Presents

A sample download on
Audio and Public Address Systems

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PUBLIC ADDRESS SYSTEM

Public address system

A Public Address System comprises of

a sound source an amplifier a loudspeaker

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The sound is an electrical signal generated by the sound source.

Anything can be the source of sound. A microphone, a playback device (CD player, cassette player etc).

The level of electrical signal (produced from the sound source) is increased by the amplifier that can be heard at sufficient volume from the loudspeaker.

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Loudspeaker receives electrical impulses from the amplifier and converts them into vibrations in the air, which our ear interprets as sounds.

Mixer is used while dealing with multiple sound sources. It mixes all the sound signals and produces a single signal for the Amplifier. It also acts as pre-amplifier for some weak sound sources (microphone), which are not suitable for direct connection to an amplifier.

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What is sound ?
Sound is the quickly fluctuating pressure wave within a medium, which can travel widely in that medium. An audible sound is produced due to fluctuations in the air pressure and detected by the ear. Sound is potentially audible at a frequency between 20Hz and 20kHz.

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Energy flows with the sound pressure waves. It is diagrammatically represented by sine curve.

Physically sound wave is longitudinal and energy moves in the direction of wave.

The wave crests and troughs represent maximum and minimum pressures respectively.

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Air pressure fluctuations can be quite small or large and can occur slowly or rapidly.

The rate at which pressure fluctuates cyclically from higher to lower and to higher is called frequency. Frequency is expressed in cycles per second and its unit is Hertz.

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Sine curve

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In this pressure vs. Time graph, 0 corresponds to Mean Air pressure. Air Pressure first increases to 100 at time 1.25 msec, decreases to 100 at 3.75 msec and then returns to mean air pressure,0 at 5.0 msec.This entire cycle is called Period and expressed in Time. Here Period of this particular wave is 5.0 msec. Frequency of this wave = 1.0/0.005= 200 Hz.

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The air pressure variations with respect to mean air pressure is called Amplitude and is expressed as a function of Time (milliseconds or thousandths of a second).

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Weak sound is produced by small variations in pressure and large variations produce strong (or loud) sounds. In this figure, amplitude is lower than the earlier one. Frequency and amplitude vary independently. Here, six cycles take 10 msec, hence frequency is 600 Hz.

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Phase describes the physical properties of sound. Two sine curves are identical in amplitude and frequency but with different phases with respect to time axis.

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The physical property of amplitude corresponds to the perceptual quality of loudness and frequency to pitch. Loudness and pitch are related to the capabilities of our auditory systems whereas the physical properties of sound are not.

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Physical properties and sensory qualities of sound are having non-linear relationship. The increase of amplitude does not increase the loudness to the same extent and it is also same with frequency vs. pitch. On the contrary, sensory qualities grow smaller with every successive increase of physical properties of sound. Phase is not directly related to perceived sound quality.
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Simple sounds are those where pressure variations over time follow the sine or cosine function. Most of the sounds are of complex in nature and described as combinations of simple sound.

In this figure sound alternates between constant high and constant low pressure and its boxy shape termed as square wave. The square wave is very similar to sine wave having same pitch with different timbre.

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Fundamental Frequency is the frequency which gives rise to the pitch we normally hear when listening to a complex sound. Square wave requires a fundamental frequency (here 200 Hz sine wave) and a sequence of higher frequency sine wave components, which are called harmonics.

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Sound In Frequency Domain

In frequency domain sound can be described by the frequency vs. amplitude graph.

Lines represent the sound and are called line spectra. Here the line touches the frequency axis at 200 Hz and the length indicates the amplitude.

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Amplitude axis shows the measure of strength of pressure changes but not the absolute pressure and the direction of relative pressure change. Negative sound energy is not possible hence in this magnitude spectrum there is no zero value in the amplitude axis. Frequency axis is labeled in kilohertz.

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In the frequency domain spectrum, sound of different frequencies can be plotted in the same figure. Each line is one of the harmonics of the 200Hz frequency and the height indicates the amplitude of the sinusoid at that particular frequency. The phase relationships between harmonics are not possible here.
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Amplitude is expressed in decibel (dB). The harmonics expressed in dB are closely related to loudness.

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Line spectra associated with periodic signals and are not bounded by time. Voice sounds are varying from one period to another and are approximately periodic. These sounds are quasiperiodic. In this figure the axes are same except the figure of the amplitude. Amplitude is based on the units used in the digitized and windowed sound.

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The harmonic lines are now pointed bars and represent the presence of sound energy at different frequencies, which are close to the true harmonic frequency. These are called harmonic spectra. Sounds, not having identifiable tone are called aperiodic sounds. These are not periodic and the spectra associated with this are continuous spectra eg. Hissing sound, clap sound etc.

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Sounds, not having identifiable tone are called aperiodic sounds. These are not periodic and the spectra associated with this are continuous spectra eg. Hissing sound, clap sound etc.

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Sound Power level

Sound Power level indicates the sound energy radiated per second. It is expressed in decibels with respect to the reference power level. The reference power level is 1 pico-watt (pW). 1 pW = 1X 10-12 Watts. One Watt of sound power , Lw = 120 dB re one picowatt.
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Sound power is indirectly measured as sound pressure level at a specific distance for every direction in a reverberation room. There are two methods- comparison & direct. Comparison method The SPL of an item is compared with the SPL of standard Reference Sound Source. Direct method- Two processes Hemianechoic & Reverberation room method
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Hemianechoic method the SPL of the item is measured in all direction on the encompassing surface and these measurements are then combined to calculate the emitted sound power. Reverberation room method The SPL is measured at several points in that room and the values are then averaged. The sound power is computed from that average as: PWL = SPL + 10Log(A)-C.

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Decibel
The decibel is used to compare the sound pressure level (SPL) in air with a reference pressure. A logarithmic scale of amplitude which is roughly associated with our perception of loudness. Zero Decibels is near the threshold for hearing and each decibel increment in amplitude is roughly one Just Noticeable Difference in loudness.

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Human ear interprets loudness more conveniently when expressed in decibel than a linear scale. A dB scale is more convenient than a linear scale. Decibel values are in between 999 to +999. The formula for computing decibels is: Decibels = 20.0 * log(Amplitude/Reference) where Reference is generally something like the smallest perceptible amplitude fluctuation. Sound Pressure Level = 20 x log (p/0.00002) dB

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dBA
In 1936, American Standards Association published the first tentative sound level meter, which was sponsored by the Acoustical Society of America. Two frequency weighting curves A and B were modeled on response of the human ear to low and high levels of respectively.

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In 1969, the A-Weighting of sound (dBA) was defacto presumed to be the "appropriate" weighting to represent sound level as a single number (rather than as a spectrum). The "A" in dBA refers to an A weighted correction curve that's used to correct for the different perceptions of sound with pressure and frequency. The term dB without the "A" would infer no correction. dBA is a term associated with human perception of sound.
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Example of levels
Just audible is 10 dBA Soft whisper at 15 feet is 30 dBA A quiet office is about 40 dBA Air conditioner, normal speech, 60 dBA Noisy restaurant, freeway traffic, noisy office, 70 dBA Hearing protection recommended at 80 dBA Heavy truck in traffic measures 90 dBA Rock concert is 110 dBA Auto horn at 3 ft, maximal vocal effort results in 120 dBA Thunderclap is 130 dBA Jet air ops on a US Navy carrier deck is 140 dBA
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Reverberation
Reverberation is the persistence of sound in an enclosed space after the original excitation sound has ceased. It consists of a series of very closely spaced reflections, or echoes, whose strength decreases over time due to boundary absorption and air losses.

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Reverberation time refers to the time a sound takes to bounce around a room before being absorbed by the materials and air. Closed room without any sound absorbent materials have long reverberation time whereas that of with absorbent materials has short reverberation time. Reverberation time of a room depends on the volume of that room and the rate at which the sound energy is absorbed by the wall surfaces and the objects in the room.
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In an empty room, the reverberation time is proportional to the ratio of volume to surface. The reverberation time is the time required for the sound level to decrease by 60 dB.

RT60 = k (V/Sa) Where k (constant) = 0.161 (meter), = 0.049 ( feet). Sa = Total surface absorption, expressed in sabins. V = Volume of the room.

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Communications and speech intelligibility

Speech Intelligibility is expressed as a percentage of words, sentences or phonemes correctly identified by a listener or group of listeners when spoken by a talker or a number of talkers. It indicates the effectiveness or adequacy of a communication system or of the ability of people to communicate in noisy environments.
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Speech intelligibility can be measured directly by using live speech. Same recorded speech material is also used to compare different communication systems. Intelligibility tests are time consuming and consequently expensive. Speech communications are of three types: Unamplified speech, normally face to face, Amplified speech where the speech waveform is transmitted, and Vocoded or synthetic speech where the waveform is not transmitted.
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Speech reinforcement system ensures the clarity of the speaking voice to the listeners. Speech intelligibility is affected by a number of acoustic, electromechanical and electronic factors. The intrusion of unwanted sounds interfere with the speech signal. This effect is called masking,

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S/N Ratio
Signal-to-noise ratio represents the relationship between the strength of the speech signal and the masking sound and is expressed in decibels. S/N ratio should be greater than 0dB i.e. the speech is louder than the noise. At 0 dB the two are of equal strength; Negative values are associated with loss of intelligibility due to masking. Positive values are usually associated with better intelligibility.
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Broadband noise spanning 20 Hz to 4 kHz, masking most effectively and the signal has to be 12dB louder for 80% word recognition.

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Word Articulation vs. Signal-To-Noise: Effect of Broadband Masking Noise

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Narrow-band noise is less effective at masking speech than broadband noise but the degree of masking varies with frequency.

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Word Articulation vs. Signal-To-Noise:Effect of BandLimited Masking Noise

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High-frequency noise masks only the consonants, and its effectiveness decreases, as the noise gets louder. Low-frequency noise at high sound pressure levels masks both vowels and consonants.

Human voice is also having the masking effect (distractor effect) particularly at or below 0dB S/N.

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Word Articulation vs. Signal-To-Noise: Effect of Competing Distractor Voices

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Frequency response affects intelligibility. High-quality speech systems required to cover the frequency range of about 80 Hz (for deep male voices) to about 10 kHz (for best reproduction of consonants, which are crucial to intelligibility). Clipping the peaks of the speech signal, and then amplifying it to restore its peak-to-peak amplitude can improve intelligibility in communication systems and useful when the signal-to-noise ratio is very poor. Intermodulation a type of distortion is very destructive to intelligibility.
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Word Articulation vs. Signal-To-Noise: Effect of Clipping

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Speech Intelligibility Measurement

Statistical tests are most accurate and reliable but complicated, time-consuming and require extensive statistical analysis. A number of automated, machine-based test methods have emerged that fall into two basic categories: Analyses of the reverberant field, Measurements based on signal-to-noise ratio.

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Reverberation Analysis
These tests explain the reverberant qualities of a space and speech intelligibility of that space but unable to take into account the majority of the factors that can affect a speech reinforcement systems performance.

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The most commonly used methods are: %ALcons Direct-to-Reverberant Ratio Useful-to-Detrimental Sound Ratios Early-to-Late Sound Energy Ratio

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%Alcons
Percentage Articulation Loss of Consonants is an indication of the loss of speech intelligibility that occurs in difficult acoustic environments. This machine measures the intelligibility from the Direct-to-Reverberant Ratio and the Early Decay Time. Lower values are associated with greater intelligibility.

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Disadvantages
This is based on measurements in a single one-thirdoctave band centered on 2 kHz; all other frequencies are ignored. Systems frequency response should be verified for meaningful score of %Alcons. At longer reverb times the %ALCONS is very high, and this definitely affects speech communication.
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The method does not account for factors that dramatically affect intelligibility, e.g. signal-to-noise ratio, the background noise spectrum, distortion, late reflections or echoes, system frequency response, compression, non-linear phase, equalization and acoustic power.

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Direct-to-Reverberant Ratio
Measure the ratio of the intensities of the direct sound and reverberation and measurements are made in a single frequency band, usually centered on 1 kHz. C50 expresses speech clarity as the energy ratio of the first 50 milliseconds of direct sound to the overall steady-state reverberation, with 0 dB being the minimum acceptable value and +4 dB or above preferred. More reliable and repeatable than %Alcons.
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Useful-to-Detrimental Sound Ratios

The logarithmic ratio between the energy of sounds that of useful to intelligibility and those that are detrimental to it, expressed in decibels. Useful sounds are the integrated energy of speech sounds arriving within the first 50 or 80 milliseconds after the direct sound, and detrimental sounds are the sum of later-arriving speech energy and ambient noise.

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Early-to-Late Sound Energy Ratio

ELR is similar to C50 but is weighted for speech and incorporates measurements in more than one frequency band. Factors other than reverberation are not taken into account.

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Signal-to-Noise Methods
Several instrument-based tests based on signal-to-noise measurements have evolved. They are: AI - Articulation Index STI - Speech Transmission Index RASTI -Rapid Action Speech Transmission Index SII - Speech Intelligibility Index

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Articulation Index
One of the earliest attempts to measure by machine the intelligibility of a speech transmission system, the Articulation Index was developed by Bell Telephone Laboratories in the 1940s. The Articulation Index measures the intelligibility of hearing speech in a given noise environment.

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The AI value is expressed either as a factor in the range zero to unity or as a percentage. The higher the AI value, it is easier to hear the spoken word. An AI of 0.3 or below is considered unsatisfactory, 0.3 to 0.5 satisfactory, 0.5 to 0.7 good, and greater than 0.7 very good to excellent.

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Contribution towards AI value is evaluated from the difference of an 'idealised speech spectrum' and the third octave spectrum levels of the background noise. Each contribution is multiplied by a weighting factor specific to the particular third octave band.

The sum of all the contributions is the AI value.

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Speech Transmission Index

Developed in the early 1970s, the Speech Transmission Index (STI) is a machine measure of intelligibility whose value varies from 0 (completely unintelligible) to 1 (perfect intelligibility). Lexington's Speech Transmission Index program is based on the Modulation Transfer Function (MTF). MTF measures the reduction in modulation of a test signal due to noise, temporal and non-linear distortion i.e. the amount of modulation reduction versus modulation frequency, Fm.
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Fourteen modulation frequencies are used in the STI stimulus signal. The modulating sine waves range in frequency from 0.63Hz to 12.7Hz in 1/3-octave steps. The STI program calculates modulation reduction factors for seven one-octave spectral bands which are then converted to effective signal to noise ratios (SNR). These SNR's are weighted and averaged to obtain the STI.

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RASTI
Rapid Speech Transmission Index, an machine method of testing for intelligibility in sound systems that is associated with Brel and Kjaer, the instrumentation company that manufactures a portable device to implement it. RASTI was developed as a simpler alternative to the more complex STI.

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RASTI measures only in two octave bands centered at 500 Hz and 2 kHz, respectively and uses a speech-like excitation signal and correlates reductions in modulation depth to loss of intelligibility. RASTI tests in only two frequency bands, which gives an overly optimistic picture. Any compression in the system may cause an artificially low RASTI value. RASTI also does not take system distortion or non-linear amplitude and phase into account.
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Speech Intelligibility Index

Identical to Speech Transmission Index . Speech Intelligibility Index (SII) is the method for by machine measuring speech intelligibility that is currently proposed in draft form as ANSI Standard S3.5-1997. It is the most reliable and accurate of the machine methods.

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In the Standard, four measurement procedures are allowed, each using a different number and size of frequency bands. In descending order of accuracy, they are: Critical band (21 bands) One-third octave band (18 bands) Equally-contributing critical band (17 bands) Octave band (6 bands) The value of SII varies from 0 (completely unintelligible) to 1 (perfect intelligibility). SII includes reverberation, noise and distortion, which are accounted for in the modulation transfer function.
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Late-arriving reflections and echoes may distort the measurement significantly. SII is also giving artificially low intelligibility scores if compression or limiting is introduced in the system. It ignores frequencies below 100 Hz, may miss significant low-frequency masking sources. SII does not take non-linear phase into account. SII does not take non-linear phase into account.
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Sound Absorption & Intelligibility.

Cathedrals & Churches are having long reverberation time (referred as live) whereas offices etc. are having short reverberation time (referred as dead). Sound absorption is the reduction of sound energy and the coefficient refers the relationship between sound absorption and reflection where 0 stands for no absorption 1 indicates no reflection.
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Increase of absorption reduces the noise level inside the room, 3 10 dB. An increase of the absorption of the ceiling from 20% to 40% or from 40% to 80% improves the audibility double. High absorption in a room is not good as the sound may not be good enough to listen.

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Voice can be heard clearly only when there is a correct balance between sound, absorption and reverberation time. Reflective hard concrete surface increases the reflection, which in turn increase the reverberation time. High reverberation time masked the original voice.

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Original waveform of a speech.

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Reverberation time 0.8 sec.

Here the reverberant sound is stretching out between the syllables to fill in the gaps with noise . The consonants are still distinct and have not been masked by the reverberant sound field.

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A Room With 0.8 Second Reverb Time

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Reverberation time 1.3 sec.

Some of the syllables are being buried or masked by the reverberant "noise".

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A Room With 1.3 Second Reverb Time

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Reverberation time 2.0 sec.

The distinct syllables and speech sounds are just swamped by the persistent sound field slowly decaying behind it.

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A Room With 2 Second Reverb Time

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Therefore it is required to be closer to the sound source in a long decay time reverberant field.

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