1. Essentials of AV Technology Audio Systems

2. Part One Hearing and Sound

3. Hearing and Sound Introduction
Sound Waves
Wavelength
Frequency
Octaves and Bands
Harmonics

4. Hearing and Sound The mechanics of human hearing Outer ear Ear canal
Ear drum
Middle ear
Cochlea
To understand sound, you should understand how your hearing works. Your ears are complex organs that have several distinct parts: Outer ear, ear canal, eardrum, middle ear, and cochlea. To receive and process sound, these parts convert sound waves into mechanical movement and then to electrical signals that your brain translates into usable information. Damage to any of these parts can reduce your ability to perceive sounds.

5. Sound Waves Created by two events Compression Rarefaction
Vibration is a way to produce sound.
A stringed instrument, like a violin or guitar, is an excellent example of this concept. When you pluck a string on a violin, the motion of the string displaces air molecules. As those molecules move, they push against, or “compress,” the air molecules next to them.
As the string reverses its direction, it pulls at the molecules that surround it. Because air is an elastic medium, the displaced molecules transmit this back-and-forth motion to the molecules surrounding them. This displacement moves the sound as a waveform out and away from the source that generated it.
The pushing together of molecules is called compression, and the pulling apart is called rarefaction. These are areas of high and low pressure in the elastic medium. In AV, you are usually concerned with sound moving in air, but there may be occasions that you are interested in its movement through other elastic mediums.
A visual example of the principle of sound waves is a rock thrown into another elastic medium, water. The rock creates a disturbance (displacement of water molecules) in the medium. Water molecules are pushed together in compression and pulled apart in rarefaction, creating the peaks and valleys, or ripples, we see on the surface of the water. We also see that these waves of energy move away from the source in concentric circles as the energy is transferred to nearby molecules.

6. Wavelength
Wavelength: Distance between two points that occur at the sample place.
Sound waves have a physical length in addition to having a particular loudness or intensity. This length is called wavelength. Wavelength is the physical distance between two points exactly one cycle apart in a waveform. Wavelength measures the distance between two points that occur at the same place.
To understand wavelength, pick any point on a wave. Now move along the wave and find the next occurrence of that exact spot. That is a wavelength. This wave is one simple, single wave, or sine wave. The middle horizontal line, known as the zero reference line or reference level, represents the molecules at their rest position.
Parts of the sine wave above the reference level represent the molecules being compressed (compression). Maximum compression occurs at the top of the waveform. Parts below the reference level represent the molecules being rarified (rarefaction). Maximum rarefaction occurs at the bottom of the waveform. One cycle is completed when the molecules have moved from the rest position through compression, back to their rest state, then through rarefaction and finally back to being the same distance from one another as they were at the beginning.

7. Frequency Frequency: Number of cycles completed in one second.
The number of times this cycle occurs per second - from rest position through maximum compression, back to rest position, maximum rarefaction and finally again at the rest position – is called frequency. If the complete cycle occurs 1,000 times per second, the sine wave has a frequency of 1,000 cycles per second (cps). The standard term used to describe frequency (number of cycles per second) is Hertz (Hz). Therefore, we usually talk about a sound wave that has 1,000 cps as having a frequency of 1,000 Hz. While the sounds you hear every day are complex waveforms, each sound can be broken down into individual sine waves of all the frequencies that make up each particular sound.
The length of a sound wave depends on the speed of sound in the medium through which the wave is propagating, and the frequency of the sound. The wavelengths of the range of frequencies most humans can hear (20 Hz to 20 kHz) range from 56.5 ft to less than 3/4 in (17.2 m to 17.2 mm).
Frequency and wavelength are inversely proportional. This means that as one gets larger the other gets smaller. The lowest frequencies have the longest wavelengths and the highest frequencies have the shortest wavelengths.

8. Octaves and Bands
Octave: Interval between a frequency and the doubling of that frequency
To study and measure sound, the frequencies we can hear (20Hz to 20,000Hz) are divided into groups. The most common example of this division is in music.
On a musical staff, each space and line indicates a musical letter A through G. The A above middle C has a fundamental frequency of 440Hz. If you start at A and count eight notes up or down the staff, you will find another A. This interval is called an octave. Each A has a similar sound, but a different frequency.
The human ear's response to frequency is logarithmic. (Think 2, 4, 8, 16) This means that the way we hear is exponential. A tone at 220Hz sounds similar to a tone at 440Hz and 880Hz.
220 * 2 = 440
440 * 2 = 880
Even though there is more frequency room between Hz than Hz, our ear hears the same interval between each sound.
Frequencies can also be divided into bands. Human hearing covers frequencies from 20Hz to 20,000Hz. This spectrum of hearing is divided into 10 bands, where each band represents one octave. Like octaves, there are not equal numbers of frequencies in each band because they are divided logarithmically. Each band is identified by its center frequency.
The spectrum of hearing can be divided into even smaller bands. In this industry, you may encounter 1 octave, 1/3 octave, or 1/10 octave bands.

9. Harmonics Harmonics: Whole number multiple of a fundamental frequency.
Complex waveforms: Comprised of a fundamental frequency plus many harmonics.
The sounds you hear are complex waveforms that can be broken down into individual sine waves. A complex waveform is made up of a fundamental frequency plus whole number multiples of that fundamental frequency. The whole number multiples are called harmonics.
For example, let’s look at a fundamental frequency of 4,000 Hz. Whole number multiples of 4,000 are 8,000; 12,000; 16,000; etc. A complex waveform with a fundamental frequency of 4,000 Hz would also have varying levels of harmonic energy at 8,000 Hz; 12,000 Hz; 16,000 Hz; 20,000 Hz; and so forth.

10. Human Perception of Sound Introduction
Logarithms
Decibels
Decibels Equations
Using the Decibel
Inverse Square Law and Sound

11. Logarithms
Number of times the number 10 must be multiplied by itself to get a desired value
Logarithmic scales make ratios easier to express
We perceive our world in a logarithmic way
A logarithm of a number is how many times the number 10 must be multiplied by itself to get a certain value. For example, the log of 10,000 is 4 and the log of is -4.
Logarithmic scales make the ratio values easier to express. For example, the ratio between the threshold of hearing (when sounds become audible) and the threshold of pain is 1 to 1,000,000. No one wants to count that many zeros.
Think of a standard ruler. Each unit on the ruler represents a unit of one (whether its an inch, millimeter, etc.) wherever it is located on the ruler. Its a one-to-one relationship between the units shown and the units represented. This is a linear scale.
What if the value of each unit on that ruler represented something other than a single unit of one? What if each time you moved to the right, each unit represented ten times as much as before? Or, each time you moved one unit to the left it represented one-tenth as much as before? In this case, comparing adjacent units on a scale would represent a ratio of 1:10 or 10:1. This is a logarithmic scale.
Humans perceive differences in sound levels logarithmically, not linearly. Because of this, a base 10 logarithmic scale is used to measure, record and discuss sound level differences. If you used a linear scale to describe the perceived difference in sound pressure level from the threshold of hearing to the threshold of pain, you would need to use numbers from 1 to well over 3 million. Not only is our perception of sound intensity logarithmic, but so are our other senses. The psychophysical law (or Fechner-Weber law) describes this generalization in psychology. It states that the intensity of a sensation is proportional to the logarithm of the intensity of the stimulus causing it. So our sensations such as sight, hearing or touch vary as the logarithm of the stimulus.

12. Decibels Describes ratios with a wide range of values
Quantifies relationship between two numbers
In AV used for power, distance, voltage, and sound pressure
The basic unit, the Bel, was named after telecommunications pioneer Alexander Graham Bell. As a unit of measurement, the Bel was too large for practical purposes, so the deci-bel (one tenth of a Bel), or dB, is used instead. It is a logarithmic scale used to describe ratios with a very large range of values. The decibel is a unit of measurement used to describe a base 10 or base 20 logarithmic relationship of a power ratio between two numbers. Decibels are also used for quantifying differences in voltage, distance and sound pressure as they relate to power. When you are quantifying differences, the numbers being compared to one another must be of the same type. For example, you could compare one voltage to another voltage. You cannot compare a voltage to a wattage. You could also compare a number or measurement to a known reference level. The amount of increase or decrease from the chosen reference level is what the decibel system measures. This is done with sound pressure levels, where you compare a sound pressure level measurement to the threshold of hearing reference of 0 dBSPL. And remember, since it's a logarithm, whether the increase is from 1 to 100 of whatever you're measuring, or 100 to 10,000, for example, the increase in both cases (in base 10) is still 20 decibels (dB).

13. Decibels Equations Power: dB = 10 * log (P1 / P2)
Voltage: dB = 20 * log (V1 / V2)
Distance: dB = 20 * log (D1 / D2)
We can state the difference in decibels for two powers or voltages or distances using the following equations: 10 times the logarithm of the ratio (of the two numbers we are comparing) describes the difference in decibels if we are comparing two power values. For example the equation: dB = 10 * log (P1 / P2) would give us the difference in decibels if we compared one power (P1) in watts against another power in watts (P2). 20 times the logarithm of the ratio describes the difference in decibels if we are comparing two voltages or distances. For example the equations: dB = 20 * log (V1 / V2) would give us the difference in decibels if we compared one voltage (V1) against another voltage (V2). dB = 20 * log (D1 / D2) would give us the difference in decibels if we compared the sound pressure level at one distance from the source (D1) as compared to the sound pressure level at some other distance from the source (D2). Now you know that you perceive differences in sound levels logarithmically, and that you use the decibel for a logarithmic scale, lets explore further how the decibel is used.

14. Using the Decibel Common References:
The 20log Equation can compare the difference in decibels between two sound pressure levels. It can also compare a sound pressure level against a known sound pressure reference.
Decibels are comparisons, not units of absolute quantity. For a measured SPL to be meaningful, it has to have a reference. It would be like saying, "it is ten degrees hotter today than it was yesterday." Without knowing yesterdays temperature, you cannot know today’s temperature.
In the case of sound pressure, the reference for 0 dB is Pascals (or dyne/cm2). This is the threshold of hearing. Therefore, an 85 dB SPL measurement is 85 decibels of sound pressure, referenced to the threshold of hearing. We can write it as 85 dBSPL, where dBSPL is dB referenced to the threshold of hearing.
These are some accepted generalities in relation to human hearing:
A one-decibel change is the smallest perceptible change noticeable. Unless listening very carefully, most people will not discern a one decibel change.
A just noticeable change, either louder or softer, requires a three-decibel change (e.g., 85 dB SPL to 88 dB SPL).
A ten-decibel change is required for us to subjectively perceive either a change as twice as loud as before or one-half as loud as before. (For example, a change from 85 dB SPL to 95 dB SPL is perceived to be twice as loud as before).

15. Inverse Square Law and Sound
Energy inversely proportional to square of distance from source
6 dB reduction is a doubling of distance
6 dB gain is a halving of distance
Regardless of whether the energy is propagating horizontally or spherically, you can follow the spread of the energy by using the inverse square law. The inverse square law states that sound energy is inversely proportional to the square of the distance from the source. Every time the distance doubles from the source of the energy, the energy spreads out and covers four times the area it did before.
Twenty times the logarithm of the ratio describes the difference in decibels if we are comparing two voltages or distances.
For example, if you are 5 meters from an energy source and you double your distance from the source to 10 meters, the energy must now cover an area four times larger than it did at the 5-meter distance. Every time you double the distance from the source, you increase the surface area of the sphere four times. Subsequently, the energy unit per unit of area is one-quarter of what it was previously.
Sound pressure is reduced by 6 dB every time the distance from the source is doubled. Conversely, the sound pressure increases by 6 dB when the distance from the source is reduced by one-half. This is often referred to as the 6 dB per doubling rule.
In practice, the 6 dB per doubling rule occurs only in a near-field or free field environment. These terms describe the space around a sound source, but they do not include the energy that is reflected back by boundaries such as walls, ceilings, and floors.

16. Part Two Acoustics

17. Acoustics Acoustics Sound Energy Reflected Sound Energy Reverberation
Absorption
Transmission
Ambient Noise
Acoustics is a branch of science that is focused on the qualities and characteristics of sound waves. As you can see from this definition, the study of acoustics covers many topics. It is more than simply hanging some fabrics on a wall to solve an acoustical problem. The study of acoustics includes:
How sound is generated
How sound energy moves through air and other media (like concrete, steel or water)
How dimensions and shapes affect the way sound behaves in an environment
How sound energy can be prevented from leaving or entering a space through partitions or vibrations
What happens to sound energy when it encounters a boundary (materials)
Your perception of a sound as processed by your ear and brain.
Sound can be generated by vibrations (for example, vibrations from a stringed instrument). Many things besides strings and loudspeakers can vibrate and generate sound. Sounds can be generated by the mechanical motion of a loudspeaker, the structural elements of a building or a multitude of other ways.

18. Sound Energy Reflection: Energy sent back into a room
Absorption: Energy absorbed into a medium
Transmission: Energy passes through a medium
When the sound energy encounters a surface or room boundary, these three things occur:
Reflected As the sound energy moves away from the source, some of it will be reflected off of various surfaces back into the room. The reflections can either be in a specular (direct) fashion or diffused (scattered). Either way, the energy remains in the space.
Absorbed This is sound energy that is absorbed either in the air (not much of an issue except in extremely large spaces) or absorbed by the materials in the space (sound energy converted into heat).
Transmitted This is sound energy that actually passes from one space into another through a partition or other barrier.

19. Reflected Sound Energy
Types of reflection
Direct (specular)
Scattered (diffuse)
Echo: Delays due to time and distance
A diffuse reflection scatters the energy back equally in all directions. This is similar to the properties of a diffusion-type projection screen.
Dealing with varying wavelengths, the transition from specular to diffuse is not immediate, and neither is how much is reflected, absorbed or transmitted. Some wavelengths will react one way or another and some will react in between. The behavior will be determined by the size, material and mass of the boundaries of the space. In every situation involving boundaries or surfaces, you will always have the direct sound the sound that arrives at the listener position in a direct, straight line from the source to the listener, and the reflected sound the sound that takes any indirect path from the source to the listener. Reflected energy arrives later in time than direct sound. This should be obvious the shortest path between two points is a straight (direct) line. Taking any other pathway requires traveling a longer distance and requires more time. Thus, reflected sound will always arrive later in time than the direct sound. Reflections can be defined as either early or late. A late reflection is called an echo.

20. Numerous persistent reflections Live environment High level energy
Reverberation
Numerous persistent reflections
Live environment
High level energy
Multiple reflections
When a room has many hard reflective surfaces, the combined energy level of the reflections can remain quite high. As the energy reflects off of more and more surfaces around the room, the listener will begin to receive reflected energy from all directions. When the energy level remains high and the number of reflections becomes quite dense in relation to one another, it is called reverberation. Reverberation is simply numerous, persistent reflections.

21. Absorption Porous Materials Carpets Acoustic tiles Curtains Clothing
One type of absorber you may be familiar with is the porous absorber – the “acoustical fuzz” that is often referred to. As the displaced air molecules pass through a porous absorber, the friction between the molecules and the material of the absorber slows the molecules down.
Typical porous absorbers include carpets, acoustic tiles, acoustical foams, curtains, upholstered furniture, people, and their clothing.

22. Energy passing through surfaces
Transmission
Energy passing through surfaces
Walls
Floors
The sound energy that is not reflected back or absorbed will be transmitted into another space through partitions (walls, windows, floors and ceilings) or structure borne vibrations. The ability of a partition to transmit or reflect sound energy is limited by such factors as weight of the partition (mass), stiffness of the partition, air gaps, design of the partition, and the materials used in construction of the partition. This ability of a partition to transmit sound energy will vary with frequency.

23. Ambient Noise Any sound other than the desired signal Air conditioning
Equipment fans
Machines
Sound through windows
Ambient noise is any sound other than the desired signal. While an electronic sound system has inherent noise in the electronic components, rooms also have noise associated with them. Anything heard in a room other than the desired signal from the sound reinforcement system would be considered noise. Unwanted background noise found within a room can come from equipment fans, office machines, the HVAC (heating, ventilation and air conditioning) system or noise from the people in the room. Noise can intrude from outside the room as well, through partitions or windows. Outside sources can include vehicular traffic, adjoining corridors and structure borne vibrations. Since excessive noise levels interfere with the message being communicated, ideally, background noise-level limits will be specified by an acoustician, audiovisual consultant or designer appropriate to the type of room and its designed purpose. In other words, the criteria and limits for background noise levels for a gymnasium will be much different than those of a conference room. The HVAC system, partitions and any necessary acoustical treatment will be designed and applied so that the background noise level criteria is not exceeded. A rooms acoustical properties (i.e., reflections and types, amount of transmission allowed) and background noise levels are very significant contributors to a sound systems overall effectiveness.

24. Part Three Capturing Sound

25. Microphone Types Introduction
Audio Signal Pathway
Dynamic Microphone
Condenser Microphone
Phantom Power
Electret Microphones
Microphone Physical Design and Placement

26. Audio Signal Pathway Energy: Acoustic to electrical to acoustic
In its most basic form, the complete audio signal path takes acoustic energy and converts it into electrical energy so it can be routed, processed, further amplified, and converted back into acoustical energy.
This conversion of energy from one form to another is called transduction. A microphone is a transducer it converts acoustic energy into electrical energy. A loudspeaker is also a transducer it converts the electrical energy into acoustic energy. This means you can have transducers on either end of an electrical audio path.

27. Dynamic Microphone Response of diaphragm to pressure
Movement induces voltage
No power source
Summarize the basic workings of a dynamic microphone and identify two dynamic microphone characteristics.
In a dynamic microphone, you will find a coil of wire called a conductor attached to a diaphragm and placed in a permanent magnetic field. Sound pressure waves cause the diaphragm to move back and forth, thus moving the coil of wire attached to it. As the diaphragm and coil assembly moves, it cuts across the magnetic lines of flux of the magnetic field, inducing a voltage onto the coil of wire. The voltage induced into the coil is proportional to the sound pressure and produces an electrical audio signal. The strength of this signal is very small and is called a mic level signal.
Dynamic microphones are used in many situations because they are economical, durable and will handle high sound pressure levels. Dynamic microphones are different from other microphones and very versatile because they do not require a power source.

28. Condenser Microphone Capacitor: Diaphragm and fixed back plate
Power Source
Electret Microphone
Size
Battery Option
Condenser microphones have a conductive diaphragm and a conductive backplate. Air is used as the insulator to separate the diaphragm and backplate. Voltage from a power supply, known as phantom power, is used to polarize, or apply, the positive and negative charges to create the electric field between the diaphragm and backplate. Sound pressure waves cause the diaphragm to move back and forth, subsequently changing the distance (spacing) between the diaphragm and backplate. As the distance changes, the amount of charge, or capacitance, stored between the diaphragm and backplate changes. This change in capacitance produces an electrical audio signal. The strength of the signal from a condenser microphone is not as strong as the microphone level signal we see from the typical dynamic microphone. To increase the signal, a condenser microphone includes a preamplifier, powered by the same phantom power supply used to charge the plates in the microphone. This preamplifier amplifies the signal in the condenser microphone to a microphone level signal, but is not to be confused with the microphone preamplifier that is found in a mixing console.

29. Phantom Power Remote Power Source Mixer Outboard Supply
volts DC
Phantom power is the remote power required to power a condenser microphone. It typically ranges from 12 to 48 VDC (volts DC).
Phantom power is most often available from an audio mixer. It may be switched on or off at each individual microphone input, or from a single button on the audio mixer that makes phantom power available on all the microphone inputs at once.
If phantom power is not available from the audio mixer, separate phantom power supplies may be used.

30. Electret Microphones Type of condenser mic
Named after prepolarized material applied to the diaphram or backplate
Requires less voltage than a typical condenser
Can be very small

31. Microphone Physical Design and Placement
Surface Mount
Shotgun
Handheld
Gooseneck
Lavalier
Handheld microphones are used mainly for speech or singing. Since it is constantly moved about, a handheld microphone includes internal shock mounting to reduce handling noise.
Surface microphones (also called "boundary microphones") are designed to be mounted directly against a hard surface, such as a conference table, wall, or ceiling. The acoustically reflective properties of the mounting surface affect the mics performance.
Mounting a microphone on the ceiling typically yields the poorest performance because the sound source is much farther away from the intended source (eg., conference participants) and much closer to other noise sources such as ceiling mounted projectors, HVAC diffusers, and other devices.

32. Microphone Specifications Introduction
Microphone Polar Patterns
Microphone Sensitivity
Microphone Frequency Response
Microphone Impedance

33. Microphone Polar Patterns
Omnidirectional
Cardioid
Supercardioid
Bi-directional
Hypercardioid: Variant of cardioid.
Directional, rejects sound from sides.
One of the characteristics to look for when selecting a microphone for a particular use is its pickup pattern. The pickup pattern describes the microphones directional capabilities or, in other words, the microphones ability to pickup the wanted sound in a certain direction while rejecting unwanted sounds from other directions.
Pickup patterns are defined by the directions from which the microphone is optimally sensitive. These pickup patterns help you determine which microphone type you should use for a given purpose.
There will be occasions when you want a microphone to pick up sound from all directions (like an interview) and there will be occasions that you do not want a microphone to pick up sounds from sources surrounding it (like people talking or someone rustling papers). The pickup pattern is also known as the polar pattern or microphones directionality.
Omnidirectional Sound pickup is uniform in all directions.
Cardioid (Unidirectional) Pickup is from the front of the microphone only (one direction) in a cardioid pattern. It rejects sounds coming from the side but the most rejection is at the rear of the microphone. The term cardioid refers to the heart shaped polar plot.
Supercardioid Provides better directionality than the hypercardioid. Less rear pickup than the hypercardioid.
Bi-directional Pickup is equal in opposite directions with little or no pickup from the sides. This is sometimes also referred to as a figure -eight pattern, referring to the shape of its polar plot.
Hypercardioid  A variant of the cardioid. More directional than the regular cardioid because it rejects more sound from the side. The tradeoff is that some sound will be picked up directly at the rear of the microphone.

34. Microphone Sensitivity
Output level referenced to input level
Condensers vs. dynamics
One performance criteria that characterizes a microphone is its sensitivity specification. This defines its electrical output signal level given a reference sound input level. Put another way, sensitivity defines how efficiently a microphone transduces (converts) acoustic energy into electrical energy. If you expose two different types of microphones to an identical sound input level, a more sensitive microphone provides a higher electrical output than a less sensitive microphone. Condenser microphones are usually more sensitive than dynamic microphones. Does this mean that lower sensitivity microphones equates to lesser quality? Not at all. Microphones are designed and chosen for specific uses. A professional singer for example, using the microphone up close can produce a very high sound pressure level. In contrast, a presenter speaking behind a lectern and a foot or two away from the microphone would benefit from a microphone with higher sensitivity.
For the singer, a dynamic microphone may be the best choice, as it will typically handle the higher sound pressure levels produced by the singer without distortion while still providing more than adequate electrical output. The presenter, using a microphone which is farther away than the singers, produces a much lower sound pressure level. The presenter would certainly benefit from using a more sensitive microphone.

35. Microphone Frequency Response
Microphone Frequency Response: The range of frequencies a microphone can transduce.
NOTE: the freq response spec on the slide is from a lectrosonics MM400C waterproof lav.
Frequency response specification is an important measure of a microphone's performance. This defines the microphones electrical output level over the audible frequency spectrum, which in turn helps to determine how an individual microphone sounds.
A microphone’s frequency response gives the range of frequencies, from lowest to highest, that the microphone can transduce. The microphone's directional and frequency response characteristics are represented graphically through a polar plot.
The polar plot shows the directional and frequency response characteristics on a two-dimensional graph of electrical output vs. frequency.

36. Microphone Impedance Low impedance ( <200 ohms)
High impedance ( >25k ohms)
Another microphone specification that must be considered is its output impedance. Impedance is the opposition to the flow of electrons in an AC circuit. Your audio signals are AC circuits.
Microphones can fall into two categories based upon output impedance:
Low impedance 200 ohms or less (some as high as 600 ohms).
High impedance More than 25,000 ohms.
Professional microphones are low impedance microphones. Low impedance microphones are less susceptible to noise and allow for much longer cable runs than high impedance microphones.
200 ohm mic level output
2000 ohm mic level input

37. Microphone Signal Transport Introduction
Wireless Microphones
Microphone Cables and Connectors

38. Radio frequency transmission Hands free
Wireless Microphones
Radio frequency transmission
Hands free
Sometimes called radio mics, wireless microphones use RF (radio frequency) transmission in place of a microphone cable. Some wireless systems use IR (infrared) transmission. For a handheld microphone, a standard microphone casing is often integrated onto the top of a transmitter and the microphone casing and transmitter are finished as one unit. At other times, a small plug-on style transmitter is attached to the bottom of a regular handheld microphone.
For hands free applications, a lavalier or headmic microphone is plugged into a bodypack style transmitter. The bodypack transmitter is then clipped onto a belt or placed in a pocket or pouch. Either way, at the other end of the RF or IR transmission is a receiver tuned to the transmitters specific frequency. Modern RF wireless microphones allow you to change frequencies in order to avoid interference from outside sources as well as interference from other wireless microphones that may be in use. This is called frequency coordination and will be specific to your geographical area. Your wireless microphone manufacturer can provide help in coordinating compatible frequencies for your area.

39. Microphone Cables and Connectors
Shielded twisted pair cable
XLR male and XLR female
Microphones and wireless microphone receivers are connected to audio mixers with a cable and a connector. Professional microphone cables utilize shielded twisted pair cable. Shielded twisted pair contains:
Two small gauge insulated copper wires (conductors) twisted together
An aluminum foil or a copper braided shield that covers the twisted conductors
The twisted pair and shield are covered with a protective jacket (rubber or plastic)
Typically, the ends of the shielded twisted pair cable are finished (terminated) with an XLR connector.
XLR is now a generic term describing the common audio connector. Although they are available in 3- to 7-pin configurations, the 3-pin XLR is used for almost all microphone cable applications.

40. Part Four Audio Signal Levels

41. Audio Signal Levels Introduction
Signal Level Compatibility
Signal Level Adjustments

42. Audio Signal Levels Microphone preamplifier boosts a mic level signal.
Description
Voltage Level
Mic Level
0.001 volts
Line Level (Professional)
1 volt
Line Level (consumer)
0.316 volts
Loudspeaker Level
2 < 100 volts
Microphone preamplifier boosts a mic level signal.
A microphone, regardless of the type, produces a signal level that is called mic level. Mic level is a very low level signal that’s only a few millivolts (abbreviated as mV to express one thousandth of a volt).
Mic level is a very low level signal and subject to interference. Therefore you need to amplify this signal by using a microphone preamplifier. A microphone preamplifier, often known as a preamp for short, takes the mic level signal and amplifies it to what is known as line level.
Line level is where all signal routing and processing is performed. Line level in a professional audio system is about 1 volt.
Consumer line level is volts (316 mV), which is less than line level in a professional device. Consumer line level often uses an RCA (phono) connector.
Once you have routed and processed the signal, it is sent to the power amplifier for final signal amplification up to loudspeaker level. The loudspeaker takes that amplified electrical signal and transduces the electrical energy into acoustical energy.

43. Signal Level Compatibility
Inputs and signal level
Microphone input, mic level signal
Line level input, line level signal
Powered loudspeaker
Operating manual
As you begin to connect your system, you need to make sure the components of your system are compatible with one another. For example, while you could plug a microphone directly into the input of a power amplifier, you would not get much sound level. The mic level signal isn’t strong enough by itself. You also would not want to connect the output of a power amplifier into a device expecting either a mic or line level – you would almost certainly damage components. What about plugging your microphone directly into the back of a powered loudspeaker? Some companies manufacture “powered loudspeakers” that are all-in-one devices meant to simplify setup and provide for easy portability. In the case of a powered loudspeaker, all of the signal requirements listed above are built in to the loudspeaker. If the powered loudspeaker has microphone inputs, it has a microphone preamplifier and any internal processing will be done at line level. It will also have the power amplifier built in to power the loudspeaker.
Read the operating manual for the device to verify correct signal levels.

44. Signal Level Adjustments
Signal adjustments = Amplitude adjustments
Unity (no change)
Gain (increase)
Attenuation (decrease)
Adjustments you make to signal levels are called gain adjustments. A gain control is the term used to describe the general ability to make adjustments to the signal levels. If the signal level is increased, it is called gain and it refers to the amount of amplification applied to the signal. If the signal level is decreased, it is called attenuation. If neither gain nor attenuation are applied, it is called unity gain. Unity gain means that the signal is passing through the gain control without any changes to the signal level.

45. Part Five Audio Components

46. Audio Components Introduction
Audio Mixers
Audio Processors: Compressions, Limiters, and Expanders
Audio Processors: Gates and Filters
Equalizers
Delays
Power Amplifiers

47. Audio Mixers Multiple inputs to one or more outputs
Identifying mixer configurations
All audio mixers serve the same purpose to combine, control and route audio signals from a number of inputs to a number of outputs. Usually, the number of inputs will be larger than the number of outputs.
Audio mixers are often identified by the number of available inputs and outputs. For example, an 8x2 mixer would have eight inputs and two outputs.
Each incoming mic or line level signal goes into its own channel. Many mixers provide individual channel equalization adjustments as well as multiple signal routing capabilities called main or auxiliary busses. A larger audio mixer is often called a mixing console, console or a mixing desk. Regardless of the size and complexity, any mixer that accepts mic level inputs will have microphone preamplifiers (preamps). Once the mic level is amplified to line level by the preamp, it can be sent through to the rest of the mixer. Between the inputs and outputs, the typical audio mixer provides multiple gain stages for making adjustments. These adjustments allow the mixing console operator to balance or blend the audio sources together for the most realistic sound appropriate for the listening audience.
Some audio mixers will turn microphone channels either on and off automatically like an on/off switch. These are called gated automatic mixers. Others will turn up microphone channels being used and turn down microphone channels that are not being used, like a volume knob being turned up or down. These are called gain sharing automatic mixers.

48. Audio Processors: Compressions, Limiters, and Expanders
Processors: Control dynamic range with defined thresholds
Compressor: keeps loud signals from being too loud
Limiter: creates a ceiling to prevent signal spikes from damaging equipment
Expander: Reduced unwanted background noise
Many types of processors can refine an audio signal. The intended use and listening environment will determine which type is right for you. Some common processors include limiters, compressors, expanders, gates, and filters.
Compressors:
Reduce the level of all signals above an adjustable threshold. In other words, they keep loud signals from being too loud.
The amount of reduction above the threshold is determined by an adjustable ratio.
The reduction reduces the variation between highest and lowest signal levels resulting in a compressed (smaller) dynamic range.
Can be used to prevent signal distortion.
Extreme compression is called limiting.
Limiters:
Limit the level of all signals above an adjustable threshold. In other words, they prevent high amplitude signals from getting through.
Limiting is used to prevent damage to components such as loudspeakers.
Triggered by peaks or spikes in the audio signal (like a dropped microphone), and reacts quickly to cut them off before they exceed a certain point.
The amount of limiting above the threshold is determined by a more aggressive ratio than a compressor reduction ratio.
The reduction limits the variation between highest and lowest signal levels resulting in a limited dynamic range.
Expanders
More properly called downward expanders.
Reduce the level of all signals below an adjustable threshold.
The amount of reduction below the threshold is determined by an adjustable ratio.
The signal level reduction increases the variation between highest and lowest signal levels resulting in an increased dynamic range.
Used for reducing unwanted background noise.

49. Audio Processors: Impact on a Signal
Audio Compressor: Impact on Signal
Audio Limiter: Impact on Signal

50. Audio Processors: Gates and Filters
Description
Gate
Mutes all signals below an adjustable threshold.
Filter
Regulates the passing of frequencies from a signal.
Gates:
Are an extreme downward expander.
Mute the level of all signals below an adjustable threshold.
Signal levels must exceed the threshold setting before they are allowed to pass.
Can be used to automatically turn off unused microphones.
Filters
Filter, remove, or pass certain frequencies from a signal.
Notch filter- notches out" a specific frequency of band of frequencies.
Low/High filters - Low pass filters pass the low frequency content of a signal while high pass filters pass the high frequency content.

51. Equalizers Frequency response management Graphic 1/3 Octave Parametric
Complex frequency management
Equalizers, or EQs, are frequency controls which allow you to boost (add gain) or cut (attenuate) a specific range of frequencies. The simplest equalizer comes in the form of the bass and treble tone controls found normally on your home stereo or surround receiver. The equalizer found on the input channel of a basic audio mixer may provide simple high, mid and low frequency controls. Going beyond the home stereo and basic input channel equalizers, you will find two common types of sound system equalizers graphic equalizers and parametric equalizers.
A common graphic equalizer is the 1/3 octave equalizer. The 1/3 octave graphic equalizer provides 30 or 31 slider adjustments corresponding to specific fixed frequencies with fixed bandwidths, with the frequencies centered at every one-third of an octave. The numerous adjustment points allow for shaping the overall frequency response of the system so that the sound system sounds more natural. The graphic equalizer is so named because the adjustments provide a rough visual, or graphic, representation of the frequency response adjustments.
A parametric equalizer offers greater flexibility than a graphic equalizer. Not only will the parametric provide boost or cut capability like the graphic, but it also allows center frequency and bandwidth adjustments.

52. Delays Delay: Compensation for distance and location
Electronic delay is used many times in sound reinforcement applications. For example, consider an auditorium with an under-balcony area. The audience seated directly underneath the balcony may not be covered very well by the main loudspeakers. In this case, supplemental loudspeakers are installed to cover the portion of the auditorium underneath the balcony. While the electronic audio signal arrives at both the main and under-balcony loudspeakers simultaneously, the sound coming from these two separate loudspeaker locations would arrive to the audience underneath the balcony at different times and sound like an echo. This is because sound travels at about 1130 feet per second (344 meters per second), much slower than the speed of the electronic audio signal. In this example, an electronic delay would be used on the audio signal going to the under-balcony loudspeakers. The amount of delay would be set so that both the sound from the main loudspeakers and the under-balcony loudspeakers arrive to the audience at the same time.

53. Power Amplifiers
Amplifiers: boost the signal with enough energy to move the loudspeaker.
The amplifier is the last device used before the signal reaches the loudspeakers. Power amplifiers boost, or amplify, electronic audio signals sufficiently to move the loudspeakers. They do this by increasing the gain (the voltage and power) of the signal from line level to loudspeaker level.
Most amplifiers have only a power switch and input sensitivity controls. Some now include digital signal processing and network monitoring and control.
Power amplifiers are connected to loudspeakers with larger gauge wire that we use at mic or line level. The size of wire will depend on the distance between the power amplifier, the loudspeaker and the current required. Loudspeaker cabling will be unshielded and may or may not be twisted.

54. Part Five Loudspeakers

55. Loudspeakers Intro Loudspeakers Introduction Loudspeakers Crossovers
Loudspeaker Sensitivity
Loudspeaker Frequency Response and Polar Patterns
Loudspeaker Impedance
Center Cluster and Distributed Systems

56. Loudspeakers Introduction
Audio Signal Chain
Electrical energy to acoustic energy
General applications
Communication
Reinforcement
Reproduction
For the purpose of sound reinforcement, loudspeakers are the end of the electrical signal path. The acoustic energy that was transduced into electrical energy by the microphone is transduced back into acoustical energy by the loudspeaker.

57. Crossovers Divide frequencies into specific ranges
Different drivers for different frequencies
Tweeters
Horns
Woofer
Subwoofer
The audio spectrum has wavelengths and frequencies that vary dramatically. No single driver can reproduce the entire frequency range accurately or efficiently. This is why, in professional audio, a loudspeaker contains multiple drivers. A loudspeaker enclosure containing more than one frequency range of drivers is known by the different frequency ranges being covered. So that each driver is sent only those frequencies that it will transduce efficiently, an electrical frequency-dividing network circuit called a crossover is used. A passive crossover (one that doesn’t require powering) would be used to take the electrical signal coming into the loudspeaker enclosure and split it into the different frequency ranges.
Examples of the different drivers and frequency ranges:
Tweeters high frequencies
Horns mid to high frequencies
Cone or midrange midrange frequencies
Woofers low frequencies
Subwoofers lower frequencies

58. Loudspeaker Sensitivity
Loudspeaker Sensitivity Specifications
Loudspeaker Performance Measurement:
88 1m
This loudspeaker will deliver 88 decibels of sound pressure when receiving 1 watt of power measured at a distance of 1 meter.
Loudspeakers vary quite a bit when it comes to efficiency. Does this mean that lower sensitivity loudspeakers are always of lesser quality? Not at all. Like microphones, loudspeakers are designed and chosen to meet specific uses.

59. Loudspeaker Frequency Response and Polar Patterns
Dispersions at different frequencies
Listener position
Polar plots map performance
As with directional microphones, a loudspeaker's overall frequency response will be best on-axis, directly in front of the loudspeaker. As you move off-axis with a loudspeaker, not only will the sound be reduced, but the frequency response will change as well. A loudspeaker with specifications showing a nominal 90 degree by 40-degree dispersion (coverage) pattern only holds that pattern over a limited frequency range. Lower frequencies will spread out in a more omnidirectional fashion due to their much longer wavelengths. Very large devices are required to control the dispersion pattern at the lower frequencies.

60. Loudspeaker Impedance
Matching amplifiers and loudspeakers
Common impedances
4, 8,16 ohms
Knowing a loudspeaker's impedance will help you find the total impedance load when you connect multiple loudspeakers to the output of a power amplifier. This information will help you to avoid wasting power, overloading your power amplifier, and damaging your loudspeakers. It will also give you optimum volume by reducing distortion and noise, and avoiding uneven sound distribution.

61. Center Cluster and Distributed Systems
Focus on presentation area
Distributed systems
Origin of sound not important
Although there are many arrangements for loudspeaker placement, depending on the application and acoustic environment, there are two general approaches to covering the enclosed listener area.
The first is the central cluster, or single source configuration, which can be a single loudspeaker or a central cluster of loudspeakers. The central cluster is normally located directly above, and slightly in front of, the stage or presentation area. In a central cluster, the sound is coming from one point in the room. Since you tend to look in the direction sound is coming from, central cluster systems naturally focus your attention on something, like a pulpit in a house of worship.
If the ceiling height is not adequate for a central cluster, or if there is no need to have a relationship between the location of the sound source and the origin of the sound, a distributed system is often used. Distributed systems use multiple loudspeakers that are strategically suspended overhead or located in the ceiling. A distributed system is often used for voice reinforcement or paging applications.

62. Part Six Audio Signal Level Monitoring

63. Audio Signal Level Monitoring Introduction
Check Signal Levels
Audio Signal Level Monitoring
Signal Levels
Balanced and Unbalanced Circuits
Balanced and Unbalanced Distinctions
Feedback

64. Check Signal Levels Verify signals at all used mixer inputs.
Adjust gain levels wherever appropriate.
Turn on the power amplifier.
Slowly increase sound pressure to desired level.
Listen for distortion and correct as needed.
Before turning on the power amplifier, check to make sure you are getting signals on all the audio mixer channels in use. Make the necessary level adjustments to the microphone preamplifier and all other gain stages in the mixer and the other audio equipment leading to the power amplifier. Checking signal levels in the audio mixer is often done with the aid of the built-in meters and verified with headphones. After setting the gain stages, turn the power amplifier input adjustments all the way down and turn on the power amplifier. Slowly bring the power amplifier input adjustments up until the desired sound pressure level is reached at the loudspeakers. Verify through listening to the system and through signal monitoring that no signals are distorting. These practices will help prevent signal levels that are too low, resulting in poor signal-to-noise, and hiss or signal levels that are too high, resulting in distortion.

65. Audio Signal Level Monitoring
Volume Unit Indicator (VU)
Peak Program Meter (PPM)
Signal levels that are too low decrease the systems signal-to-noise ratio and can result in background hiss. Two types of signal meters are typically used for signal monitoring. They are the standard volume unit indicator known as the VU (Volume Unit) meter and the PPM (Peak Program Meter). The VU meter is commonly found to monitor broadcast signals and the PPM is often found in audio mixers.
A PPM (Peak Program Meter) shows instantaneous peak levels and is very useful for digital recording.

66. Signal Levels Analog Signal Levels: 0 dBu or more
Digital Signal Levels: can never exceed 0 dBu
Signal is distorted if exceeded
With analog, running signal levels at around 0 dBu for line level signals is often preferred and there may be some occasions when the normal signal level exceeds 0 dBu. With digital however, the level must never exceed 0 dBFS. dBFS is the full scale (FS) of the digital signal. Exceeding 0 dBFS with a digital signal causes immediate distortion of the signal.

67. Balanced and Unbalanced Circuits
Combats noise
Unbalanced
Susceptible to noise
One way to reduce the noise in a circuit or cable is to use a balanced electrical design. Electronic circuitry can either be balanced or unbalanced.
In a balanced design, the equipment outputs a balanced signal to the cable which is sent to equipment with a balanced input. The design of balanced circuits offers a defense mechanism against noise. This defense mechanism removes the noise, or most of it, leaving only the intended signal. Obviously then, as a general rule, you should use balanced components whenever you can. The cabling used with balanced circuitry requires two signal conductors. In audio, the two signal conductors are surrounded by a shield.
In an unbalanced circuit design, the equipment outputs an unbalanced signal to the cable that is connected to an unbalanced input. As with a balanced circuit, the cable picks up noise from surrounding sources. However, with an unbalanced circuit design, there are no noise defense mechanisms. If noise gets onto the signal conductor, it is there to stay. Cabling used with unbalanced circuitry uses a single signal conductor. The single conductor is surrounded by a cable shield that also acts as the return electrical path for the circuit.

68. Balanced and Unbalanced Distinctions
Number of conductors and circuit type
Unbalanced circuits are also called single-ended circuits. They require two conductors in the cable and connector. The first conductor, an insulated wire, transports the signal. The second conductor is a shield around the wire used as the electrical circuits return path and provides the ground reference for the circuit.
As with most things that offer higher quality, balanced components are more expensive due to design and manufacturing. Like an unbalanced circuit, they require two conductors in the cable and connector. However, in this case, both conductors are used to transport the signal. The first signal conductor carries a signal and the second conductor carries an inverse or mirror image of the first signal conductor. In this case, the impedances on the two signal conductors, as well as the input and output circuitry connected to them, are the same with respect to one another. Since the impedances are the same for the two signal conductors they are said to be equal or balanced.
Although a balanced circuit only requires two conductors, with audio, we also use a foil or braided shield wrapped around the two signal conductors that serves as the circuits ground reference. An audio balanced circuit has a connector that requires three pins two for the signal conductors and a third for the shield connection.
The noise however, is now 180 degrees out-of-phase (one is negative when the other is positive) and this cancels out the noise. So, a balanced circuit provides greater signal strength for longer distances and also has less noise.

69. Feedback Feedback Definition Loudspeaker placement
Feedback is the "squealing" or "howling" generated between microphones and loudspeakers. Feedback occurs if a microphone is either too close to the front of a loudspeaker or the gain (volume) somewhere in the sound system has been turned up too high. A sound system is an amplification system. If the microphone “hears” itself through the sound system, it goes through the signal path where it gets amplified and then it comes out of the loudspeaker at an even louder level than before. That’s why feedback typically gets so loud so quickly – it is a regenerative amplification loop. The signal continues to receive additional amplification each time it goes through the system. One way to avoid feedback is through proper microphone and loudspeaker placement. The best strategy is to place the microphones as close to the sound source and as far from, and behind the loudspeakers as practical.
Best practices for controlling feedback include:
Keeping the microphone as close to the sound source as possible.
Keeping the loudspeakers in front of, and as far from, the microphones as is practical.
Select directional microphones with polar patterns that fit the usage requirements.
Select loudspeakers with coverage patterns that cover only the audience area.

70. Part Seven Audio System Applications

71. Audio System Applications Introduction
Sound Reinforcement
Mix-Minus Systems
Playback and Combination Systems
Intercom and Paging Systems
Audio Conferencing and Videoconferencing
Sound Masking Systems
Audio Systems Summary

72. Sound Reinforcement Sound amplification Music reinforcement
Speech reinforcement
If you cant hear something at an adequate level acoustically (unamplified), microphones, audio mixers, signal processors, power amplifiers, and loudspeakers are used to electronically amplify that sound source so that you can hear it and it can be distributed to a larger or more distant audience. The general term sound reinforcement can be broken down into subcategories: music and speech reinforcement.
You can reinforce a live musical performance. As musical instruments cover a good bit of the audible spectrum, music reinforcement systems tend to be full bandwidth systems, capable of reproducing a wider frequency range with higher sound pressure levels.
If you cant adequately hear an unamplified presenter, a speech reinforcement system will help to amplify and distribute the sound. Since the human voice has limited bandwidth, the speech reinforcement system need not be designed for full bandwidth.

73. Mix-Minus Systems Mix-minus system characteristics
Multiple subsystems (zones)
In the diagram on the slide, "S" is a loudspeaker, "M" is a microphone.
A mix-minus system is a type of speech reinforcement system.
When both meeting presenters and participants need to be heard, microphones must be distributed and mixed with each group of loudspeakers. A live microphone placed near a loudspeaker amplifies its signal and causes feedback, so creating a stable system presents several engineering challenges.
The best approach is to create a separate sound subsystem for each loudspeaker, or group of loudspeakers (called loudspeaker zones). The term "mix-minus" means that each subsystem mixes the microphone signals, minus the microphones closest to a group of loudspeakers. These microphones would cause feedback, and, since they are so close to that group of loudspeakers, they would not require reinforcement anyway.
You would find such systems in large boardrooms, meeting rooms, or classrooms.

74. Playback and Combination Systems
Playback system
Reinforce recorded material
No microphone sources
Combination system
Single system supporting multiple purposes
A playback system is similar to a music reinforcement system in that it has a wider bandwidth, and is capable of higher sound pressure levels. The difference between a music reinforcement system and a playback system is that a playback system does not include microphones. It simply plays prerecorded material.
A single system can be used for music reinforcement, speech reinforcement, and playback. It is much easier to include and design speech reinforcement as part of a music reinforcement system than the other way around.

75. Audio Conferencing and Videoconferencing
Communicate with groups
Videoconferencing
Supporting system
Audio conferencing is a way for groups of people to communicate. It is different from a telephone call in that telephones only allow two people to communicate at a time.
Depending on the size of the conference, a central audio conferencing pod may fulfill audio conferencing needs. This portable tabletop pod typically contains a loudspeaker and several microphones. Larger conferences may require multiple individual microphones, installed as a part of a larger integrated audiovisual system.
Part of the equipment included in audio conferencing will include either line or acoustic echo-cancellation technology. Larger systems may also include speech reinforcement and playback capabilities.  

76. Sound Masking Systems Minimize transmission Create intelligibility
Provide sense of privacy
Noise systems
For most types of sound systems, you will want to minimize background noise levels so that the intended message can be communicated clearly.
Sound masking, or a speech privacy system, is a system that creates background noise like pink noise to reduce clarity and create privacy. It also helps to reduce distractions from other noises. Often, an electronic noise generator is sent to loudspeakers located in the ceiling above the drop tiles.
In a quiet, non-reverberant environment, conversations can be overheard unintentionally, invading privacy. Mechanical or man-made noises can become disruptive to the work effort. When properly installed and balanced, the sound masking system increases privacy and makes these noises less noticeable. Reducing noticeable noise in the workplace can also increase efficiency.
Sound masking systems are commonly applied in medical, government, and industrial facilities.

77. Audio Systems Summary Sound Reinforcement Mix-Minus
Playback and Combination Systems
Intercom and Paging Systems
Audio Conferencing and Audio for Videoconferencing
Sound Masking Systems
Much of what you need to know as an audiovisual professional revolves around sound. In this Audio section, you learned how your hearing works when receiving and processing sound. You read about the basics of sound propagation, sound wave frequency and wavelength, harmonics, the decibel, the sound environment, and how these basics apply to the electrical pathway used to amplify sound. You explored the electrical audio signal chain from start to finish microphones to loudspeakers; and learned about the various signal levels, the cable used, and even the types of circuits (balanced) that are preferred for professional audio. In summary, the complete audio signal path takes acoustic energy and converts it into electrical energy so it can be routed, processed, further amplified, and converted back into acoustical energy.