudspeakers could be connected in a series/parallel arrangement, as illustrated, to exactly match the amplifier's low output impedance. This is only a feasable solution if the power leads to the loudspeakers are reasonable short, otherwise line losses are considerably.

8Ω 4Ω 8Ω 8Ω 8Ω 8Ω

8Ω

8Ω

8Ω 4Ω 8Ω

8Ω

8Ω

8Ω 8Ω

4Ω

8Ω

8Ω

8Ω

4Ω

4Ω 4Ω 4Ω 4Ω 4Ω 4Ω 4Ω 4Ω 4Ω

4Ω

4Ω

4Ω

4Ω

4Ω

4Ω

matching loudspeakers to amplifier low impedance output

If the loudspeakers differ in power and impedance, it is very difficult indeed to match them to the power amplifier. In this type of situation, or in an application requiring long loudspeaker cable lengths ( e.g. public address systems), the 100 Volt line system should be used. When loudspeakers are connected to the 100V tap on the amplifier's line matching transformer, their full power is used, whereas if they 100V 0V are connected to the 70V tap, only 1/2 of their rated power is used. 70V This means that the 70V tap enables the amplifier to power twice as P 50V 1/2P many loudspeakers, with each loudspeaker producing 1/2 of its 0V 1/4P potential power. Similarly, the 50V tap allows loudspeakers to use amplifier loudspeaker 1/4 of their rated power, so that the amplifier is able to power 4 times more loudspeakers, with each producing 1/4 of its potential power. 100V P
full power

70V

P

1/2 power

P 50V

1/4 power

100V

1/2P

1/2 power

The transformers fitted to loudspeakers have similar taps, but in this case the actual power which the loudspeaker will draw (e.g. P, P1/2, P1/4, or 6W, 3W, 1,5W), instead of the voltage, is printed beside each power (+) tap. A reduced loudspeaker volume can be set by using these taps. For instance if the same type of loudspeakers are powered from a common amplifier, and it is desired to have one of them producing less volume than the others, then it is a simple matter of connecting the signal to either the 1/2 or the 1/4 power (+) tap. This would reduce the output of the loudspeaker by 3dB or 6dB respectively. Note: When using the 100 Volt line matching system, the Rated Power of the amplifier corresponds to the Rated Load Impedance of the loudspeaker network. The total rated power required should be calculated, by simply adding the Rated Power of the connected loudspeakers together, taking into account the difference in power drawn when using the loudspeaker power taps. It is important that this total should not exceed the rated power of the amplifier.

100V 1/4P

1/4 power

70V

1/2P

1/4 power

Loudspeakers in the Bosch product range are manufactured with a Power Handling Capacity (PHC) according to the IEC268-5 standard. These loudspeakers are actually capable of withstanding power input greater than the PHC, which enables them to avoid damage during times of excessive signal overload (acoustic feedback! )

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• If 2 loudspeakers are placed side by side and given the same input signal (so that both are in phase. at 4 meters 100 dB etc. 112dB 106dB 100dB 94dB 1m 2m 3m 4m 5m 6m 7m 8m The sound pressure level is decreased by 6dB per distance doubling • As we move further away from the sound source the SPL drops. For instance.1 BASIC PRINCIPLES • • • Loudspeaker power handling capacity is measured in watts (W). 47 . Therefore if we know the sensitivity of a loudspeaker. operating as one unit) the SPL at the listeners would be 6 dB more than the SPL of a single speaker. Each time the distance from the loudspeaker is doubled. This causes the total SPL to increase by only 3 dB. That problem is dealt with separately in chapter 18. Each time the input power of a loudspeaker is doubled. 80 dB 80 dB 86 dB Total sound pressure level is raised by 6dB 80 dB 83 dB 80 dB Total sound pressure level is raised by 3dB • If those same loudspeakers are placed some distance away from each other (even so small a distance as 1 meter). The 'sensitivity' of a loudspeaker is the Sound Pressure Level (SPL). each time the quantity of loudspeakers is doubled. A 6W loudspeaker would be able to accept a maximum of 6 watts from a power amplifier. Each time the quantity of loudspeakers is doubled. etc. until it reaches maximum rated power. to 102 dB. on an axis with its centre. 4W would increase it to 105 dB. not taking into consideration any (indirect)sound returned from reflective surfaces.g. when it has an input of 1 watt. at 1 kHz. the SPL is increased by 6dB. instead of 6dB. the SPL rises by 3 dB. at 2 meters distance the SPL would be 106 dB. Again a simple rule is in force. expressed in dB. E.: If a loudspeaker has a sensitivity of 99 dB (1W/1m).0 Technical Principles 16. This rule only deals with direct sound. it is a simple matter to calculate its SPL at any given power input. 2W would raise the SPL by 3 dB.. the SPL drops by 6 dB. measured at a distance of 1 meter. assuming that we have a loudspeaker cabinet producing 112 dB at 1 meter. there will always be a shift in phase at the ears of the majority of the listeners.16.

250 Hz

1 kHz

4 kHz

The opening angle is frequency dependent. We need full spectrum equal coverage, minimising the 4 kHz variation at ear level.

• All of these examples so far have dealt with loudspeakers producing a 1000 Hz tone, being measured in line with the loudspeaker's axis. By looking at the polar diagram, we can see that the SPL differs depending on the frequency being transmitted, and at what angle the listener is relative to the axis (0o). This effect will be used in the formula for the direct sound (LQ.) see chapter 17.

The number of degrees between the points where LQ = 6 dB is the opening angle normally defined for 4 kHz for clarity reasons. In the polar diagrams, this is indicated with a grey shading. The opening angle upto 4 kHz is vital for the intelligibility & clarity reasons.

16.2 DETAILED CONSIDERATIONS
16.2.1 Resonant frequency
At the resonant frequency the impedance is very high in relation to the average impedance. This varies from cone loudspeakers (20 Hz to 300 Hz) to horn drivers (200 Hz to 1 kHz). The 'nominal impedance' is the impedance of the lowest part of the curve above the resonant frequency (fo) - usually around 400 Hz. Damage can occur to the loudspeaker if power is sustained at the resonant frequency. So where continuous alarm signals are required, care should be taken to ensure that the frequency of the signal is well above the resonant frequency of the loudspeakers used.

16.2.2 Sensitivity
The sensitivity level of a loudspeaker is the loudness expressed in dB (SPL) at 1 kHz and at a distance, on axis, of 1 m with an input of 1 W. The importance of this figure may be illustrated by examining the effect of varying the two main parameters, namely, distance and power. Because the efficiency of loudspeakers, horn drivers, columns, etc., vary so much, it is impossible to define the number of loudspeakers required for a room (and the amplifiers required to drive them) without first calculating. Assume that it is required to produce a SPL of 80 dB at a distance of 32 m. To calculate the required power for the loudspeaker (For simplicity an outdoor situation is chosen): Reduction in acoustic level due to distance = 20 Log 32 = 30 dB To compensate for this reduction, 80 + 30 = 110 dB (SPL) is required at 1 m distance from the loudspeaker. If the loudspeaker has a sensitivity of 100 dB (SPL) the missing 10 dB should be compensated for with an extra 10W power applied to the loudspeaker.

16.2.3 Efficiency
A loudspeaker's ability to convert electrical energy into acoustical energy is defined as its efficiency, and can be stated as a percentage figure (values between 0.5-10%). This value is required for calculations of the reverberant sound field. See chapter 18 for details. Because this varies with frequency, Bosch specifies the loudspeaker's efficiency per octave band, on the technical documentation.

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16.2.4 Directivity (Q)
The directivity factor (Q) of a loudspeaker is the ratio of the mean squared sound pressure level at a fixed distance, measured on axis (which is normally the direction of maximum response), to the mean squared sound pressure level at the same distance, averaged over all directions. Q is therefore a measure of the response of the loudspeaker in a three dimensional plane.

125 Hz

1 kHz

8 kHz

At low frequencies the radiation of a loudspeaker has a spherical form which becomes more directional as the frequency increases. This indicates that Q is frequency dependent. Since readings are normally taken in 10° intervals in a sphere, for each of seven octave bands, this requires the processing of more than 2000 readings. Because of this only a few of the leading manufacturers actually quote figures for the directivity factor.

130 120 110 100 90 80 70

dB(SPL)

Q-Factor

100

10

1 125Hz 250Hz 500Hz 1kHz 2kHz 98 109 19 0.42 140 20 4kHz 97 108 25 0.25 110 12 8kHz 93 104 36 0.07 80 8 76 87 3.3 0.02 180 87 98 3 0.21 180 97 108 5.1 1.24 180 70 97 108 9.8 0.64 180 40

1W, 1m(dB) Max,1m(dB) Q-Factor Effic. (%) Hor. Angle Vert.Angle

Standard format for loudspeaker technical specifications, showing average performance for each of the seven octave bands.

Typical average Q values are: Loudspeaker in sealed (infinite baffle) enclosure Average male human speaker Column Loudspeaker Cardioid Column Loudspeaker :2 : 2.5 :7 : 20

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These specifications are measured in an anechoic room following the procedures defined below: 1. The frequency response is measured on axis (0o) at 5 metres and calculated to 1 metre: - with “slow” damping using a gliding tone and/or a 1/3 octave warble (woble) tone. - in 7 octaves, using a stepped pink noise measurement signal.

dB(SPL)
0 -10

80

125
100

160 200

250

315

500
400

630 800

1k

1.25 1.6

2k

2.5

4k
3.15

5

8k
6.3

10

Hz
12.5

The effective frequency range is defined as being the range between those points at which the level drops by 10 dB. 2. For enclosures with single loudspeakers, polar diagrams are measured using pink noise. This is done in octave steps with centres at 125 Hz, 250 Hz, 500 Hz, 1000 Hz, 2000 Hz, 4000 Hz, and 8000 Hz. For enclosures with asymmetrical or multiple loudspeakers the directivity balloon is measured using a pan and tilt device. The definition is every 100 for all 7 octave bands. Using the measurements in 1. and 2., the software package EASE is used to calculate the “Q” and “Efficiency” values for all relevant octave bands. Using the measurements in 2., the horizontal and vertical opening angles (-6 dB) are determined for all relevant octave bands. The Power Handling Capacity is determined by applying the IEC pink noise test signal shown below for 100 hours. After this test the loudspeaker should still be able to perform according its specification.

3.

4.

5.

dB
0 -10 -20
50

Special noise signal acc. IEC 268-3 for Power Handling Capacity test of loudspeakers (100 hours duration)
63
80

125
100

160 200

250

315

500
400

630 800

1k

1.25

2k
1.6

2.5

4k
3.15

5

8k
6.3

10

Hz
12.5

50

it is possible to differentiate between two situations: the outdoor and the indoor environment.g. 17.delivering the message to the ears of the listener clearly. In general. The same audio signal would sound quite different in a sports stadium as compared to a large reverberant church or to a heavily damped lecture room. Speech Intelligibility . 2. Quality of Reproduction .The Acoustic Environment The characteristics of sound and the way it is transmitted are very much altered by the environment in which it is generated.0 Outdoors In the outdoor environment several factors must be considered which influence sound reproduction and reception: • • • • • • • • • • • Sensitivity Power Directivity Distance Reflection Absorption Refraction Air absorption Humidity Temperature Echoes 51 . music to the ears of the listener as unchanged as possible.delivering e. In both situations though we are striving primarily at: 1.

1 TECHNICAL CONSIDERATIONS 17.17. The effect. for an increase of 12 W the calculation is: 100 + 10 Log 12 = 100 + 10.1. from a nominal value of 100 dB (SPL) Intermediate powers may be accounted for by: dB (SPL) at measured power = SPL 1.1 + 10 Log P/P0 where: SPL 1.1 Power Power 1W 2W 4W 8W 16 W 32 W dB (SPL) 100 dB 103 dB 106 dB 109 dB 112 dB 115 dB Each time the input power of a loudspeaker is doubled. which lists the increase in SPL with doubling of power.8 dB for a power increase of 12 W can also be seen in the accompanying table.1 = sensitivity of loudspeaker in dB (SPL) for 1 watt at 1 meter P = power (W) = reference power (1W) P0 Using our reference of 100 dB(SPL).8 = 110. is shown in the table. at a distance of 1m.8 dB (SPL) power ratio 2 3 1 2 3 10 4 5 6 7 89 100 2 3 1000 4 5 6 7 89 4 5 6 7 89 This SPL increase of 10. 1 2 3 4 0 5 6 7 8 9 10 11 12 13 14 15 dB 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 52 . the SPL rises by 3 dB.

1 = sensitivity of loudspeaker in dB(SPL) 1W. This is particularly vital when the microphone is also outdoors. Take care to place the loudspeakers in such a position that there is a "quiet" area around the microphone location.1) of 100 dB(SPL).1 . This is the dispersion (measured as an angle) of sound which radiates from the front of the speaker. Alternatively it may be necessary to concentrate a beam of sound in a particular direction. the microphone. without any objects to cause reflection. the shape of the total beam of sound can be altered to make it more directional. An input of 1 W gives the following results: For intermediate distances: dB(SPL) at measured distance = SPL1.17. by grouping several of them in a vertical configuration. the calculation of the SPL at 25 metres is: 100 .0 for details of using a delay line in this type of situation.1.1. it is necessary to know a little about the different characteristics of certain types of loudspeakers. from a nominal value of 100 dB(SPL) Distance 1m 2m 4m 8m 16 m 32 m dB(SPL) 100 dB 94 dB 88 dB 82 dB 76 dB 70 dB Assume that a loudspeaker source has a sensitivity (SPL1. which disperse (spread) their sound over a wide area. if possible with the loudspeakers in front of. causing acoustic feedback or howl. but could reflect off nearby buildings. or disturb people in neighbouring areas. In installations with low output level loudspeakers. This would be important where an unnecessarily wide spread of sound is not only wasteful in amplifier energy. See 10. commonly called a column. which will be amplified again.20 Log r/r0 where: SPL1. the listener hears only direct radiation.1. 17.20 log 25 = 100 . The sound pressure level drops by 6 dB(SPL) each time the distance is doubled.1m r = measured distance (m) = reference distance (1m) r0 Using the nominal value of 100 dB. 2 4 6 8 0 10 12 14 16 16 20 22 24 26 28 30 dB 32 34 36 38 40 42 44 46 48 50 52 54 56 58 60 53 . it may be necessary to use loudspeakers with a wide opening angle. and pointing away from. and exposed to sound coming from the loudspeakers.28 = 72 dB(SPL) 1 2 3 10 4 5 6 7 89 distance in metres 2 3 100 2 3 1000 4 5 6 7 89 4 5 6 7 89 The SPL decrease of 28 dB at a distance of 25 metres can also be seen in the accompanying table. The table below shows SPL decrease with the doubling of distance. spacing the loudspeakers less than 15 meters apart will help minimise echo.3.3 Attenuation due to Distance When sound is reproduced in an outdoor situation. Dependent upon the environment and the particular application needs. One of the fundamental differences in loudspeaker types is their 'opening angle'. This is discussed in greater detail in 15.2 Directivity Before attempting to calculate coverage. mounted along the length of an area. An uncontrolled spread of sound could return a large amount of the audio signal into the microphone. Even though certain types of loudspeakers produce a fairly wide spread of sound.

occurs when sound passes from one medium to another.6 Reflection Although the effect of reflection is mainly of concern in an indoor situation.105.1.3 dB (SPL) 54 .17. can create a significant ambient noise level.5 Refraction Refraction. Warmer 17.28. with the source shut off (e. L2 is the reading of the noise alone. Pel = power consumption of loudspeakers (W) = on/off axis level difference LQ r = distance from the source 17. the listener will be able to hear. or bending. it is necessary to subtract the ambient noise level reading from the combined (total) reading in order to find the actual level of the source alone. which must be compensated for. gain in dB(SPL) = 20 log 26 = 10 log 10 = 28. Knowing the speed of sound in air to be 340 m/s. This is calculated by: Ls = 10 Log [10 L1/10 L2/10 -10 ] where: L1 is the reading taken of the source and the noise combined (e.4 Variations of both distance and power Assume that a loudspeaker has a sensitivity of 100 dB.1. 55 dB(SPL)). To calculate the dB(SPL) at 26 m with an input of 10W: At 26 m the loss in dB(SPL) And at 10 W. a reflected sound as a whole "echo" of the original.3 + 10 = 81.3 dB = 10 dB The total effect of both variations is simply their algebraic addition: 100 . the reflected sound will have the effect of reinforcing the direct sound.5] = 58. the rumble of heavy industry or even the hum of conversation from a large crowd. reflections from buildings outdoors give distinct and very disturbing echoes. and to recognise. 60 dB(SPL)) and.1. When the sound level of a source is being measured in a situation where ambient noise is present. If this is not done it is not possible to measure the source level accurately.g.1 = SPL value for 1W at 1m on axis.7 Ambient Noise The perceived quality from a sound reinforcement and/or public address distribution system can be particularly effected by ambient noise. 17. If the difference between the direct and the indirect distances is significantly shorter than 17 m. This effect is also noticeable when sound passes through layers of air which have different temperatures and thus different sound velocities.LQ . rather than causing an echo. Cooler The illustration shows the effect of refraction. In this example the level of the source is: Ls = 10 log [106 . The constant sound of passing traffic. causing sound to bend upwards.g.20 Log(r) Ls = SPL1. If the time delay between the original sound and the reflected sound is more than 50 ms.7 dB(SPL) Generally we can calculate as follows: Ldir = Ls + 10 Log(Pel) .1. then the time difference of 50 ms is equivalent to a distance of 17 m.

20 18 dB 16 14 12 10 8 6 4 2 0 -2 -4 -6 -8 -10 Direct Field Reverberant Field SPLrev SP L di r 0. This means that.6 2. no sound is absorbed by the material (α = 0).0 Indoors 18. If all the sound is reflected (r = 1). but the consonants in the speech are hidden or masked by the reverberation. while the lower signals activate reverberation as they bounce off hard walls and ceilings. with increasing distance.3. discussed in 17. the situation is made difficult by a number of problems which must be taken into consideration. a fraction α of the incident energy is lost during reflection. high frequency signals are absorbed by the air.5 4 6. The intensity of the reflected sound wave (Iref) is smaller than the incident one (Iinc).2 0. a higher figure per octave band = greater absorption.16 0. in a reverberant environment.125 0. so that they cannot understand what is being said.15 5 8 DC DL D/DC 18.8 1.1 Reflection & Absorption When a sound source is in a room and enclosed e.25 0.1.25 2 3.18.63 1 1. As can be seen.4 0. 55 . α + r = 1.1.g. This means that the listeners may hear everything loudly. • An reverberant low toned indirect/reflected speech spectrum (SPLrev).3 10 0. The list with absorption coefficients is provided (see appendix) for a selection of materials.1 TECHNICAL CONSIDERATIONS When designing a sound system for indoors. these surfaces will partly reflect and partly absorb the sound. soft materials generally have more effect on higher frequencies. causing low speech intelligibility. or: Iref = (1-α) Iinc α is called absorption coefficient Most of the building materials have measured absorption coefficients (α) and reflection coefficients (r). we encounter two problems at the listeners: • A decreasing original (direct) speech spectrum (SPLdir).5 0. Because the listener is often seated some distance from the source of the sound.1 0.315 0. by walls and a ceiling.

1. The actual level of the reverberant field is determined by three factors: • the nature of the sound source • the physical volume of the room • the reverberation time. The effects of the number of people in the room should normally be taken into consideration.3 Reverberation time The reverberation time (T) of a room is a measure of the time taken for the sound level of the reverberant field to fall by 60 dB. and still more after successive reflections. part will travel directly to the listener.2 Reverberation If sound is generated in a room. • the lack of intelligibility in a room is almost always due to a long reverberation time.4 for details. and total amount of sound absorption in it. 18. called the reverberant field. the effect of the variation in audience numbers is minimised by the use of plush soundabsorbing seating. The reverberation time according to Sabine: T = 0.161 V/( α S + 4mV + nAP ) total volume of the room (m3 ) α = average absorption coefficient n = number of persons total absorption (m2 or Sabine) AP = absorption per person (m2 or Sabine) total surface area (m2 ) surface area (m2 ) atmospheric absorption (attenuation constant) see chapter 1.1. • the reverberation time in a room is the same wherever the listener happens to be.18. which leads to the build-up of diffuse sound throughout the room. having the same absorption as a person actually in the seat. more will arrive after having been reflected. The following points regarding reverberation time are assumed: • the reverberation time in a room is the same whatever the position of the sound source. • reverberation time is determined by the room volume. The effect of these repeated reflections is called reverberation. In many theatres and cinemas however. 56 .161 Volume / Absorption The absorption : Thus: where: V A S Si m = = = = = A = α S + 4mV + nAP α S = ∑ (Si αi) T = 0.

1 x 10 x 0. 57 .161 x 6000 / 412 = 2.161 x 6000 / A A = Absorption is the sum of all surfaces multiplied with the corresponding absorption coefficients.1 x 10 x 0.34 s (neglecting the atmospheric absorption & audience occupation).15 Absorption Sabine = 222 = 30 = 30 = 20 = 20 = 90 = 412m2 T = 0.In other situations like airports where the reverberation time of the empty hall is known. SURFACE Floor Side wall Side wall Front wall End wall Ceiling MATERIAL (carpet) (bricks) (bricks) (woodpanel) (woodpanel) (hardboard) α = 30 = 30 = 30 = 20 = 20 = 30 Total x 20 x 0. the influence of the audience can be calculated by: VT _____________ V + ( 6 T n Ap) New reverberation time = where: V T n Ap = = = = volume of room (m3 ) reverberation time of empty room (s) number of persons absorption per person (e.1 x 10 x 0.37 x 10 x 0. 0.5 Sabine) 5 T 4 r li pe up t mi r ch hu c 3 concert hall 2 it theatre upper lim ma ture cine otion pic m r speech studio fo opera 1 1 000 10 000 100 000 V (m3) Preferable reverberation times. dependent upon the room’s volume Calculation example for determining the Reverberation Time (T) Room with dimensions of 30 x20 x 10m Total Volume = 6000m3 T= 0.g.1 x 20 x 0.

This is the direct sound coming straight from the source(s) plus the indirect sound due to reflections as long as they are within the time window (splittime).1. = Ls + 10 Log Pel .18.20 Log r1 Direct Sound: Ldir = Ls + 10 Log Pel . LQ = off axis level difference α = absorption coefficient r = distance in meters (m) 58 . late sound gives the disturbance. Early reflections First arrived sound Late reflections Reverberation Reverberation Reverberation 0 U s e fu l 10 20 30 40 D is tu r b in g 50 60 70m s The level of this early useful sound (Ldir) can be calculated according the approach as explained in ch.1 = SPL value for 1W at 1m on axis. Early sound carries the intelligibility.17. Early sound is experienced by our ears as the sum of all speech related sounds arriving in a time window of 20-30 ms.LQ1 .4 Calculation of Direct and Indirect Sound Fields Zero order reflections Second order reflections First order reflections It is important to have a good understanding of the different sound fields in a room.20 Log r3 via floor Pel = power rating of loudspeakers (W) Ls = SPL1.LQ3 + 10 Log(1-α2) .20 Log r2 via ceiling Indirect Sound: Lindir Lindir = Ls + 10 Log Pel .LQ2 + 10 Log(1-α1) .

5 +10Log0.18.24 = 89 dB(SPL) Due to sound from 2 loudspeakers 89 + 3 = 92 dB(SPL) On half distance (+6 dB) and .1 + 10 Log Pel .20Log16 = 96 + 17 .5 dB(SPL) Difference between Lrev and Ldir is 4. The reverberant sound level is only caused by the total radiated sound energy and the room properties like V. The useful sound level will vary at different positions in the room.1. The early/late ratio is the input for the speech intelligibility graph on the next pages and gives a quick insight in either STI or Alc. After calculating on a particular position in the room this level difference we enter the chart at the bottom and go up to the intersection with the actual reverberation time (e.2 PHC = 50W Distance = 16m SPLdir = 96 + 10Log50 .5s >> RASTI = 0.10 Log V + 10 Log T + 10 Log (1.1.LQ . We calculate with the 1000Hz data of the room and the loudspeakers.α) + 10 Log ∑ η[%] Pel 18. depending on distance.α).1 = 96dB(SPL) RT60 = 2.5 dB >> T = 2. Volume = 20 x 20 x 10 = 4000m3 Loudspeaker: SPL 1.g.10Log4000 +10Log2. T.36x50 = 96.α) Volume Pac T α = reverberation time (s) = RT60 = average absorption coefficient = loudspeaker efficiency as fraction Volume 1 x = 6T Surface η Pac = Electrical Power x efficiency = Pel x η η [%] = loudspeaker efficiency as percentage Lrev = 120 + 10Log25/100 .36 % α = 0.1 + 10 Log Pel . The following formulas can be used to calculate the reverberant sound field: Lrev α = 120 + 10 Log 25T (1. the volume and the reverberation time of the room.585 ) Calculation procedure for the different sound pressure levels.6 Calculation of the early / late ratio The level difference between useful Direct sound and disturbing Reverberant sound is called early/late ratio and is a good measure for Speech Intelligibility.5 Calculation of Reverberant Sound Fields All the speech related sound which arrives later than 20-30ms is regarded as useless and disturbing and consists of a chaos of reflections and is called reverberation.8 +10Log2x0. T = 3s) and read the Speech Transmission Index (STI) value at the right edge of the chart.20 Log r (off axis) Lindir = SPL1. (1.5s η= 0.1 + 10 Log Pel . angle and useful reflections.10LogV + 10LogT(1-α)η[%] Pel Ldir = SPL1.α) .20 Log r Lrev = 114 . ( Example: 4dB > STI = 0.6 dB off axis we expect the same level = 92 dB(SPL) SPLrev = 114 . Example: In a rectangular room 20 x 20 x 10m we recommend loudspeakers mounted at the left and right hand side of the stage and both aiming to the centre(16m) of the auditorium.60 59 .LQ + 10 Log (1. The level of this reverberant disturbing sound (Lrev) depends of the source(s) .20 Log r (on axis only) Ldir = SPL1.

Lrev = 114 .10 -1 -2 -3 -4 -5 Lr . The corresponding formulas are used in the example below.52 .10 Log V + 10 Log T + 10 Log (1.20 Log r2 .74 .6 .5 3 4 5 6 8 10 10 9 8 7 6 5 4 3 2 1 0 STI .α ) + 10 Log ∑η(%) Pel = Ldir (1) = SPL1.3 .20 Log r1 .5 .LQ4 = 87 Lref (2) = SPL1.14 .1 + 10 Log Pel .60 .20 Log r4 .18 . 2.Ld [dB] This diagram helps to make a direct translation from the level difference between useful Direct sound and disturbing Reverberant sound into Speech Intelligibility.1 + 10 Log Pel .1 Difference: Reverberant minus Useful dB(SPL) dB(SPL) dB(SPL) dB(SPL) 96 4 dB(SPL) dB 100 dB(SPL) After calculating on a particular position in the room this level difference we enter the chart at the bottom and go up to the intersection with the actual reverberation time (e.6 2 2. ( Example: 4dB > STI = 0.70 .40 .65 .1 + 10 Log Pel .78 .g.25 1.35 .23 .1 + 10 Log Pel .2.48 .31 .2 . T = 3s) and read the Speech Transmission Index (STI) value at the right edge of the chart.4 .585 ) 60 .25 .8 1 1.44 .SPEECH INTELLIGIBILITY GRAPH STI & RASTI T[s] .LQ2 = 90 Lref (1) = SPL1.82 .LQ1 = 93 Ldir (2) = SPL1.20 Log r3 . The useful sound level will vary at different positions in the room. depending on distance.LQ5 = 87 Total useful sound (within 25 ms) added acc. which is a sound reinforcement system application with two loudspeakers (1) & (2). angle and useful reflections.57 .27 .

The subjective methods used have been based upon word scores which are arrived at using teams of trained speakers and listeners. which allows objective measurement of the quality of speech transmission with respect to intelligibility. Efforts have therefore been directed to devise methods which yield an objective measure of the speech intelligibility based upon the important acoustic factors background noise and reverberation. and emergency PA-systems.18. The STI method is being standardised by the IEC and published in IEC 268. STI stands for Speech Transmission Index. and analyses the received signal. is based primarily on the measured signal to noise ratio with corrections being applied to allow for the effects of reverberation. using both subjective and objective methods. the STI value is derived. for example. The system can be used for the objective measurement of speech intelligibility in auditoria. theatres. churches. The index is derived from the measured reduction in signal modulation between the speaker and listener positions. called STI. it is of paramount importance. MTF. STI A method is now available. and no corrections need be applied to the results. Part 16. Since the STI value is derived from the measured degradation of the signal. The modulation reduction factor expressed as a function of modulation frequency is called the Modulation Transfer Function.1. STI values can be converted to an expected PB-word score (phonetically balanced cvc nonsense words) on the basis of comparative studies which have been made. it automatically includes the effects of room reverberation and background noise. and. Measuring RASTI Special software has been developed to enable measurements of STI values and to provide additional information of diagnostic value. which are associated with the fluctuation rhythms encountered in speech. theatres. -The measurement is made with the signal and background noise present. The signal and noise need not be measured separately. The STI method is in some respects similar to the method of Articulation Index but offers several advantages. The reduction in modulation can be described by a modulation reduction factor. STI is an index which varies between 0 and 1 and serves as a measure of the speech intelligibility. conference rooms etc. on the basis of the measured change in signal modulation. and from it. the quality of the speech transmission is of interest. Speech intelligibility can be quantified in terms of the changes brought about in the modulation of the speech envelope as a result of noise and reverberation in the room. railway stations. Many different approaches have been taken in the past to try and assess 'intelligibility'. -A STI measurement can be made in less than ten seconds. industry. The STI measurement system consists of a Laptop PC which sends out the special STI test signal. 61 .7 Speech Transmission Index (STI & RASTI) Introduction In any location where verbal communication takes place. The approach used in the STI method is based on the theory that the ability to understand speech is mainly determined by the correct reception of the low-frequency modulations of the speech carrier signal. It will also find application in the testing of Public Address systems in airports. schools. and in many cases such as auditoria. calculates the STI value. It is the ideal instrumentation for rapid and unambiguous assessment of the effectiveness of measures intended to improve the speech intelligibility in different situations.. Perfect transmission of speech implies that the temporal speech envelope at the listener's position replicates the speech envelope at the speaker's mouth. STI values measured in different buildings can therefore be directly compared. The method of Articulation Index (AI). some of these being: -The effects of both background noise and reverberation are automatically taken into account and no corrections need be applied to the results. The STI Method STI is a method of quantifying the intelligibility of transmitted speech and is based upon the method of the Speech Transmission Index STI. This method is time consuming and not always practicable due to the human factors involved. This function provides an objective means of assessing the quality of the speech transmission. and published.

An example of a human voice signal is shown in Fig.Speech STI -Signal The Test Signal The STI method involves measurement of the reduction in modulation of a transmitted test signal. A given STI value could be produced by any combination of these two factors. Measurement A STI measurement is made by transmitting the special test signal and analysing it at the listener's position with a view to calculating the reduction in modulation index for each of the nine modulation frequencies. at 1 m. The STI value is the arithmetic average of these "apparent" signal to noise ratios. S/N. The levels in these octave bands are chosen to equal the average levels found in normal speech. The low-frequency modulations present in human speech are simulated in the STI test signal by 9 discrete modulation frequencies between 1 Hz and 11. Background Noise and Reverberation The reduction in modulation of a speech signal is determined by two factors: the signal to noise ratio. In practice. then the MTF will be flat. Theoretical Models 62 . ie. The 9 modulation reduction indices obtained are interpreted as though they were brought about by background noise alone. MTF. The STI carrier signal consists of two octaves of band limited pink noise centred at 500 Hz and 2 kHz. The signal used in the RASTI method consists of an intensity modulated noise carrier signal and the modulation envelope is shown in Fig. 2. The test signal used has certain characteristics which are representative of a human voice.2 Hz. This is because rapid fluctuations in the sound intensity envelope become more blurred as a result of reverberation. The signal to noise ratios which alone would have resulted in the measured reduction in modulation are calculated. and the low frequency intensity modulations. the relative contribution of S/N ratio and room reverberation will not be known exactly. compared to the slower fluctuations. 3. 59 dB in the 500 Hz octave and 50dB in the 2 kHz octave. The reduction in modulation is therefore greater at higher modulation frequencies. as the background noise affects the modulation to the same extent at all modulation frequencies. but useful information can be obtained from a study of the Modulation Transfer Function. and the reverberation (which is a function of the speaker and listener positions). The characteristics of the human voice which the STI signal is designed to simulate are: the carrier signal. normalized so that the index lies between 0 and 1. If the predominant factor is background noise. at the listener's position. These frequencies span the range found in human speech. If reverberation is the major factor. the MTF will have a negative slope.

45 . %ALcons = 30% RASTI = 0.56 0.1.8 Subjective %ALcons and RASTI requirements.0 1.72 0. -The reverberation time of the room should not be strongly dependent upon frequency.5 2. T.6 1.44 0.50 Speech intelligibility adequate for complicated messages and lectures and for untrained speakers & listeners.82 0.64 0.22 0.1.84 0.88 0.6 5.30 0.86 0. the results obtained can not be interpreted as absolute measurements but can nevertheless be used for comparison purposes for measurements made under the same conditions. 18. -Intense pure tones in the background spectrum outside the 500 Hz and 2 kHz bands are not allowed -The background noise should also be reasonably stationary during the measurement.28 0.36 0.9 Converting RASTI to %ALcons RASTI 0.50 .68 .32 Complicated messages require trained speakers & listeners.9 5.2 2.34 0. can be calculated from a knowledge of the reverberation time. Limitations There are certain limitations on the measurement conditions within which the RASTI method can produce valid absolute results: -Linear transmission is assumed in the STI method.20 0.1845 Ln (%ALcons) + 0. RASTI = 0. Non-linear distortion and clipping are not taken fully into account.46 0.62 0.30% Speech intelligibility adequate only for simple messages and announcements.48 0. If these conditions are not met.0. %ALcons = 1 .3 0.26 0.45 but still adequate for complicated messages in a clear and well articulated voice. upon which STI is based.52 0.419(STI) 16 14 13 11 10 9. actual measurement of STI values in the finished building can then serve as a check on specifications and as a valuable feedback into the theoretical models.78 0.10% RASTI ≥ 0.1 8.90 3.58 0.32 0.76 0. 18. MTF.15% Speech intelligibility adequate for less complicated messages by untrained speakers.70 0. %ALcons = 10 .0. %ALcons = 15 .50 0.3 4.8 3.38 0.54 0. and the signal to noise ratio. RASTI = 0. At the design stage it is therefore possible to specify the desired speech intelligibility in terms of STI values.3 %ALcons = 170.74 0.0.42 RASTI %ALcons 58 52 47 42 37 34 30 27 24 22 20 18 %ALcons 0.8 2.1 2.24 0.80 0.66 RASTI 0.4 3.9482 Source: Farrel Becker 63 .STI values can be quite readily calculated from a knowledge of the acoustic characteristics of the building in question.32 Limit of acceptable intelligibility for simple messages.4 6.8 1.2 7.8 %Alcons 4. The Modulation Transfer Function.5.5405 e STI = .60 0. for trained speakers & listeners.4 1.40 0.

or ambient noise. either by using the formulas provided in chapters 17 or 18 in this manual or using "EASE". For ceiling systems the spacing of the loudspeakers should be determined by looking at the covered areas (-6dB) at 4 kHz. 7.1 LOUDSPEAKER PLACEMENT AND COVERAGE A few practical considerations must be taken into account when selecting. which is the peak level in average conversation measured on a distance of 1m. It means that the audience will hear the announcements at about the same level for the required spectrum. upto the 4 kHz octave band is essential for the annunciation of consonants. the software package described in the Simulating and Measuring Appendix at the end of this book. and therefore intelligibility. while others may have difficulty in actually hearing the audio signal sufficiently. spacing the loudspeakers less than 15 meters apart will help minimise echo otherwise proper delayed signals should be applied. To keep the level more than 15 dB louder than the ambient noise. some listeners could be exposed to an uncomfortably high SPL. Speech requires generally a good transmission and reproduction of the 500 Hz to 5 kHz frequency band. A popular speech peak level. In installations with multiple loudspeakers. while music requires at least 100 Hz to 10 kHz to give satisfactorily results.20 meters from the floor. 6. If this is not so. can make a great deal of difference to the level required for an adequate intelligibility. 3. especially in noisy environments such as factories or airports. a good general rule would be to calculate the level (SPL) at 1. For speech applications. which is the average ear height of a person sitting. Therefore we use the loudspeaker opening angle data at 4 kHz for the calculations for equal coverage. it is possible to calculate the SPL at any point in a room or area. Depending on the application. The audience area divided by this coverage area gives the number of speakers. (See chapter 10 for a description of time delay) Given the specifications of the loudspeakers we intend to use. 8. Background noise. 1. This should be taken into consideration in selecting a loudspeaker type. 5.19. 2. which is not always the case. 4. known as the Comfortable Listening Level (CLL) is generally agreed upon as 80 dB(SPL). 64 . the use of proper callstations with build-in compressor/limiter is required.0 Designing For The Acoustic Environment 19. The loudspeakers must be positioned in such a way that they are able to produce an even spread of sound. This assumes that the ambient noise level is low in the room. placing and aiming a loudspeaker in a sound system design. reaching all audience areas of the room with adequate loudness and clarity.

2000 .α) + 10 Log ∑ η[%] Pel 65 . (on axis only) Ldir = SPL1. 9.10 Log V + 10 Log T + 10 Log (1.19. Check the intelligibility in STI or Alcons(%). 4.250 . 3. The performance of a sound system can be predicted before it is installed or purchased. to verify that the listeners can hear the reproduced sound clearly. Select the correct loudspeaker type(s).6dB points. The level of the direct sound as received from the loudspeakers. including beneficial early reflections from side walls and/or ceiling.2 SUMMARY OF THE LOUDSPEAKER-DESIGN One of the vital requirements of any sound system is its ability to produce an even spread of sound. 6. Calculate the SPLdir on the aiming point(s). Select the Powertapping(s).LQ + 10 Log (1.1000 . the complete speech (and/or music) spectrum should reach the listener's ears as unchanged and true to the original as possible. 7. are calculated for the important octave bands. 8.20 Log r Lindir = SPL1.α) . The level of the reverberant sound caused by the selected solution can be calculated if the Volume / Reverberation time / Absorption is known.20 Log r Ldir = SPL1. With these calculations we optimise the coverage for the audience at 4000 Hz. reaching all parts of an area or room with equal intensity and clarity. Select the optimum loudspeaker position(s).1 + 10 Log Pel . In doing this. 5. Calculate the SPLdir on the . After that the intelligibility is calculated with the values for 1000 Hz. This can be done per octave band (125 . Repeat(?)1-7/9 for other loudspeaker/place/aiming.LQ .500 . 2. In reverberant rooms calculate SPLrev. Select the best aiming points. SUMMARISING THE DESIGN PROCEDURE 1.1 + 10 Log Pel .8000 Hz). Check the coverage at 4000 Hz.1 + 10 Log Pel .20 Log r Lrev = 114 .4000 .

taking care that every seat is within the audible reach of a column. There is a frequency spectrum (which is the lowest bass sound through to the highest treble one). large amounts of money are often spent on a sophisticated sound installation intended to get the word into every corner of the listener area. At the listeners ears we therefore encounter two problems : 1. and because the reverberation is mainly low toned. The loudspeaker system suitable for this venue is a few columns with a multiple loudspeaker-array. Let's start at the beginning . The complete speech spectrum should arrive at the listeners ears as unchanged as possible. Unfortunately. Because of this. and they create the impression of loudness. Unfortunately the final result is often disappointing and at times even worse than having no sound system at all.3 SPEECH INTELLIGIBILITY IN CHURCHES & COMMUNITY HALLS The impact of a good sermon or message is greatly affected by the degree to which speech is intelligible at the ears of the listener. 19.equalisers and loudspeakers are the things wrestled with to achieve the desired effect. Words contain both vowels and consonants. having a frequency spectrum above 1000 Hz. The columns should be placed vertically and mounted close to. Ancient churches are often high and monumental. this causes lots of reverberation.3.with the human voice: Speech consists of words and pauses. This means that the loudness might be okay but the consonants in the speech are hidden or masked by the reverberation. 2. the only thing we need to do is reproduce the mid and high frequency spectrum of speech (above 1000 Hz). 66 . An increasing reverberant low toned indirect/reflected speech spectrum.organ . Let's first look at loudness. There is pitch variation. The vowels in a sentence have a frequency spectrum below 1000Hz (two octaves above middle C for the musicians). because the listener is often seated some distance from the source of the sound.19. Unfortunately this is per definition always in conflict with speech requirements.amplifiers . If we look around in many churches. See the graph showing the loudness of speech and the contribution of each octave band to speech intelligibility. causing low speech intelligibility or in technical terms the so called "Speech Transmission Index" (STI or RASTI) is low. Often the reverberation is so high that a sound system adds to the problem instead of solving it. The human mouth producing these sounds does so with a wide opening angle and because the sound hits everything within range. while the lower notes activate reverberation as they bounce off hard walls and ceilings. For good speech intelligibility we need a sound system that avoids an increase of the reverberation but still amplifies articulation.mixing desk . There is loudness variation. The consonants of the words in a sentence. Microphones . is very directional. the acoustic difficulties are obvious. These radiate in a nice wide horizontal pattern while almost no sound is radiated vertically. and somewhat at each side of the person speaking. This avoids sound hitting reflective ceilings etc. A decreasing original direct speech spectrum.choir and community singing. having awesome acoustics for traditional music .1 Small reverberant traditional church building. The human mouth produces these sounds with a narrow opening angle and. the higher notes are absorbed by the air. provide the articulation. avoiding unwanted reverberation. The sound system is normally needed for enhancing the speech intelligibility. because of this.

3 Small low ceiling auditorium. These two functions should be considered in all applications. with direct sound from the loudspeakers providing good intelligibility.3. full range loud-speakers. will help. If music amplification is also needed. Unfortunately. Ceiling loudspeaker groups with electronic time delay will solve this. placed left & right at the front of the room. Mounting the long cardioid loudspeaker columns on every pillar. 19. If music must be reproduced too. the sound from each (group of) loudspeaker(s) must be delayed (5 metres = 15 milliseconds) using time delay equipment. Cabinets with "constant directivity" (the same opening angle for all the relevant frequencies) are ideal for this purpose. 67 . due to larger areas to be covered. we need to lip synchronise our audio to the video image. and in this case no time delays should be applied. we can group several loudspeaker cabinets in a cluster high above the stage. Unfortunately the loudspeakers produce the sound at the same time. the fact that the sound is coming from above may appear very spiritual. The delayed ceiling system gives the extra mid and high tones needed for clarity and intelligibility.5 Wide low ceiling auditorium. to improve on the decreasing direct speech spectrum. which means that the congregation hears the preacher's voice from the direction of the closest loudspeaker. and the mixer could also feed a separate induction loop amplifier for the hard of hearing.2 Large reverberant monumental cathedral. but then we need proper time delays to avoid sound coming from the wrong direction. To overcome this disturbing effect. We can achieve this with columns and/or with a nice pattern of (delayed) ceiling loud-speakers in order to cover the complete audience area. will give good low tone reproduction and solve the orientation problem. we now need more and/or more powerful full range loudspeakers. full range two-way loudspeakers should be used in front. but due to the room's absorbent character. The advantage here is that an almost ideal coverage can be achieved. but it isn't audibly correct. If a TV-monitor system is used. the loudness is normally low and in need of pure amplifica-tion. Because a high mounted cluster might be impracticable. 19. we must now consider a ceiling mounted system. controlled by an operator.3. As an added bonus. To ensure that every seat receives roughly the same direct sound (<5dB).3.3. We can use the previous solution but due to larger distances we now need a more widely distributed loudspeaker column solution.19. In this case several sound signals (speech & music) are amplified by the system at the same time and a mixing desk. This type of venue does not cause too much trouble with respect to reverberation. a cassette recording can be made of the music and sermon. is needed.4 Large high ceiling auditorium. Cabinets located throughout the audience area can provide an alternative solution. We can use the previous solution but. 19.

with radio receivers connected directly to the mixing desk. For speech and music. this would guarantee maximum amplification for the frequency spectrum required.6 The total (church) sound system chain Microphones are the receivers of sound generated by such sources as preacher. the amount of direct sound (received from the loudspeakers). Used with care. tape recorder and/or monitor loudspeaker(s). The individual microphone volume levels are controlled on the amplifier. These microphones are either directional and mounted on stands. After that the timing of the sound from individual loudspeakers is checked and the speech intelligibility is calculated. For small churches needing only speech amplification. Nowadays this time-consuming calculation work and the corresponding presentations can all be done with PC software like Catt Acoustics or E A S E. The Amplifier is used to amplify the power of the mixing desk to a level that will feed the loudspeakers properly. the favourite place for this desk is in the middle of the auditorium. It can even equalise the complete audio chain. at the same time compensating for the menace of acoustic feedback. loudness(SPLtot) clarity(dB).500 . wireless microphones with transmitters can be used. such a system should work without intervention. we can reduce the equipment to a few microphones.2000 . intelligibility(STI). screened (balanced!) cables (with XLR plugs) etc.1000 . singer. which also allows tone-control of the loudspeakers. guitars. use good full range two-way cabinets. make sure that the amplifier wattage is always lower than the loudspeaker wattage so that the amplifier is not able to overload the loudspeakers. An Equaliser gives extensive control over the whole frequency spectrum and can be used for optimising the frequency response of the loudspeakers. etc. Loudspeakers used solely for speech are mostly column types.3. The Mixing Desk is the heart of the audio system and is the control console where all the microphones. In order to give the sound engineer an undistorted judgement of the total sound.8000 Hz). come together. (Electro Acoustic Simulator for Engineers). The program provides pictures of the audience areas with plotted calculation results regarding: coverage(SPLdir). musical instruments and congregation. 68 . are also needed.7 Predicting & calculating the performance of the church system. or omni-directional and tie clipped or lavalier. The performance of a sound system can be predicted before it is installed or purchased.4000 . electronic organ. etc. cassette player. Accessories like floor stands. If using the latter. which generate the full speech and music spectrum. from microphone to ear. and the amount of reverberant sound (via the reflections) can be calculated for every relevant octave band (125 -250 . This is either a 100V or 70V line level amplifier (favourable if long loudspeaker distances are involved) or a direct low impedance 4 or 8 ohm amplifier. having small vertical opening-angles to reduce reverberation. mixed sound is sent to the amplifiers.19. Per seat. To get more freedom of movement. one mixing amplifier and a few loudspeaker columns. 19. This is where the final. Once carefully set up. every time you switch on the amplifier. table stands. arrival times(ms).3.

52.38 = 83.98 2.4 .84.1. Bosch installations with Plena Voice Alarm or Praesideo Call Stations have those limiters and therefore ideal for Public Address installations.19.4 dB(SPL) >>>>>>> ALcons = 5. The intelligibility is then not further affected by a bad signal to ambient noise ratio as long as it is better than 15 dB.25 m = 188820 m3 α = V/6TS = 188820 / (6 x 2.4.1 dB Signal to Noise ratio = (88.g.32 x 50) = 88.9 dB Roomdata: Volume = 172 x 72 x 15.1 High ceiling area e.25 m T: 2.4 dB(SPL) dB(SPL) = 84.76 + 3.4 SYSTEM CALCULATION 19.3 = 4.4 + 0. Large exhibition hall -3dB -6dB at 4kHz Dimensions : 172 x 72 x 15.Ldir = 88.4 & 84.4 dB(SPL) & 77 dB(SPL) = 83.5 x 32840) = 0.08 + 10 Log (21 x 0.7% (see Chapter 18.62 Lrev = 114 .5s Ambient Noise : 70 dB(SPL) Mounting height loudspeakers : 15 m Total number: 3 rows of 7 = 21 Loudspeaker data for 1000 Hz: PHC = 50 Watt Sensitivity = 89 dB(SPL) Efficiency = 0.70 = 20 dB This is not good enough.3) .α = 0.20 Log(15 -1. therefore call stations with limiters are needed to minimise the dynamic behaviour of the announcements.3 dB(SPL) Surface : 32840 m2 1.10 Log V + 10 Log T + 10 Log(1-α) + 10 Log{Σ η(%) x PHC(W)} = 114 . 69 .32% Dispersion = 110o What is the expected intelligibility in ALcons and/or RASTI ? CALCULATION PROCEDURE Ldir = 89 + 10 Log50 .6 Benefit of the closest four neighbour speakers (estimated) 4 x 71 dB = 77 Ldir(tot) = 83.22.5) Intelligibility : ALcons = 20% Lrev .5) = 89 + 17 .

Volume = 20 x 20 x 10 = 4000m3 RT60 = 2.2 Sound Reinforcement System Calculation & EASE In this rectangular room we recommend loudspeakers mounted to the left and to the right hand side of the stage and both aiming to the centre(16m) of the auditorium.1 = 96dB(SPL) 70 . We calculate with the 1000Hz data of the room and the loudspeakers.36 % PHC = 50W Loudspeaker: SPL 1.2 Distance = 16m η= 0.19.5. In the pictures the beams (3 & 6 dB) of the speakers and the covered areas are shown.5s α = 0.

10Log4000 +10Log2.44 .60 .74 .5 dB >> T = 2.5 3 4 5 6 8 10 10 9 8 7 6 5 4 3 2 1 0 STI .25 .60 SPEECH INTELLIGIBILITY GRAPH STI & RASTI T[s] .Calculation procedure for the different sound pressure levels.57 .4 . 71 .3 .Ld [dB] This quick calculation is only valid for two positions in the room (on the aiming point and close to the stage centre).40 .5 +10Log0.70 .18 .6 dB off axis we expect the same level = 92 dB(SPL) SPLrev = 114 .25 1.5s >> RASTI = 0.8 +10Log2x0.35 .78 .48 .24 = 89 dB(SPL) Due to sound from 2 loudspeakers 89 + 3 = 92 dB(SPL) On half distance (+6 dB) and .65 .36x50 = 96.14 .5 .52 .82 .20Log16 = 96 + 17 .2 .5 dB(SPL) Difference between Lrev and Ldir is 4.8 1 1.23 .27 .31 .6 2 2.10 -1 -2 -3 -4 -5 Lr . To repeat these calculations for more or even every position in the room an EASE simulation can be of great help. SPLdir = 96 + 10Log50 .6 .

STI 72 .Simulations for 1000Hz from the software package EASE SPLdir SPLdir + SPLrev Speech Intelligibility acc.

Free Field : An environment in which there are no reflective surfaces within the frequency range of interest. The pressure at a point in space minus the static pressure at that point.20. 73 . Sound Pressure Level p The fundamental measure of sound pressure defined as: Lp = 20 Log p dB o The reference pressure po is 20 µPa for measurements in air. When A-weighting is used.) the mean square sound pressure at the same distance from a nondirectional source which radiates the same acoustic power. Equals the ratio of the speed of sound in the medium to the fundamental frequency.0 Appendix 20. Characterised by the existence of maxima and minima amplitudes. the sound level is given in dB(A). Directivity Factor The ratio of (1. Pink Noise Broadband noise whose energy content per constant bandwidth (Hz) is inversely proportional to frequency. 1/3 octave. White Noise A broadband noise having constant energy per constant bandwidth (Hz) but increasing 3dB with each doubling of relative bandwidth (octave. but constant per octave or 1/3 octave of bandwidth. Standing Wave A periodic wave having a fixed distribution in space which is the result of interference of progressive waves of the same frequency and kind. Decibel Scale A linear numbering scale used to define a logarithmic amplitude scale. Sound Pressure 2 A dynamic variation in atmospheric pressure (in N/m = Pascal). thereby compressing a wide range of amplitude values to a small set of numbers. Sound Intensity (= Sound Power) The sound energy transmission per unit area and per unit time in a specified direction. decade).) the mean square sound pressure at a specified distance and direction from the sound source to (2. Sound Level The level of sound pressure measured with a sound level meter and one of its weighting networks. Isolation : Resistance to the transmission of sound by materials and structures. Unwanted sound is commonly called noise. thereby creating essentially free-field conditions. Wavelength The distance along a periodic wavefront between points of comparable amplitude with a phase difference of one period. Sound Energy Energy that is transmitted by pressure waves in air or other materials per unit area and is the objective cause of the sensation of hearing.1 Definitions Anechoic Room A room whose boundaries effectively absorb all incident sound over the frequency range of interest.

1845Ln(Alcons) Splittime Remarks e.9482 .ϕ)/p(r) Γ(θ.g. 200C) ρ ≈ 1. 200C) 1 Pa = 1N/m2 ≅ 94 dB(SPL) pref ≡ 20 µPa Lp = 20 Log(p/pref) Ls ≡ 20 Log(ps/pref) See Polar diagram of LS.20.ψ. Γ(θ.ϕ) = p(r.13K = 00C c ≈ 344 m/s (air.21 kg/m3 (air.Octave T ≈ 0.ϕ =0) = Γ(max) =1 Consumption Radiation η = Pac /Pel Q = Imax /Iav.2 Symbols and Units Symbol / Unit B V S A T α m f ψ Tr c ρ w I p pdir pdif pref ps Lp Ldir Ldif Ls LQ Γ(θ.ϕ Pel Pac η Q D r Alcons STI ts Hz m3 m2 m2 s m-1 Hz % K m/s kg/m3 J/m3 W/m2 Pa Pa Pa Pa Pa dB(SPL) dB(SPL) dB(SPL) dB(SPL) dB W W dB m %! Quantity Bandwidth Room Volume Room Surface Absorption Reverberation Time Absorption Coefficient Air absorption constant Frequency or Centre frequency of band Relative Humidity Room Temperature Sound Velocity Specific Density Energy Density Sound Intensity Sound Pressure Direct Sound Pressure Diffuse Sound Pressure Reference Pressure Characteristic Sensitivity Sound Pressure Level Direct Sound Level Diffuse Sound Level Characteristic Sensitivity Level Off axis level difference Directivity Function Angles with reference axis Electrical power Acoustical power Efficiency Directivity Factor Directivity Index Distance to source Loss of Consonants Speech Transmission Index RASTI = 0.θ. Decade .0.ϕ) θ.Tr) 273.161 V/A m = m(f. Iav = Pac /(4πr2) D = 10 Log(Q) Articulation Loss IEC Norm (Farrell Becker) Time window useful sound s 74 .

161 V/A Absorption Sabine: A = 4mV + αS Eyring: A = 4mV .. Γ = 1(on axis). r = 1m.20. by: m ≈ 1.3 Equations Sound velocity c = 20. I = wc = p2/ρc Direct Sound Pressure and Level pdir2 = ρc QP/4πr2 ps = √(ρcη /4π) pdir2 = ρcPacΓ2/(4πr2) = ps2PelΓ2/r2 Ldir = Ls + 10 Log(Pel) .] Air Absorption Constant Can be approx.7 10-8 f2/ψ for: Tr = 20 0C and the Relative Humidity ψ in % ! Characteristic Sensitivity ps = pdir for Pel = 1W. 293 K = 200C) Reverberation Time T = 24 Ln10 V/cA ≈ 0. Characteristic Sensitivity Level (Sensitivity) Ls ≡ 20 Log(ps/pref) Energy Density w = p2/ρc2 Sound Intensity I = pv.20 Log(r) + 20 Log(Γ) 75 .1 √Tr ≈ 344 m/s (air.S Ln(1-α) ≈ 4mV + S[α + α2/2 + α3/3 + ..

Diffuse Sound Pressure and Level pdif2 = 4ρcPac(1-α)/A = 25ρcPacT(1-α)/V = 25ρcηPelT(1-α)/V Ldif = 10 Log(pdir2/pref2) Ldif = 10 Log(25ρc/pref2) + 10 Log(Pac(1-α)T/V) = 134 + 10 Log(ηPel(1-α)T/V) = 120 + 10 Log(25Pac(1-α)T/V) 10 Log(25ρc/pref2) ≈ 134.90 Tpdif2/pdir2 ≤ 9T (% !) Average frequency spacing of adjacing maxima ∆fmax ≈ 4/T (Kuttruff.70) Level difference (between most probable maximum and mean value) ∆L = 10 Log(Ln(BT)) (Kuttruff.057 √(QV/(1-α)T) Direct-to-Reverberant-Ratio D/R = wdir/wdif = pdir2/pdif2 = rg2/r2 Loss of Consonants ALcons = 200 r2T2/V ≤ 9T (Peutz) ALcons = 0. 10 Log(ρc/pref2) ≈ 120 Total Sound Ltot = 10 Log(ptot2/pref2) ≈ LP + 10 Log[Γ2/(4πr2) + 4(1-α)/A] pref2/ρc ≈ Pref. p.10-12 (W) Reverberation Radius wdir(rg) ≡ wdif rg2 = QA/(16π(1-α)) = (6 Ln10/4πc) QV/(1-α)T ≈ 3. Pref ≡ 1.20 10-3 QV/(1-α)T rg ≈ 0. LP = 10 Log(Pac/Pref). 71) 76 . p.

07 .02 .06 .01 .07 .04 .01 .15 Bricks clay .15 CHAIRS LUX .39 .58 CHAIRS WD .60 Breeze blocks rough .06 .02 Chip board 16mm on 3cm air .04 .10 .00 .90 .02 .19 .70 .71 CELOTEX .93 .57 .07 .34 .04 .44 BAFFLE1 .10 .30 .60 .07 .03 .35 .00 .02 .07 .11 .51 .20 .04 .97 1.90 .80 .03 .02 .32 .14 .02 CARPT COMM .00 1.15 .59 FAB.73 .59 .92 1.09 .26 .00 1.79 .60 CINDBLK S .60 .0 .60 1.27 .04 .10 .10 1"THK AP INSULATION W/FOIL SURFACE .00 1.15 .18 .05 .00 1.97 1.18 .40 .91 .04 CONCRETE S .03 .05 .02 .5# W/FABRIC COVER .02 .19 .01 .02 Door .01 .52 .07 Bricks compressed clay .03 Unglazed bricks painted .92 .48 CARPT PAD .02 .62 .65 .00 .30 .15 .54 .38 ELASCON FIB/BD AIR FIB/BD PTD FIB/BD UP FIB/GLS 1" FIB/GLS 2" FIB/GLS1"A FIB/GLS2"A FIB/GLS4"A FLEXBOARD FLOOR CONC FLOOR TILE FLOORS 1 FLOORS 2 FLOORS 3 FLOORS 4 FOAM 2" FOAM 3" FOAM 4" GLASS 1 GLASS 2 GLS/WL 1" GLS/WL 2" .25 .24 .03 .20 .38 Draperies thin .00 1.73 .75 .60 .56 .23 .4 Surface material list with absorption coefficients MATERIAL 125 250 500 1000 2000 ABS BLOCK .06 .02 .92 1.30 .05 .02 .02 CINDBLK R .07 .31 .39 Draperies .57 .00 2"THK #6 core fabric covered panel .03 .00 1.00 1.00 1.11 .03 .95 4000 .15 .25 .67 .03 .96 .10 DOOR 2 .00 .02 .74 .98 .03 .15 .00 1.99 .00 1.04 .00 .30 .15 .02 .35 .52 .74 DOOR .32 .74 .39 .02 .6 .02 .01 .25 .13 BRICKS CC .75 BAFFLE2 .10 GLZD TILE RAND PERF 8"X16" TILES/GLS .01 .05 BRICKS CCP .04 CHPBRD16MM .02 .19 .38 CHPBRD 8MM .60 CINDER OR CONCRETE BLOCKS ROUGH .03 .02 .36 .92 1.56 1.05 .02 Chip board 25mm on 3cm air .03 .25 .13 .56 .79 .56 1.23 .38 CARPT HVY .11 .82 .20 AC PLASTER .00 1"THK #6 core fabric covered panel 1.89 .91 .84 .00 1.05 .45 LEATHER UPOLST SEATS UNOCCUP .07 Door 1 3/4"SOLID CORE WOOD .40 .60 .61 .95 1.02 .06 .04 .43 .02 .WELL UPOLST.09 .03 .80 .73 .00 1.60 CARPT INOT .31 .56 SUFACE TILE GLAZED/PERFORATED(AIR) .08 .10 .00 COLORSON2 .07 .97 COLORSON1 .18 .20 .11 .95 . SEATS UNOCCUP .07 .36 .07 .73 .38 .25 .03 .30 .35 .78 .41 CIRRUS .37 .20 .60 .39 Draperies medium .15 .07 .40 .12 .80 .90 Elascon 1/2" MOUNTED OVER 1" AIR SPACE 1/2" MNT SOLID BCK SOME PAINTED 1/2" MOUNTED/SOLID BACKING UNPAINTED AF 100 1" MOUNTING 4 AF 100 2" MOUNTING 4 AF 530 1" MOUNTING 4 AF 530 2" MOUNTING 4 AF 530 4" MOUNTING 4 3/16" ASBESTOS MNT OVER 2" AIR SPACE Concrete floor with thin carpet LINOLEUM ASPHALT RUBBER CORK TILE CONCRETE OR TERRAZZO LINOLEUM ASPHALT RUBBER CORK TILE WOOD WOOD PARQUET IN ASPHALT ON CONCRETE SONEX 2" THICK ON FLOOR SONEX 3" THICK ON FLOOR SONEX 4" ON FLOOR LARGE PANES HEAVY GLASS ORDINARY WINDOW GLASS 1" MNT/SOLID BACKING COVERED/FABRIC MNT 1" AIR SPACE OPEN WEAVE 77 .0 .00 CIRRUS 75 ceiling material armstong 1.07 .63 LATEX BACKING ON 40 OZ HAIRFELT .19 .45 CARPT LPAD .03 .84 .02 CHPBRD25MM .54 .56 .02 CORTEGA .20.04 .00 .07 .22 .88 .00 CONCRETE R .90 .05 .71 Commercial grade carpet .64 .30 DRAPE THK .70 Heavy carpet on concrete .30 CHAIR METAL OR WOOD UNOCCUP .42 .41 BRICKS C .14 .67 .50 .92 CIRRUS 75 .69 .77 .63 .99 .08 .61 CHAIRS LTH .10 .50 .90 .03 .12 .18 .44 .10 .5#W/2MIL PLAS COV PER SIDE 1.32 .60 .10 .00 .02 .35 .93 .11 .40 .35 .64 .04 Chip board 8mm on 3cm air .90 .08 .50 CINDER OR CONCRETE BLOCKS SMOOTH .70 .13 .20 .78 .37 .30 .30 .48 .02 .10 .06 .60 BREEZE B-S .04 .60 .50 Breeze blocks smooth .81 .15 .03 .11 .20 .15 .39 .79 CORTEGA ceiling material armstong .67 .84 1/2" Thick Zonolite .35 .50.04 .60 .15 .11 .15 .02 .02 .24 .54 .02 .10 .09 .37 .07 .10 .02 .08 .22 .31 2"THK ACOUS DECK 16GA PERF STL DECK .88 ACDECK 2.0 .70 .07 .03 .77 .35 .51 .05 .30 .65 .63 .02 .25 .60 .5 .14 .24 .23 .77 1.03 .15 .00 BREEZE B-R .96 CIRRUS ceiling material armstong 1.98 1.03 .60 .06 .85 .08 Concrete wall or floor (Rough) .72 .06 DRAPE MED .15 .00 1"X1.88 .14 .05 .75 ON 40 OZ HAIRFELT OR FOAM RUBBER .05 .01 .90 8000Hz DESCRIPTION IN DETAIL .02 .46 1.02 .14 .07 .09 .81 .5"X1.29 .10 .01 .55 ACOUSTILE .15 .62 .30 .06 Concrete wall or floor (Smooth) .32 DRAPE THIN .05 .14 .16 .07 .00 .03 .80 Indoor-outdoor carpet .32 .15 .09 .10 .89 .19 .70 .30 .70 .

92 .93 .51 .04 .01 .10 .99 .20 .29 .80 .86 .45 .40 .01 .61 .15 .00 .09 .02 .03 .5mm on >30cm air 1/2' DRYWALL 5/8" THK MOUNTED 16" CNTR WITH GLS 1/2" NAILED TO 2X4 16" O.60 .01 MASONITE .78 .01 .03 .5mm on 5cm air Plaster board 12.15 .19 .75 .07 .00 LINEAR .76 1000 .05 .09 1.05 .31 .24 .02 .02 .55 .98 .07 .80 .11 .05 .98 .55 .10 .69 .02 .00 .57 .00 .11 .55 .43 .93 .00 .25 .02 .30 .40 .86 .02 .31 .5mm on 5cm air Plaster board 12.05 .77 NUB WALL1" .02 .02 .00 .00 .70 .05 1.36 .15 .54 .11 .02 .07 .10 .05 .16 .07 .98 .02 .00 1.02 .03 .18 .05 1.83 .02 .5MM GYP12.20 .05 1.02 .95 .68 .92 1.73 .05 .09 250 .70 .10 .90 .52 .82 .MATERIAL GLASS-ROOF GRAVEL GRASS 1" GRASS SPRT GYP 12.11 .97 .05 .02 .75 .20 MARBLE .5MMB GYPBRD 1/2 GYPBRD 5/8 GYPSUM BRD HARDBOARD 125 .03 .84 .95 .09 .98 .01 .85 .02 .07 .08 OMEGA .99 8000Hz .90 .05 .05 1.12 .0 .76 .00 .00 2000 .83 .3 .90 .34 .10 1.99 DESCRIPTION IN DETAIL Movable glass roof GRAVEL LOOSE MOIST 4" THK GRASS MARION BLUEGRASS 2" HIGH GRASS FOR SPORTFIELDS (SOCCER) Plaster board 12.24 .90 .05 .18 .75 .28 .12 .13 .28 .95 .50 .20 .05 LAPENDARY11.11 .04 .65 .25 .65 .61 .70 .02 .98 1.52 .60 .90 .04 .00 1.34 .10 .30 .10 .45 .00 .30 .82 .05 .08 .10 .60 .00 .00 .20 .85 .64 .12 .32 .29 .61 .03 .05 1.78 .70 .02 .07 78 .87 .81 .20 .30 .22 .07 .51 .02 .06 .50 .98 .40 .38 .60 .75 .05 1.08 .00 .18 .45 .00 .30 . 1/8" MOUNTED OVER 2" AIR SPACE LAKE MODERATLY SMOOTH 2"THK LAPW/PERF PLAS COVER linear Frequency Dependence GLAZED TILE 1/2" MNT/OVER 1" AIR SPACE MASONRY PAINTED 1" NUBBY ceiling material armstong 1" NUBBY wall materials armstrong Carpet like wall cover for acous tmt Wooden parquet floor Pillow-baffle 1" thk mounted hanging ROUGH FINISH ON LATH SMOOTH FINISH ON LATH GYPSUM OR LIME SMOOTH FINISH ON TILE Plaster board 9.14 .02 .25 .55 .40 .05 .05 .84 .92 .70 .34 .04 .30 500 .40 .30 .02 .95 .26 .26 .04 .90 .08 1.03 .92 1.02 .11 .15 .09 .32 .69 .19 .95 1.88 .25 .11 .05 .86 .75 .02 .0 .05 .04 .25 .08 .01 .03 .10 .08 1.02 .66 .77 .5mm on 3cm air Contruction #8 2 layer 5/8" Plaster board 9.09 .06 .15 .04 .92 1.60 .80 .80 .57 .75 .17 .04 .20 1.30 .30 .64 .09 .02 . 1/4" MNT/OVR 3" AIR SPACE 2"GLUED TO 2 1/2" PLASTER ON LATH 3/8" Plywood panelling 6mm on 5cm air SWIMMING POOL Congregation in wooden pews Public on thick upholstered chairs Public on thin upholstered chairs Public on wooden chairs 2" THICK MNT/SOLID BCK MOUNTED OVER 1" AIR SPACE MOUNTED OVER 2" AIR SPACE Delta sorb AS2 on 25 cm space RPG DIFFUSER UNPAINTED ABSORPTION RPG DIFFUSER PAINTED ABSORPTION SANSERRA TRAVERTONE SCORED BEVELED TEGULAR CIRRUS SECOND LOOK CORTEGA CEILING MATERIAL SNOW FRESHLY FALLEN 4" THK SOIL ROUGH SONEX 2" THICK ON FLOOR SONEX 3" THICK ON FLOOR SONEX 4" ON FLOOR Sprayed cellulos fiber 1"on concrete SS60 FR701A Wall materials armstrong SS85 Classic vinyl wall materials SS85 FR701A Wall materials armstrong LAKE/POND .92 .90 .10 .48 .08 1.03 .60 .05 .00 .02 .06 .10 .85 .02 .69 .09 .83 .10 .14 .62 .20 .90 .03 .10 .00 1.01 .95 .09 .90 .10 PARQUET FL PILLOWBAFL PLAST/LTHR PLAST/LTHS PLAST/TILE PLASTERBD1 PLASTERBD2 PLASTERBD3 PLATGLS1/4 PLYWD 1/2 PLYWD 1/4 PLYWD 2" PLYWD 3/8 PLYWD 6MM POOL/SWIM PUB IN WDP PUBLIC TKC PUBLIC TNC PUBLIC WC ROCKWOOL 1 ROCKWOOL 2 ROCKWOOL 3 ROOF AS2 RPG DIFSR1 RPG DIFSR2 SANSERRA T SBV CIRRUS SNDLKCORTG SNOW SOIL SONEX 2" SONEX 3" SONEX 4" SPRAY ACOU SS60FR701A SS85CLASSC SS85FR701A .60 .94 .94 .82 .82 .00 .04 .05 .00 .55 .70 .25 .71 .00 .70 .05 .07 .42 .05 .02 .04 .32 .04 .68 .17 .5mm on >30cm air 1/4" PLATE GLASS 1/2" THICK OVR/2"/4" AIR SP.24 .12 MASONRY PT .71 .80 .05 .02 .09 .93 1.92 .70 .01 .65 .50 .25 .95 .76 .99 .02 .0 .32 .77 .90 .10 NUB CLG 1" .88 .05 .60 .86 .83 1.05 1.02 .07 .00 .02 .01 .01 .5mm on 3cm air Plaster board 12.00 4000 .40 .90 .C.99 .02 .04 .75 .03 .51 .91 .05 .72 .31 .02 .10 .86 .11 .05 .93 .07 .02 .20 .00 1.77 .09 .29 .28 .60 .03 .00 .05 .92 .02 .80 .10 .5MM GYP 2X 5/8 GYP 9.

40 .37 .08 .1 13.07 .10 .15 .60 .15 .50 .80 .17 .84 .07 .15 .02 .01 .20 .30 .90 1.07 .37 .05 .07 .90 1.85 .85 .32 UNGL BRICK .25 5.30 .33 .25 .02 .25 .30 .MATERIAL STDNTS WC STEEL TBLE TP WD TEG CIRRUS TERRAZZO TILE FLOOR TILE GLAZD TREES TRIBUNE ES TUNDRA 1" 125 .Humidity RH=50% 0.08 .25 .14 VELOUR LT .70 .90 1.46 .75 .10 .70 .87 .02 .02 .00 500 .79 .20 .07 .01 .10 .10 .50 .02 .15 0.02 .05 .99 .01 .11 .Humidity RH=80% 0.74 .10 .Humidity 79 .02 .08 .20 .01 .04 .23 0.47 .86 .85 .30 .02 .02 .50 .07 .60 .72 .09 .40 .65 .05 .11 .06 .10 .80 .70 .06 .62 .60 .16 0.65 .15 .31 .02 .89 .84 .01 .80 .55 .05 .09 .05 .1 Air with 20% Rel.37 0.05 .01 .85 .10 .00 2000 .07 .10 .10 0.25 .80 .07 .35 .30 .50 .80 .2 23.30 .07 .72 .90 1.14 0.2 Air with 50% Rel.20 .70 .35 .06 .05 .81 Students in wooden seats .20 .03 .1 19.70 .02 .02 .01 .06 .06 .09 .30 . 14oz draped to 1/2 area Water Swimming pool Double pane glass Single pane glass Wooden floor on beams Wooden floor covered with linoleum Wooden floor with thin carpet Wooden floor or linoleum on concrete Hardwood floor on beams Wood parquet in asphalt on concrete Wooden grid 90/15mm on 6cm air + gw Wooden grid 35/15mm on 2cm air Wooden grid 35/15mm on 2cm glaswool Wooden grid 35/15mm on 40cm air Wooden grid 35/15mm on 40cm glaswool Wooden grid 60/15mm on 40cm air + gw Wooden grid120/15mm on 40cm glaswool Wooden grid 90/15mm on 40cm air + gw Wooden grid 90/15mm on 40cm air only Wooden grid 90/15mm on 6cm air only 3/8" to 1/2" thick ovr/2"/4" air sp.40 .10 .50 .10 .04 .90 1.Humidity (acc.05 .0 .40 0.70 .00 Atmospheric absorption (m/km) versus Rel.06 .50 .01 .40 .10 .65 .05 .15 .12 1.05 .02 .02 .92 .30 .00 8000Hz DESCRIPTION IN DETAIL .11 .07 .10 .00 1000 .03 .10 .05 .80 .01 .20 .56 1.50 .10 .90 1.60 .70 .06 .33 .02 .36 .08 .5 47.60 .80 .02 .30 .35 .0 .10 .11 .03 .26 .90 .17 .07 .01 .2 Air with 60% Rel.50 .Humidity RH=40% 0.07 0.Cyril Harris) Label 125 250 500 1000 2000 4000 8000 Description (T=200C) RH=20% 0.60 .03 VELOUR MED .87 .94 .07 .07 .07 .60 .03 .50 .12 .02 .0 .60 .08 .40 .71 .05 .20 .05 .55 .60 .01 .10 .10 .60 .08 .70 .15 .87 .49 .0 .05 .07 VELOUR HVY .02 .34 0.18 0.75 .03 .03 .40 .85 .70 .07 .60 .44 .05 .97 .07 WATER RVR WATER POOL WINDOW DP WINDOW SP WOOD FLR 1 WOOD FLR 2 WOOD FLR 3 WOOD FLR 4 WOOD FLR 5 WOOD FLR 6 WOOD GRID0 WOOD GRID1 WOOD GRID2 WOOD GRID3 WOOD GRID4 WOOD GRID5 WOOD GRID6 WOOD GRID7 WOOD GRID8 WOOD GRID9 WOOD PANEL WOODPANEL1 WOODPANEL2 MIRROR α = 10% α = 20% α = 30% α = 40% α = 50% α = 60% α = 70% α = 80% α = 90% α =100% .20 .02 .20 .01 .0 .91 2.31 .06 .70 .80 .39 4.82 2.80 .12 .04 .10 .00 .80 .40 .40 .02 .30 .20 .7 Air with 40% Rel.99 .35 .50 .70 .60 .01 .00 4000 .40 .50 .30 .56 .33 .40 6.97 2.02 .34 .20 .99 .02 .06 2.90 .28 .80 .99 .10 .92 .14 .15 .08 0.10 .41 .20 .00 Table top wood Tegular cirrus ceiling Concrete or terrazzo Linoleum asphalt rubber cork tile Glazed tile Trees firs 20 sq ft grd area pertree Tribune with empty seats (Stadium) 1"open plan tundra ceiling material Ultima ceiling material armstong Unglazed brick 1"thk urethane foam panel Heavy 18oz draped to 1/2 area Light (10 oz) hung touching wall Med.28 14.80 .50 .08 .15 .49 .20 . Wood panelled wall 16mm on 4cm air Wood panelled wall 18mm on 4cm air Ideal Sound Reflector 10% Sound absorbing 20% Sound absorbing 30% Sound absorbing 40% Sound absorbing 50% Sound absorbing 60% Sound absorbing 70% Sound absorbing 80% Sound absorbing 90% Sound absorbing 100% Sound absorbing ULTIMA .Humidity RH=60% 0.23 .10 .24 .60 .50 .15 .10 .43 1.17 .60 .01 URETH PANE .09 .10 .10 .10 .03 .11 .07 .60 .07 0.20 .70 250 .30 .04 .40 .6 16.09 .06 .80 .90 1.10 Steel panel or wall or surface .06 0.20 .03 .50 .20 .05 .02 .59 7.10 .60 .60 .06 .03 .02 .08 .07 5.04 .10 .05 .04 .40 .20 .32 .27 .80 .65 .50 .07 .20 .3 Air with 80% Rel.

caution messages and sub-menus add further to the ease of operation and prevent operation errors. distributed by Renkus Heinz.. have gained worldwide acceptance and recognition as the most advanced and accurate of all design programs. Every Licence receives An ongoing license with no fixed termination or renewal dates and no periodic license review or renewal fees. create shape options “on-the-screen” graphic editing. Now available in entirely new versions. Screen processing capabilities allow the easy assembly of slide shows. and multiple views for easy visualization. intuitive loudspeaker aiming. Engineers and Acoustical Consultants. all simplify and speed up the modeling process. 80 . Software Support EASE and EASE JR licenses also include an extensive operation manual and comprehensive software support. including AutoCAD. BitMap and TIFF files allow screens to be exported to other presentation programs. Plus. An acoustic simulation program called "E A S E" (Electro Acoustic Simulator for Engineers). DXF file exchange allows 3-D room drawings to be imported from virtually all architectural CAD programs. the scientists behind EASE. Open Loudspeaker & Surface Materials Data Bases Extensive open databases include loudspeaker data on most major manufacturers products and on over 100 surface materials. EASE. including loudspeaker performance data and physical data files form over 25 major loudspeaker manufacturers. Simplified Room Modeling + DXF File Exchange A library of prototype rooms. reduces installation time and makes it easier to share design results with clients.21. These databases are continually expanded and up-dated. Pop-up menus. multiple viewing angles with spin and rotate options. are dedicated to continually improving the program. Feistel. This programs takes the guesswork out of system design. and spectacular “Speaker” and “Spectator” views all enhance visual presentations. eye catching cut-away views. As one of Europe’s leading acoustical consultants. EASE workshops are held on a continuing basis. The license covers the complete software package. helps eliminate costly mistakes. permit color changes and the gathering of slides from numerous projects into a single show. Ahnert of ADA and Dr. User Friendly EASE and EASE JR operate with a Mouse in a familiar “windows” like environment. they have taken a quantum leap forward in performance and versatility. simplifies the design of a sound system and allows its performance to be accurately predicted. Ahnert uses the program in his own work. Outstanding Graphic Capabilities Full color screens & color prints. Ongoing program improvements Dr.0 E A S E Software Advanced Acoustical Design and Analysis Tool for Contractors. the Electro Acoustic Simulator for Engineers. Dr. A useful feature is the ability to enter your own favorite loudspeaker or surface material databases..

Direct SPL. side or top views can be selected. to take full advantage of the program’s color graphic capabilities. open loudspeaker and surface materials data bases.Proven Accuracy The accuracy of mirror image modelling and of EASE simulations has been conclusively documented in over three years of use in all types of venues. Delays of up to 999 ms allow accurate simulations in systems with cluster delay lines. Individual reflections can be selected. C7. Alcons. Measured RT times and those calculated with Eyring. displays direct sound arrival times and angles. especially in complex rooms or rooms with multiple speakers. This has resulted in extensive verification of the correlation between EASE simulations and actual measurements. Extended time frame investigations can be used to produce the reflectograms in the EARS auralization process. end. magnitude and phase. Unique Acoustical Probe Allows detailed analysis of suspected problem areas. Epson standard and HP Paintjet. simplified room modeling techniques. Computer Requirements EASE and EASE JR run under MS-DOS. It features the same user friendly program. Laserjet and Deskjet printers are supported. on IBM or 100% compatible computers with an EGA or VGA graphics system. Accurately shows shadowing and the effects of phase interference. 3D isometric. viewed and analyzed. Most of the realizations were measured using Bruel & Kjaer sound level meters and RASTI equipment for verification. provides ETC and EFC curves. Initial Time Delay (delay between first two direct signals) and Coverage Overlap.) Beam Show Accurately traces reflections allowing sound paths to be easily visualized. Also maps First Signal Arrival Times. EASE Junior EASE JR was developed specifically for the system designer who does not need the detailed acoustic analysis features like the Acoustical Probe. 81 . Movie Uses Ray Tracing to provide an animated display (movie) of the dispersion of sound into and around the room This feature quickly shows potential reflection problems and thus provides excellent troubleshooting assistance. Ray Tracing Reduces the calculation time required. Sabine or Ray Tracing methods can be used in the intelligibility projections. This enables quick identification of surface areas that may require acoustical treatment. Rasti. Extensive Simulation Menu Maps. (EASE also offers calculations based on Schroeder’s back integration method. outstanding graphic capabilities. and extensive simulation menus complete with Beam Show. Total (direct + reflected) SPL. C50 and C80 projections onto the audience areas in multiple illuminating presentation modes. Shows comb filtering and identifies the exact frequency and level of severe peaks and notches. Ray Tracing and the Movie module.

Receiver Type 4419 which analyses the signal and calculates the RASTI value. 82 . ambient noise analysis and system performance checking. enabling measurements in the time and frequency domain. due to storage capabilities. various decay settings. Compact and transportable. (Production stopped) 1.various scales and an optional PC connection. 22. mains operated 2. Performs highly diverse measurement tasks quickly and precisely 3. 22.1 MLSSA .3 B & K Speech Transmission Meter 3361 Speech Transmission Meter Type 3361 (Production stopped) Consisting of Transmitter Type 4225 and Receiver Type 4419 This set measures RASTI values directly using two instruments: Transmitter Type 4225 which sends out a special test signal. but its primary application is in the area of audio and acoustics measurements. Easy to operate due to built-in generator & plotter intelligence MEASUREMENTS : Reverberation time Frequency response of loudspeakers Frequency response of amplifiers & equalizers Frequency response of acoustic feedback loop 22. PC based transfer analysis of linear systems.22.Transfer Analyzer MLSSA is a powerful system analyzer based on maximum-length sequences. enabling rapid and objective measurement of the quality of speech intelligibility and provide further information of diagnostic value.2 Neutrik Audiograph 3300 The Neutrik Audiograph 3300 is a universal audio/acoustic analysis system. This unit can be used for equalisation using pink noise. The instrument is handy for field measurements.4 Gold Line Audio Spectrum Analyzer DSP30 The Real Time Analyzer DSP30 is a portable Digital 1/3 Octave Sound Level Meter.0 Measuring Equipment 22. The instruments are battery powered and fully portable.

The transfer functions of filters. MLSSA is a single channel analyzer which employs a special type signal called a maximumlength sequence (MLS) as the preferred alternative to the conventional white noise stimulus.the most fundamental description of any linear system . 6.Transfer Analyzer MLSSA is a powerful system analyzer based on maximum-length sequences. The transfer function. but its primary application is in the area of audio and acoustics measurements. From the transfer function MLSSA then derives and displays the frequency response. MLSSA should not run under Microsoft Windows. low frequency (1 kHz) transfer functions with 0. transducers and amplifiers in the measuring channel. it performs transfer analysis of linear systems. The microphone response data obtained from your microphone's calibration data sheet. 11. 8. 12. 10. 83 .from which a wide range of important functions are derived through computeraided post-processing. 2. The time-bandwidth product of a MLSSA measurement can exceed 20. 5. Below is a list of important functions MLSSA can derive by post-processing the impulse response. 2. You can use this bandpass filter to display filtered versions of any post-processed function. Unlike white noise. The periodic nature of the MLS means zero windowing error as long as the entire period of the sequence is used to make the measurement (which MLSSA does automatically). 1. The deterministic nature of the MLS means that it can be pre-computed and need not be measured simultaneously with the system response. 4. The MLSSA software consists of dedicated MLSSA software which runs on a IBM compatible PC. 9. Step response Energy-time-curves Schroeder reverberant decay curves Cumulative energy STI & RASTI for speech intelligibility Cumulative spectral decay waterfall 7. 3. An important advantage of the MLS approach is its ability to make extended wideband impulse response measurements containing up to 65535 points. The MLSSA hardware consists of a full size A2D-160 sampling digitizer board (AT-16bit or PC-8bit) with two AFM-50 antialiasing filter modules.22. a maximum-length sequence is deterministic and periodic yet still retains many desirable characteristics of white noise. is obtained by applying an FFT to a segment of the impulse response. Energy-time-frequency waterfall Wigner distribution waterfall Polar response waterfall Early/late ratios Reverb/direct ratio Sound Pressure Level (dB-SPL) In addition. phase response and group delay.5 MLSSA . Thus you can easily correct for minor microphone response errors as well as other components in the measuring chain.000. The MLS technique measures the impulse response . MLSSA provides a fully programmable digital bandpass filter which allows calculation of reverberant decay curves and reverberation time for any octave or 1/3 octave band. Therefore this MLS approach easily makes wideband (20 kHz) transfer function measurements with true 1 Hz frequency resolution or. for instance.065 Hz resolution. to be installed in a (transportable) PC. MLSSA also supports two references which can be used as corrections for: 1.

MLSSA EXAMPLE IN TIME DOMAIN MLSSA EXAMPLE IN FRQUENCY DOMAIN 84 .

494 ¦ 0.746 ¦ 0.519 ¦ 0.977 ¦-0.510 ¦ 0.6 ¦ 42.426 ¦ 3.893 ¦ 0.769 ¦ ¦ 3.9 ¦ 128.469 ¦ 2.640 ¦ 0.736 ¦ ¦ 10.60 ¦ 0.206 ¦ 3.8 ¦ 55.74 ¦ 1.733 ¦ 2.461 ¦ 3.591 modified) ALcons= 7.629 ¦ 0.993 ¦-0.1 ¦ 51.458 ¦ 0.143 ¦ ¦RT-20dB [s] ¦ 2.437 ¦ 3.6 ¦ 98.550 ¦ 0.35 ¦ 2.031 ¦ 2.721 ¦ +---------------------------------------------------------------------+ STI value= 0.00 ¦ 0.745 ¦ ¦ 12.000 ¦ 1.592 ¦ 0.000 ¦ 0.296 ¦ ¦(-5.10 ¦ 2.000 ¦-0.486 ¦ 0.417 ¦ 0.9 ¦ 52.440 ¦ 0.532 ¦ 0.590 ¦ 0.485 ¦ 0.920 ¦ 0.741 ¦ ¦ 8.505 ¦ 0.00 ¦ 0.00 ¦ 0.154 ¦ 0.306 ¦ 0.111 ¦ ¦C80 [dB] ¦ -2.716 ¦ 0.4 ¦ 20.433 ¦ 0.746 ¦ ¦ 4.-35) r ¦-0.706 ¦ 0.7 ¦ 27.742 ¦ 0.453 ¦ 1.999 ¦-0.00 ¦ 0.5 ¦ 56.MLSSA EXAMPLE LEVELS & REVERBERATION IEC Octave Band Acoustical Parameters +------------------------------------------------------------------------------+ ¦ Band ¦ 3 ¦ 4 ¦ 5 ¦ 6 ¦ 7 ¦ 8 ¦ 9 ¦ 500.30 ¦ 2.994 ¦-0.832 ¦ ¦ 2.746 ¦ 0.498 ¦ 1.353 ¦ 0.431 ¦ 0.527 ¦ 0.00 ¦ 6.25 ¦ 0.¦ ¦ Parameter ¦ 125 ¦ 250 ¦ 500 ¦ 1000 ¦ 2000 ¦ 4000 ¦ 8000 ¦ 4000 | +-------------+-------+-------+-------+-------+-------+-------+-------+--------¦ ¦S [dB-SPL] ¦ 45.698 ¦ 0.9 ¦ 70.21 ¦ 1.200 ¦ 3.999 ¦ 1.116 ¦ 0.6% Rating= FAIR 85 .794 ¦ 0.882 ¦ 2.582 ¦ 0.2 ¦ 75.211 ¦ ¦(-10.214 ¦ 0.495 ¦ 0.6 ¦ 123.8 ¦ 23.524 ¦ 0.957 ¦ ¦EDT-10dB [s] ¦ 2.357 ¦ 0.999 ¦-1.761 ¦ 0.000 ¦ 1.855 ¦ 0.994 ¦-0.998 ¦-1.999 ¦-0.00 ¦ 0.000 ¦-1.548 ¦ 0.2 ¦ 43.998 ¦ ¦ 0.18 ¦ 0.798 ¦ ¦ 2.808 ¦ 0.729 ¦ 0.9 ¦ 80.995 ¦ ¦RT-USER [s] ¦ 2.1 ¦ 78.63 ¦ 0.575 ¦ 0.572 ¦ 0.518 ¦ 0.2 ¦ 51.740 ¦ ¦ 6.932 ¦ ¦ 1.617 ¦ 0.681 ¦ 0.8 ¦ 76.3 ¦ 35.6 ¦ 27.519 ¦ 0.137 ¦ 0.7 ¦ 65.636 ¦ 0.9 ¦ 78.8 ¦weighted¦ ¦SNR [dB] ¦ 13.528 ¦ 0.999 ¦-0.210 ¦ 0.28 ¦ 5.337 ¦ 3.4 ¦ 62.993 ¦-0.4 ¦ 78.994 ¦ ¦(-5.056 ¦ 3.707 ¦ 0.221 ¦ 0.688 ¦ 0.599 ¦ 0.21 ¦ 6.-25) r ¦-0.999 ¦ ¦RT-30dB [s] ¦ 2.4 ¦ 23.015 ¦ 3.8 ¦ 54.736 ¦ 3.8 ¦SPL¦ ¦N [dB-SPL] ¦ 31.997 ¦-1.000 ¦-1.800 ¦ 0.710 ¦ 0.557 ¦ 0.87 ¦ 3.217 ¦ 2.045 ¦ ¦TS [ms] ¦ 173.604 ¦ 0.602 ¦ 0.995 ¦-0.16 ¦ 2.3 ¦ 75.6 ¦ 22.000 ¦ 0.30 ¦ 0.445 ¦ 0.521 ¦ 0.706 ¦ 3.5 ¦60.544 ¦ 0.576 ¦ 0.0 ¦Averages¦ ¦C50 [dB] ¦ -4.451 ¦ 0.9 ¦ 70.783 ¦ 0.882 ¦ 2.757 ¦ ¦ octave TI ¦ 0.15 ¦ 0.685 ¦ 0.07 ¦ 1.777 ¦ ¦D50 [%] ¦ 27.903 ¦ ¦ 1.515 ¦ 0.742 ¦ 0.50 ¦ 0.952 ¦ ¦ 0.492 ¦ 0.000 ¦-0.-25) r ¦-0.030 ¦ 2.586 ¦ 0.189 ¦ 0.50 ¦ 0.735 ¦ ¦ 5.5 ¦ 78.589 ¦ 0.761 ¦ 0.470 ¦ 0.998 ¦ +------------------------------------------------------------------------------+ MLSSA EXAMPLE SPEECH TRANSMISSION INDEX +---------------------------------------------------------------------+ ¦Frequency-Hz ¦ 125 ¦ 250 ¦ 500 ¦ 1000 ¦ 2000 ¦ 4000 ¦ 8000 ¦ +-------------+-------+-------+-------+-------+-------+-------+-------¦ ¦level dB-SPL ¦ 41.0 ¦ 72.443 ¦ 1.575 (0.843 ¦ 0.000 ¦-1.596 ¦ 3.80 ¦ 0.734 ¦ 0.000 ¦-0.000 ¦-1.2 ¦116.994 ¦-0.755 ¦ 0.995 ¦-0.535 ¦ 0.273 ¦ 3.572 ¦ 0.03 ¦ 7.8 ¦ 72.886 ¦ 3.816 ¦ 0.000 ¦ 1.0 ¦ ¦m-correction ¦ 1.714 ¦ 0.716 ¦ 3.637 ¦ 0.33 ¦ 1.073 ¦ 2.460 ¦ 0.531 ¦ 0.712 ¦ 3.8 ¦ 150.00 ¦ 0.869 ¦ ¦ 1.205 ¦ 3.5 ¦ 66.3 ¦ 57.529 ¦ 0.504 ¦ 0.6 ¦ 51.045 ¦ 0.517 ¦ 0.061 ¦ 2.752 ¦ 0.

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