Technology

Mechanical recording
The first devices for recording sound were mechanical in nature. In 1796 a Swiss watchmaker named Smooth Nikola described his idea for what we now call the cylinder musical box. This can be considered an early method of recording a melody, although it does not record an arbitrary sound and does not record automatically. "Playback" however is automatic.[citation needed] The Player piano was a device that could playback a piano performance which had earlier been mechanically recorded onto a piano roll. The first recording and playback of sound waves used rotating cylinder In 1857, Leon Scott invented the 'phonoautograph', the first device to record arbitrary sound. It used a membrane (which vibrated in response to sound) attached to a pen, which traced a line roughly corresponding to the sound's waveform onto a roll of paper. Although able to record sound, the phonoautograph was unable to immediately play back the recording, although, in one laboratory experiment, a phonoautograph recording was photoengraved onto a metal plate, creating a groove, which was then played back. This photographic methodology, however, was technologically limited, as photography was still in its infancy at this time. Improvements to this invention were required in order to make it commercially viable as a sound recording and reproduction medium.
The recording medium was a piece of smoked paper attached to the surface of a drum which, when rotated, moved forwards along a helical screw. A stylus was attached to a diaphragm through a series of levers, which moved in a lateral direction when the diaphragm was vibrated by a voice. This caused a wavy line to be traced on the smoked paper. A barrel shaped mouthpiece was also included in the design.

The phonograph and the gramophone

Edison's cylinder-based Phonograph The phonograph built expanding on the principles of the phonoautograph. Invented by Thomas Edison in 1877, the phonograph was a device with a cylinder covered with a soft material such as tin foil, lead, or wax on which a stylus drew grooves. The depth of the grooves made by the stylus corresponded to change in air pressure created by the original sound. The recording could be played back by tracing a needle through the groove and amplifying, through mechanical means, the resulting vibrations. A disadvantage of the early phonographs was the difficulty of reproducing the phonograph cylinders in mass production.

The disk-based gramophone 1

This changed with the advent of the gramophone (phonograph in American English), which was patented by Emile Berliner in 1887. The gramophone imprinted grooves on the flat side of a disc rather than the outside of a cylinder. Instead of recording by varying the depth of the groove (vertically), as with the phonograph, the vibration of the recording stylus was across the width of the track ( horizontally). The depth of the groove remained constant. Berliner called this audio disc a "gramophone record", although it was often called a "phonograph record" in U.S. English. Early disc recordings and phonograph cylinders had about the same audio fidelity (despite the cylinder's theoretical advantages of constant linear groove speed and greater dynamic range of the hill-and-dale groove geometry). However, disc records were easier and cheaper to mass produce. From the beginning, the flat disks were easily massproduced by a molding process, pressing a master image on a plate of shellac. Originally, cylinders could only be copied by means of a pantograph mechanism, which was limited to making about twenty-five copiesall of significantly lower quality than the original while simultaneously destroying the original. During a recording session, ten or more machines could be ranged around the talent to record multiple originals. Still, a single performance could produce only a few hundred salable copies, so performers were booked for marathon sessions in which they had to repeat their performances over and over again. By 1902, successful molding processes for cylinder recordings were developed. The speed at which the disks were rotated was eventually standardized at 78 rpm. Later innovations allowed lower rotations: 45 and 33 rpm, and the material was changed to vinyl. Both phonograph cylinders and gramophone discs were played on mechanical devices most commonly hand wound with a clockwork motor. The sound was amplified by a cone that was attached to the diaphragm. The disc record fell into public favor quickly, and cylinders were not produced after 1929. The advent of electrical recording in 1925 drastically improved the quality of the recording process of disc records. Oddly, there was a period of nearly five years, from 1925 to 1930, when the premiere technology for home sound reproduction consisted of a combination of electrically recorded records with the specially-developed Victor Orthophonic phonograph, a spring-wound acoustic phonograph which used waveguide engineering and a folded horn to provide a reasonably flat frequency response. Electrically powered phonographs were introduced c. 1930, but crystal pickups and electronic reproduction did not become common until the late 1930s.

Magnetic recording
Main article: magnetic storage Magnetic recording was demonstrated in principle as early as 1898 by Valdemar Poulsen in his telegraphone. Magnetic wire recording, and its successor, magnetic tape recording, involve the use of a magnetizable medium which moves with a constant speed past a recording head. An electrical signal, which is analogous to the sound that is to be recorded, is fed to the recording head, inducing a pattern of magnetization similar to the signal. A playback head can then pick up the changes in magnetic field from the tape and convert it into an electrical signal. With the addition of electronic amplification developed by Curt Stille in the 1920s, the telegraphone evolved into wire recorders which were popular for voice recording and dictation during the 1940s and into the 1950s. The reproduction quality of wire recorders was low, however significantly lower than that achievable with phonograph disk recording technology. Wire 2

recorders could not prevent the wire from undergoing axial twisting, and hence could not ensure that the wire was oriented the same way during recording and playback. When oriented the wrong way, high frequencies were reduced and the sound was muffled. The hysteresis of the steel material resulted in nonlinear transfer characteristics, manifesting as distortion. There were other practical difficulties, such as the tendency of the wire to become tangled or snarled. Splicing could be performed by knotting together the cut wire ends, but the results were not very satisfactory. Early tape recorders were first developed in Germany. On Christmas day 1932 the British Broadcasting Corporation first used a tape recorder for their broadcasts. The device used was a Marconi-Stille recorder, a huge tape machine which used steel razor tape 3 mm wide and 0.08 mm thick. In order to reproduce the higher audio frequencies it was necessary to run the tape at a 90 metres per minute past the recording and reproducing heads. This meant that the length of tape required for a half-hour programme was nearly 3 kilometres and a full reel weighed 25 kg!

7" reel of " recording tape, typical of audiophile, consumer and educational use in the 1950s-60s Magnetic tape recording as we know it today was developed in Germany during the late 1930s by the C. Lorenz company and by AEG . In 1938, S. J. Begun left Germany and joined Brush Development Company in the United States, where work continued but attracted little attention. Multitrack recording The next major development in magnetic tape was multitrack recording, in which the tape is divided into multiple tracks parallel with each other. Because they are carried on the same medium, the tracks stay in perfect synchronization. The first development in multitracking was stereo sound, which divided the recording head into two tracks. First developed by German audio engineers ca. 1943, 2-track recording was rapidly adopted for modern music in the 1950s because it enabled signals from two or more separate microphones to be recorded simultaneously, enabling stereophonic recordings to be made and edited conveniently. (The first stereo recordings, on disks, had been made in the 1930s, but were never issued commercially.) Stereo (either true, twomicrophone stereo or multimixed) quickly became the norm for commercial classical recordings and radio broadcasts, although many pop music and jazz recordings continued to be issued in monophonic sound until the mid-1960s. Much of the credit for the development of multitrack recording goes to guitarist, composer and technician Les Paul, who also helped design the famous electric guitar that bears his name. His experiments with tapes and recorders in the early 1950s led him to order the first custom-built eight-track recorder from Ampex, and his pioneering recordings with his then wife, singer Mary Ford, were the first to make use of the technique of multitracking to record separate elements of a musical piece asynchronously that is, separate elements could be recorded at different times. Paul's technique enabled him to listen to the tracks he had already taped and record new parts in time alongside them. Multitrack recording was immediately taken up in a limited way by Ampex, who soon produced a commercial 3-track recorder. These proved extremely useful for popular music, since they enabled backing music to be recorded on two tracks (either to allow the overdubbing of separate parts, or to create a full stereo backing track) while the third track was reserved for the lead vocalist. Three3

track recorders remained in widespread commercial use until the mid-1960s and many famous pop recordings including many of Phil Spector's so-called "Wall of Sound" productions and early Motown hits were taped on Ampex 3-track recorders. The next important development was 4-track recording. The advent of this improved system gave recording engineers and musicians vastly greater flexibility for recording and overdubbing, and 4track was the studio standard for most of the later 1960s. Many of the most famous recordings by The Beatles and The Rolling Stones were recorded on 4-track, and the engineers at London's Abbey Road Studios became particularly adept at a technique called "reduction mixes" in the UK and "bouncing down" in the United States, in which multiple tracks were recorded onto one 4-track machine and then mixed together and transferred (bounced down) to one track of a second 4-track machine. In this way, it was possible to record literally dozens of separate tracks and combine them into finished recordings of great complexity. All of the Beatles classic mid-60s recordings, including the albums Revolver and Sgt Pepper's Lonely Hearts Club Band, were recorded in this way. There were limitations, however, because of the build-up of noise during the bouncing-down process, and the Abbey Road engineers are still justly famed for the ability to create dense multitrack recordings while keeping background noise to a minimum. 4-track tape also enabled the development of quadraphonic sound, in which each of the four tracks was used to simulate a complete 360-degree surround sound. A number of albums including Pink Floyd's Dark Side of the Moon and Mike Oldfield's Tubular Bells were released both in stereo and quadrophonic format in the 1970s, but 'quad' failed to gain wide commercial acceptance. Although it is now considered a gimmick, it was the direct precursor of the surround sound technology that has become standard in many modern home theatre systems. In a professional setting today, such as a studio, audio engineers may use 24 tracks or more for their recordings, utilizing one or more tracks for each instrument played. The combination of the ability to edit via tape splicing, and the ability to record multiple tracks, revolutionized studio recording. It became common studio recording practice to record on multiple tracks, and mix down afterwards. The convenience of tape editing and multitrack recording led to the rapid adoption of magnetic tape as the primary technology for commercial musical recordings. Although 33 rpm and 45 rpm vinyl records were the dominant consumer format, recordings were customarily made first on tape, then transferred to disc, with Bing Crosby leading the way in the adoption of this method in the United States.

Recording on film
The first attempts to record sound to an optical medium occurred around 1900. In 1906 Lauste applied for a patent to record sound on film, but was ahead of his time. In 1923 Lee de Forest applied for a patent to record to film; he also made a number of short experimental films, mostly of vaudeville performers. William Fox began releasing sound-on-film newsreels in 1926, the same year that Warner Brothers released Don Juan with music and sound effects recorded on discs, as well as series of short films with fully synchronized sound on discs. In 1927 the sound film The Jazz Singer was released; while not the first, it made a tremendous hit and made the public and the film industry realize that sound film was more than a mere novelty. The Jazz Singer used a process called Vitaphone, a process that involved synchronizing the projected film to sound recorded on disk. It essentially amounted to playing a phonograph record, but one that was recorded with the best electronic technology of the time. Audiences used to 4

acoustic phonographs and recordings would, in the theatre, have heard something resembling 1950s "high fidelity." In the 1920s, when the first talkies came out, especially The Jazz Singer, theatre orchestra musicians were being replaced with mechanical music which cost the loss of many jobs.[1] The American Federation of Musicians took out ads in newspapers, protesting the replacement of real musicians with mechanical playing devices, especially in theatres. [2] In the days of analog technology, however, no process involving a separate disk could hold synchronization precisely or reliably. Vitaphone was quickly supplanted by technologies which recorded a sound track optically directly onto the side of the strip of motion picture film. This was the dominant technology from the 1930s through the 1960s and is still in use as of 2004. There are really two different types of synchronised film soundtrack, optical and magnetic. Optical sound tracks are visual renditions of sound wave forms and provide sound through a light beam and optical sensor within the projector. Magnetic sound tracks are essentially the same as used in conventional analog tape recording. Magnetic soundtracks can be joined with the moving image but it creates an abrupt discontinuity because of the offset of the audio track relative to the picture. Whether optical or magnetic, the audio pickup must be located several inches ahead of the projection lamp, shutter and drive sprockets. There is usually a flywheel as well to smooth out the film motion to eliminate the flutter that would otherwise result from the pull-down mechanism. If you have films with a magnetic track, you should keep them away from strong magnetic sources, such as televisions. These can weaken or wipe the magnetic sound signal. Magnetic sound on an acetate base is also more prone to vinegar syndrome than a film with just the image.

A variable density soundtrack (left)and a bi-lateral variable area soundtrack (right) For optical recording on film there are two methods utilized. Variable density recording uses changes in the darkness of the soundtrack side of the film to represent the soundwave. Variable area recording uses changes in the width of a dark strip to represent the soundwave. In both cases light that is sent through the part of the film that corresponds to the soundtrack changes in intensity, proportional to the original sound, and that light is not projected on the screen but converted into an electrical signal by a light sensitive device. Optical soundtracks are prone to the same sorts of degradation that affect the picture: e.g. scratches, copying. Unlike the film image that creates the illusion of continuity, sound tracks are continuous. This means that if you cut and splice film with a combined soundtrack, the image will cut cleanly but the sound track will probably produce a cracking sound. Fingerprints on the film may also produce cracking or interference. Should you wish to use sound, there is of course no reason for employing either a magnetic or an optical sound track. You could use a secondary source to play alongside your images but bear in 5

mind that precise synching may be difficult. If you do use a secondary source, make sure you look after and document it precisely, as you would with the film it accompanies. You might also want to keep yourself informed for the future by looking into the technological information and history of the medium of this other source. In the late 1950s the cinema industry, desperate to provide a theatre experience that would be overwhelmingly superior to television, introduced wide-screen processes such as Cinerama, ToddAO, and CinemaScope. These processes at the same time introduced technical improvements in sound, generally involving the use of multitrack magnetic sound, recorded on an oxide stripe laminated onto the film. In subsequent decades, a gradual evolution occurred with more and more theatres installing various forms of magnetic-sound equipment. In the 1990s, digital systems were introduced and began to prevail. Ironically, in many of them the sound recording is, as in Vitaphone, again recorded on a separate disk; but now, digital processes can achieve reliable and perfect synchronization.

Digital recording

The DAT or Digital Audio Tape The first digital audio recorders were reel-to-reel decks introduced by companies such as Denon (1972), Soundstream (1979) and Mitsubishi. They used a digital technology known as PCM recording. Within a few years, however, many studios were using devices that encoded the digital audio data into a standard video signal, which was then recorded on a U-matic or other videotape recorder, using the rotating-head technology that was standard for video. A similar technology was used for a consumer format, Digital Audio Tape (DAT) which used rotating heads on a narrow tape contained in a cassette. DAT records at sampling rates of 48 kHz or 44.1 kHz, the latter being the same rate used on compact discs. Bit depth is 16 bits, also the same as compact discs. DAT was a failure in the consumer-audio field (too expensive, too finicky, and crippled by anti-copying regulations), but it became popular in studios (particularly home studios) and radio stations. A failed digital tape recording system was the Digital Compact Cassette (DCC). Within a few years after the introduction of digital recording, multitrack recorders (using stationary heads) were being produced for use in professional studios. In the early 1990s, relatively low-priced multitrack digital recorders were introduced for use in home studios; they returned to recording on videotape. The most notable of this type of recorder is the ADAT. Developed by Alesis and first released in 1991, the ADAT machine is capable of recording 8 tracks of digital audio onto a single S-VHS video cassette. The ADAT machine is still a very common fixture in professional and home studios around the world. In the consumer market, tapes and gramophones were largely displaced by the compact disc (CD) and a lesser extent the minidisc. These recording media are fully digital and require complex electronics to play back. Digital sound files can be stored on any computer storage medium. The development of the MP3 audio file format, and legal issues involved in copying such files, has driven most of the innovation in music distribution since their introduction in the late 1990s.

As hard disk capacities and computer CPU speeds increased at the end of the 1990s, hard disk recording became more popular. At this writing (early 2005) hard disk recording takes two forms. One is the use of standard desktop or laptop computers, with adapters for encoding audio into two or many tracks of digital audio. These adapters can either be in-the-box soundcards or external devices, either connecting to in-box interface cards or connecting to the computer via USB or Firewire cables. The other common form of hard disk recording uses a dedicated recorder which contains analog-to-digital and digital-to-analog converters as well as one or two removable hard drives for data storage. Such recorders, packing 24 tracks in a few units of rack space, are actually single-purpose computers, which can in turn be connected to standard computers for editing.

Sound Anatomy. Principles of Sound Synthesis

This article aims to discuss SOUND-DEF, principles, techniques and popular equipment to synthesise musical instruments How does sound travel? Sound comes from a series of vibrations. Sound travels in waves. When a source, or something that produces sound, vibrates, it transfers its energy to the surrounding particles causing them to vibrate. Those particles then bump into the ones next to them and so on. This causes the particles to move back and forth but waves of energy to move outward in all directions from the source. Your vocal chords and the strings on a guitar are both sources which vibrate to produce sounds. Without energy, there would be no sound. Let's take a closer look at sound waves. What do waves consist of? Waves are made up of compressions and rarefactions. Compression happens when molecules are forced, or pressed, together. Rarefaction is just the opposite, it occurs when molecules are given extra space and allowed to expand. Remember that sound is a type of kinetic energy. As the molecules are pressed together, they pass the kinetic energy to each other. Thus sound energy travels outward from the source. These molecules are part of a medium, which is anything that carries sound. Sound travels through air, water, or even a block of steel, thus, all are mediums for sound. Without a medium there are no molecules to carry the sound waves. In places like space, where there is no atmosphere, there is no sound. Let's look at the example of a stereo speaker. To produce sound, a thin surfaced cone, called a diaphragm, vibrates back and forth and creates energy. When the diaphragm moves to the right, its energy pushes the air molecules on the right together, opening up space for the molecules on the left to move into. We call the molecules on the right compressed and the molecules on the left rarefied. When the diaphragm moves to the left, the opposite happens. Now, the molecules to the left become compressed and the molecules to the right are rarefied. These alternating 7

compressions and rarefactions produce a wave. One compression and one rarefaction is called a wavelength. Different sounds have different wavelengths. Speeds of Sound Material Rubber Air at 40oC Glass Lead Stone Copper Elasticity The speed of sound is also different for different types of solids, liquids, and gases. Some materials, like nickel or iron, are more elastic than others. Elasticity describes how quickly the molecules return to their original positions. Molecules that return to their original shape quickly are ready to move again more quickly, thus they can vibrate at higher speeds. Sound can travel faster through mediums that vibrate faster. Sound travels faster through elastic solids like nickel or iron than through solids like lead, which is less elastic. Density The density of a medium also affects the speed of sound. Density describes the mass of a substance per volume. A substance that is denser per volume has more mass per volume. Usually, larger molecules have more mass. If a material is denser because its molecules are larger, it will transmit sound slower. Sound waves are made up of kinetic energy. It takes more energy to make large molecules vibrate than it does to make smaller molecules vibrate. Thus, sound will travel at a slower rate in the denser object. If sound waves of the same energy were passed through a block of wood and a block of steel, which is more dense than the wood, the molecules of the steel would vibrate at a slower rate. Thus, sound passes more quickly through the wood, which is less dense. Suppose, however, that two substances are made of different molecules which weigh the same amount. They have the same volume, but one substance is more dense. We know the denser substance must have more mass per volume. Since both substances have molecules of similar weight, this extra mass means the denser substance has more molecules per volume. More molecules are squeezed into the same volume, therefore the molecules are closer together. Since sound is more easily transmitted between close molecules, it travels faster in the denser substance. Sound moves 8 Speed of Sound 60 m/s 355 m/s 4540 m/s 1210 m/s 5971 m/s 3100 m/s

faster through denser air because the molecules are closer together in dense air and sound can be more easily passed on. Wave Interference When two or more sound waves from different sources are present at the same time, they interact with each other to produce a new wave. The new wave is the sum of all the different waves. Wave interaction is called interference. If the compressions and the rarefactions of the two waves line up, they strengthen each other and create a wave with a higher intensity. This type of interference is known as constructive.

When the compressions and rarefactions are out of phase, their interaction creates a wave with a dampened or lower intensity. This is destructive interference. When waves are interfering with each other destructively, the sound is louder in some places and softer in others. As a result, we hear pulses or beats in the sound. Dead spots Waves can interfere so destructively with one another that they produce dead spots, or places where no sound at all can be heard. Dead spots occur when the compressions of one wave line up with the rarefactions from another wave and cancel each other. Engineers who design theaters or auditoriums must take into account sound wave interference. The shape of the building or stage and the materials used to build it are chosen based on interference patterns. They want every member of the audience to hear loud, clear sounds.

1. THE HUMAN EAR

The human ear has three main sections, which consist of the outer ear, the middle ear, and the inner ear. Sound waves enter your outer ear and travel through your ear canal to the middle ear. The ear canal channels the waves to your eardrum, a thin, sensitive membrane stretched tightly over the entrance to your middle ear. The waves cause your eardrum to vibrate. It passes these vibrations on to the hammer, one of three tiny bones in your ear. The hammer vibrating causes the anvil, the small bone touching the hammer, to vibrate. The anvil passes these vibrations to the stirrup, another small bone which touches the anvil. From the stirrup, the vibrations pass into the inner ear. The stirrup touches a liquid filled sack and the vibrations travel into the cochlea, which is shaped like a shell. Inside the cochlea, there are hundreds of special cells attached to nerve fibers, which can transmit information to the brain. The brain processes the information from the ear and lets us distinguish between different types of sounds. Microphones. STEREO MICROPHONE TECHNIQUES Stereo miking is the preferred way to record classical-music ensembles and soloists, such as a symphony performed in a concert hall or a string quartet piece played in a recital hall. Stereo mic techniques capture the sound of a musical group as a whole, using only two or three microphones.When you play back a stereo recording, you hear phantom images of the instruments in various spots between the speakers. These image locations -- left to right, front to back -- correspond to the instrument locations during the recording session.In this article we'll look at several techniques for recording in stereo. Advantages of Stereo Miking When recording popular music, we put a mic near each instrument, record it, and pan its image somewhere between our two monitor speakers. Then we hear where each instrument is -- left, center, half-right, or whatever. But panned mono tracks are not the same as true stereo. A two mic stereo recording captures the holistic sound of the ensemble playing together in a shared space. Large single instruments -- such as piano, drums, and pipe organ -- also benefit from being recorded in stereo. Stereo miking adds lifelike realism to a recording because it captures: The left-to-right position of each instrument. The depth or distance of each instrument. The distance of the ensemble from the listener (the perspective). The spatial sense of the acoustic environment, the ambience or hall reverberation. The timbres of the instruments as heard in the audience. These characteristics are lost with multiple close-up microphones. Another advantage of stereo miking is that it tends to preserve the ensemble balance as intended by the composer. The composer has assigned dynamics (loudness notations) to the 10

instruments in order to produce a pleasing ensemble balance in the audience area. Thus, the correct balance or mix of the ensemble occurs at a distance, where all the instruments blend together acoustically. But this balance can be upset with multiple miking. You must rely on your own judgment (and the conductor's) regarding mixer settings to produce the composer's intended balance. Of course, even a stereo pair of mics can yield a faulty balance. But a stereo pair, being at a distance, is more likely to reproduce the balance as the audience hears it. Some outstanding examples of non-orchestral two-mic stereo recordings are those by Bob Katz Goals of Stereo Miking One goal we aim for when miking an ensemble in stereo is accurate localization. That is, instruments in the center of the group are reproduced midway between the two speakers. Instruments at the sides of the group are heard from the left or right speaker. Instruments halfway to one side are heard halfway to one side, and so on. Comparing the Four Techniques 1. Coincident pair: Uses two directional mics angled apart with grilles touching. Level differences between channels produce the stereo effect. Images are sharp. Stereo spread ranges from narrow to accurate. Signals are mono compatible. 2. Spaced pair: Uses two mics spaced a few feet apart, aiming straight ahead. Time differences between channels produce the stereo effect. Off-center images are diffuse. Stereo spread tends to be exaggerated unless a third center mic is used, or unless spacing is under 2 to 3 feet. Provides a warm sense of ambience. Provides excellent low-frequency response if you use omni condensers. Tends not to be mono compatible, but this may not be audible. 3. Near-coincident pair: Uses two directional mics angled apart and spaced a few inches apart horizontally. Level and time differences between channels produce the stereo effect. Images are sharp. Stereo spread tends to be accurate. The hall sounds more spacious than with coincident methods. Tends not to be mono compatible. 4. Baffled omni pair: Uses two omni mics, usually ear-spaced, with a baffle between them. Level, time, and spectral differences produce the stereo effect. Images are sharp. Stereo spread tends to be accurate. Excellent low-frequency response. Good imaging with headphones. The hall sounds more spacious than with coincident methods. Stereo spread is not adjustable except by panning the two channels toward the center. More conspicuous than other methods. 11

 Tends not to be mono compatible, but this might not be audible. Boundary (surface-mounted) mics, either with a hemispherical or half-cardioid pattern, can be used for any type of stereo miking.

5.Mixing console
In professional audio, a mixing console, digital mixing console, mixing desk (Brit.), or audio mixer, also called a sound board or soundboard, is an electronic device for combining (also called "mixing"), routing, and changing the level, tonality and/or dynamics of audio signals. A mixer can mix analog or digital signals, depending on the type of mixer. The modified signals (voltages or digital samples) are summed to produce the combined output signals. Mixing consoles are used in many applications, including recording studios, public address systems, sound reinforcement systems, broadcasting, television, and film post-production. An example of a simple application would be to enable the signals that originated from two separate microphones (each being used by vocalists singing a duet, perhaps) to be heard through one set of speakers simultaneously. When used for live performances, the signal produced by the mixer will usually be sent directly to an amplifier, unless that particular mixer is powered or it is being connected to powered speakers. The input strip is usually separated into these sections:

Input Jacks / Microphone preamps Basic input controls Channel EQ Routing Section including Direct Outs, Aux-sends, Panning control and Subgroup assignments Input Faders Subgroup faders Output controls including Master level controls, EQ and/or Matrix routing

On the Yamaha Console to the right, these sections are color coded for quick identification by the operator. Each signal that is input into the mixer has its own channel. Depending on the specific mixer, each channel is stereo or monaural. On most mixers, each channel has an XLR input, and many have RCA or quarter-inch Jack plug line inputs.

Basic input controls

Below each input, there are usually several rotary controls (knobs, pots). The first is typically a trim or gain control. The inputs buffer the signal from the external device and this controls the amount of amplification or attenuation needed to bring the signal to a nominal level for processing. This stage is where most noise or interference is picked up, due to the high gains involved (around +50 dB, for a microphone). Balanced inputs and connectors, such as XLR or Tip-Ring-Sleeve (TRS) quarter-inch connectors, reduce interference problems. There may be insert points after the buffer/gain stage, which send to and return from external processors which should only affect the signal of that particular channel. Insert points are most commonly used with effects that control a signal's amplitude, such as noise gates, expanders, and compressors. 12

Auxiliary send routing

The Auxiliary send routes a split of the incoming signal to an auxiliary bus which can then be used with external devices. Auxiliary sends can either be pre-fader or post-fader, in that the level of a pre-fade send is set by the Auxiliary send control, whereas post-fade sends depend on the position of the channel fader as well. Auxiliary sends can be used to send the signal to an external processor such as a reverb, which can then be routed back through another channel or designated auxiliary returns on the mixer. These will normally be post-fader. Pre-fade auxiliary sends can be used to provide a monitor mix to musicians onstage, this mix is thus independent of the main mix.

Channel EQ
Further channel controls affect the equalization of the signal by separately attenuating or boosting a range of frequencies (e.g., bass, midrange, and treble frequencies). Most large mixing consoles (24 channels and larger) usually have sweep equalization in one or more bands of its parametric equalizer on each channel, where the frequency and affected bandwidth of equalization can be selected. Smaller mixing consoles have few or no equalization control. Care must be taken not to add too much EQ to a signal that is already close to clipping; additional energy will overdrive the channel. Some mixers have a general equalization control (either graphic or parametric) at the output.

Subgroup and mix routing

Each channel on a mixer has an audio taper pot, or potentiometer, controlled by a sliding volume control (fader), that allows adjustment of the level, or amplitude, of that channel in the final mix. A typical mixing console has many rows of these sliding volume controls. Each control adjusts only its respective channel (or one half of a stereo channel); therefore, it only affects the level of the signal from one microphone or other audio device. The signals are summed to create the main mix, or combined on a bus as a submix, a group of channels that are then added to get the final mix (for instance, many drum mics could be grouped into a bus, and then the proportion of drums in the final mix can be controlled with one bus fader). There may also be insert points for a certain bus, or even the entire mix.

Master output controls

Subgroup and main output fader controls are often found together on the right hand side of the mixer or, on larger consoles, in a center section flanked by banks of input channels. Matrix routing is often contained in this master section, as are headphone and local loudspeaker monitoring controls. Talkback controls allow conversation with the artist through their wedges, headphones or IEMs. A test tone generator might be located in the master output section. Aux returns such as those signals returning from outboard reverb devices are often in the master section.

Metering
Finally, there are usually one or more VU or peak meters to indicate the levels for each channel, or for the master outputs, and to indicate whether the console levels are overmodulating or clipping the signal. Most mixers have at least one additional output, besides the main mix. These are either individual bus outputs, or auxiliary outputs, used, for instance, to output a different mix to on-stage monitors. The operator can vary the mix (or levels of each channel) for each output. 13

As audio is heard in a logarithmic fashion (both amplitude and frequency), mixing console controls and displays are almost always in decibels, a logarithmic measurement system. This is also why special audio taper pots or circuits are needed. Since it is a relative measurement, and not a unit itself (like a percentage), the meters must be referenced to a nominal level. The "professional" nominal level is considered to be +4 dBu. The "consumer grade" level is 10 dBV.

Hardware routing and patching

For convenience, some mixing console racks contain a patch bay or patch panel. These may be more useful for those not using a computer with several plugins on their software. Most, but not all, audio mixers can

add external effects. use monaural signals to produce stereo sound by adjusting the position of each signal on the sound stage (pan and balance controls). provide phantom power (typically 48 volts) required by some microphones. create an audible tone via an oscillator, usually at 440 Hz, 1 kHz, or 2 kHz

Some mixers can

add effects internally. interface with computers or other recording equipment (to control the mixer with computer presets, for instance). be powered by batteries.

6.Compressor
Compressors reduce the dynamic range of an audio signal. In the real world, its generally an amplifier with two gain levels: the gain is unity for input signal levels below a certain threshold, and less than unity for signals with levels above the threshold. For example, compressors can be used to eliminate the variations in the peaks of an electric bass signal by clamping them to a constant level, thus providing an even, solid bass line. Compressors can also be useful in compensating for the wide variations in the level of a signal produced by a vocalist who moves frequently or has an erratic dynamic range.

Parameters
Threshold - The Threshold parameter sets the dB level where the compressor will kick in (with a range from 0.0 to -60.0 dB). The threshold should be adjusted according to the relative input level and the type of audio material. Once the threshold level is reached, compression will start, reducing the gain of the input signal according to the current Ratio, Type, Attack and Release settings. Ratio - Controls the amount of compression (gain reduction) that will be applied to the signal once the threshold level is reached (with a range from 0.4:1 to 30:1). Ratio denotes the difference in dB between input level and output level, i.e. how much the signal above threshold level will be compressed (or expanded, at ratios below 1:1). For example, a ratio of 4:1 means that when the input level increases by 4dB, the output level of the signal above threshold will only increase by 1dB. 14

Gain - Controls the amount of make-up gain to be added or subtracted from the compressed output signal (with a range from 30.0 to -30.0 dB). The gain should be adjusted to normalize the signal amplitude after compression, or to control the amount of limiting. Attack - Controls the time it takes to reach full compression once the threshold level has been reached (with a range from 0.0 to 400.0 ms). A fast attack setting means that compression will be more or less instant. Using a slower attack setting results in the compression being gradually increased, allowing for more variations in the signal than the fast setting. Attack should be adjusted according to the nature of the audio material. Release - The Release parameter sets the time the compressor takes to go back to an inactive state after the level has fallen below threshold (with a range from 1 to 4000 ms). Short release times will make the compression more flexible and able to adapt to the input signal, but can cause fast changes in gain that may sound displeasing to the ears. Longer release times produce a signal with a more even level and less distortion, but make it harder to maximize the overall compression because small variations in signal level will be ignored. Type - This parameter controls the knee type and TCR. The possible values are: Hard, Medium, Vintage, Soft, Hard/R, Medium/R, Vintage/R, Soft/R. The knee determines the dB range above and below the threshold where the compression goes from 1:1 to the selected compression ratio. A hard knee setting means that compression will take place immediately after the threshold level is reached, whereas a soft knee setting indicates that compression is gradually applied over a range in the signal. See Knee Type below for more information on the meaning of those values.

Knee Type
The actual values for this property mean:

Hard - 0 dB Medium - 6 dB Vintage - 7 dB Soft - 15 dB

The Vintage compression type emulates the compression curve found on some analog compressors, such as the classic Teletronix LA2A. The major difference is that the compression ratio is gradually reduced at a distance above threshold, slowly allowing the level to go back to a ratio of 1:1. This allows the loudest parts of the signal, such as drum beats and other peaks, to pass without being compressed as much as the rest of the signal. In this way, the Vintage compression type emulates electro-optical analog designs and can greatly enhance warmth and 'punch'. The Vintage compression type also affects the TCR parameter, utilizing a different release time adjustment method. The /R types enable TCR (Transient Controlled Release), a special algorithm that automatically adjusts the release time in real-time to avoid fast compression changes. The release time is adjusted in relation to the current Release parameter setting. Enabling TCR can have positive effects on some types of audio material, and help to reduce "pumping and breathing", while increasing the overall loudness of the signal.

Explanation
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Different compression ratios A compressor reduces the gain (level) of an audio signal if its amplitude exceeds a threshold. The amount of gain reduction is determined by a ratio. For example, with a ratio of 4:1, when the (time averaged) input level is 4 dB over the threshold, the output signal level will be 1 dB over the threshold. The gain (level) has been reduced by 3dB. When the input level is 8dB above the threshold, the output level will be 2dB; a 6dB gain reduction. A more specific example for a 4:1 ratio: Threshold = 10dB Input = 14dB (4 dB above the threshold) Output = 11dB (1 dB above the threshold) A compressor uses a variable-gain amplifier to reduce the gain of the signal. There are a number of technologies used for this purpose, each having different advantages and disadvantages. Vacuum tubes are used in configuration called 'variable-' where the grid-to-cathode voltage changes the to alter the gain.[2] Also used is a voltage controlled amplifier which has its gain reduced as the power of the input signal increases. Optical compressors use a light sensitive resistor (LDR) and a small lamp(LED or Electroluminescent panel) to create changes in signal gain. This technique is believed by some to add smoother characteristics to the signal, because the response times of the light and the resistor soften the attack and release. Other Technologies used include Field Effect Transistors and a Diode Bridge.[3] When working with digital audio, digital signal processing techniques are commonly used to implement compression via digital audio editors, or dedicated workstations. Often the algorithms used emulate the above analog technologies.

Limiting
It has been suggested that Limiting be merged into this article or section. (Discuss) Compression and limiting are no different in process, just in degree and in the perceived effect. A limiter is a compressor with a higher ratio, and generally a fast attack time. Most engineers consider a ratio of 10:1 or more as limiting, although there are no set rules.[5] Engineers sometimes refer to soft and hard limiting which are differences of degree. The "harder" a limiter, the higher its ratio and the faster its attack and release times. Brick wall limiting has a very high ratio and a very fast attack time. Ideally, this ensures that an audio signal never exceeds the amplitude of the threshold. Ratios of 20:1 all the way up to :1 are 16

considered to be 'brick wall'.[6] The sonic results of more than momentary and infrequent hard/brick-wall limiting are usually characterized as harsh and unpleasant; thus it is more appropriate as a safety device in live sound and broadcast applications than as a sound-sculpting tool. Some modern consumer electronics devices incorporate limiters. Sony uses the Automatic Volume Limiter System (AVLS), on some audio products and the PlayStation Portable.

Limit ing and Clipping compared. Note that clipping introduces a large amount of distortion whereas Limiting only introduces a small amount while keeping the signal within the threshold.

Common uses
Public spaces
Compression is often used to make music sound louder without increasing its peak amplitude. By compressing the peak (or loudest) signals, it becomes possible to increase the overall gain (or volume) of a signal without exceeding the dynamic limits of a reproduction device or medium. The net effect, when compression is applied along with a gain boost, is that relatively quiet sounds become louder, while louder sounds remain unchanged. Compression is often applied in this manner in audio systems for restaurants, retail, and similar public environments, where background music is played at a relatively low volume and needs to be compressed not just to keep the volume fairly constant, but also in order for relatively quiet parts of the music to be audible over ambient noise, or audible at all. Compression can be used to increase the average output gain of a power amplifier by 50 to 100% with a reduced dynamic range. For paging and evacuation systems, this adds clarity under noisy circumstances and saves on the number of amplifiers required. 17

Music production
Compression is often used in music production to make performances more consistent in dynamic range so that they "sit" in the mix of other instruments better and maintain consistent attention from the listener. Vocal performances in rock music or pop music are usually compressed in order to make them stand out from the surrounding instruments and to add to the clarity of the vocal performance. Compression can also be used on instrument sounds to create effects not primarily focused on boosting loudness. For instance, drum and cymbal sounds tend to decay quickly, but a compressor can make the sound appear to have a more sustained tail. Guitar sounds are often compressed in order to obtain a fuller, more sustained sound. Most devices capable of compressing audio dynamics can also be used to reduce the volume of one audio source when another audio source reaches a certain level; see Side-Chaining above.

Voice
A compressor can be used to reduce sibilance ('ess' sounds) in vocals by feeding the compressor with an EQ set to the relevant frequencies, so that only those frequencies activate the compressor. If unchecked, sibilance could cause distortion even if sound levels are not very high. This usage is called 'de-essing'. [1] Compression is used in voice communications in amateur radio that employ SSB modulation. Often it is used to make a particular station's signal more readable to a distant station, or to make one's station's transmitted signal stand out against others. This occurs especially in pileups where amateur radio stations are competing for the opportunity to talk to a DX station. Since an SSB signal's amplitude depends on the level of modulation, the net result is that the average amplitude of the signal and hence average transmitted power would be stronger than it would be had compression not been used.[7] Most modern amateur radio SSB transceivers have speech compressors built in. Compression is also used in land mobile radio, especially in transmit audio of professional walkietalkies and in remote control dispatch consoles.

Broadcasting
Compression is used extensively in broadcasting to boost the perceived volume of sound while reducing the dynamic range of source audio (typically CDs) to a range that can be accommodated by the narrower-range broadcast signal. Broadcasters in most countries have legal limits on instantaneous peak volume they may broadcast. Normally these limits are met by permanently inserted hardware in the on-air chain (see multiband compression above). The same recording can have very different dynamics when heard via AM, FM, CD, or other media (although frequency response and noise are large factors as well).

Marketing

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With the advent of the CD and digital music, record companies, mixing engineers and mastering engineers have been gradually increasing the overall volume of commercial albums by using higher degrees of compression and limiting during mixing and mastering. Compression algorithms have been engineered specifically to accomplish the task of maximizing audio level in the digital stream. Hard limiting or hard clipping are the result, affecting the tone and timbre of the music in a way that one critic describes as "dogshit". [8] The effort to increase loudness has been referred to as the "loudness wars". Most television commercials are compressed heavily (typically to a dynamic range of no more than 3dB) in order to achieve near-maximum perceived loudness while staying within permissible limits.

Other uses
A compressor is sometimes used to reduce the dynamic range of a signal for transmission, to be expanded afterwards. This reduces the effects of a channel with limited dynamic range. See Companding. Gain pumping, where a regular amplitude peak (such as a kick drum) causes the rest of the mix to change in volume due to the compressor, is generally avoided in music production. However, many dance musicians use this phenomenon in a creative way, causing the mix to alter in volume rhythmically in time with the beat. A compressor is used in hearing aids to bring the audio volume in the range of the ears of the patient. To allow the patient to still hear the direction from which the sound is coming, binaural compression may be required.

Compressors for software audio players

Some software audio players support plugins which implement compression. These can be used to increase the perceived volume of audio tracks, or to even out the volume of highly-variable music (such as classical music, or a playlist spanning many music types). This improves the listenability of audio when played through poor-quality speakers, or when played in noisy environments (such as in a car or during a party). Such software may also be used in micro-broadcasting or homebased audio mastering.
7.STUDIO ACOUSTICS High frequencies are best absorbed by porous, fibrous materials such as fiberglass insulation, acoustic tile, foam plastic, carpeting, and curtains. Spacing these materials several inches from the wall extends their absorbtion into the midbass region. Low frequency absorbers called bass traps can be formed of flexible surfaces such as wood panelling or linoleum mounted over a sealed air space of several inches. Cavities such as closets or air spaces behind couches are also effective sound absorbers. If a room is absorbent at high frequencies both the live and recorded sound are likely to be boomy and muddy, due to the persistance of low frequency reverberation, ie. If the room has an abundance of fibrous materials but has no bass traps. A completely dead room though is stifling and musicians may feel they are playing in a vacuum so some reflections are beneficial, not only for comfort but for a sense of air and liveliness. Reflections also enhance the apparent loudness, transient response and timbre of acoustic instruments.

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The following acoustic treatments reduce reverberation: Open closet doors and place couches and books a few inches from the walls. Carpet the floor.(Not the walls) Hang canvas from the ceiling in deep folds. Hang thick curtains or blankets at least 2 feet from the walls, if possible. Attach open cell acoustic foam wedges (such as Sonex or Cutting Wedge) on or near the walls. The thicker the foam the better the low absorbtion. 4 inch foam on the walls absorbs from about 400 Hz up. In a basement studio, nail acoustic tile to the ceiling with fibreglass insulation in the air space between tiles and ceiling. For bass trapping make panel absorbers by nailing 1/4" and 1/8" ply to 2" furring strips (battens) and place fiberglass insulation in the air space behind the panel. Cover about half the wall. For wide range absorbtion attach 2" or 4" pressed fiberglass board (Owens-Corning Type 703, 3lb/cu.ft) onto 2" by 6" studs, spaced 4 ft apart on the existing wall with fiberglass insulation in the air space. Placing the absorbent material in sections rather than all together promotes an even diffusion of sound in the room. The following tips muffle noises from outside the studio: Weatherstrip doors all around. Replace hollow doors with solid. Block openings in thw room with thick plywood and caulking. Put several layers of plywood and carpet on the floor above the studio, and put insulation in the air space between the studio ceiling and the floor above. When building a new studio, reduce noise transmission through walls by using plastered concrete blocks. Nail gypsum board to 2x4" staggered studs.

Summarized Requirements

requirement Non parallel or absorbent walls

result No flutter echoes.

Sufficient sound absorbent surfaces on walls, ceiling Fairly low reverberation surfaces on walls, and floor. ceiling and time (about 0.4 second) Flexible panels, cavities and fibrous materials, or fibrous materials spaced from walls and ceiling. A few hard reflective room surfaces A large or non square room and bass traps. A large sound absorbent room. Equal reverb time at frequencies up to 4 kHz Early reflections to enhance the sound of acoustic instruments Minimized room modes. Low leakage.

8.Foley artist
The Foley artist on a film crew is the person who creates many of the natural, everyday sound effects in a film, which are recorded during a session with a recording engineer. Before the session, a project will be "cued", with notes kept about what sounds need to be created during the foley 20

session. Often, the project will have a sound supervisor who will dictate what sounds need to be covered in a foley session, and what needs to be created by special (audio) effects, which is generally left to the sound designer. The roles of Foley artists, sound designers, editors, and supervisors are highly specialized and are essential to producing a professional-sounding soundtrack that is suitable for distribution and exhibition. Sound effects and foley are added during post-production to dialog and real effects which were picked up by microphones on set. Sometimes (especially in the case of cartoons) there is no additional sound, and all the sounds need to be added by the foley artist and sound designer. The Foley artist may also accent existing sounds to make them more effective; enhancing the sounds of a fistfight may require thumping watermelons or cracking bamboo. Many Foley artists take pride in devising their own sound effects apparati, often using simple, commonly-found materials. Some "making-of" featurettes show Foley artists at work. The term "Foley artist" is named after Jack Foley, one of the earliest and best-known Hollywood practitioners of the art. Foley began his career in the film industry as a stand-in and screenwriter during the silent era, and later helped Universal make the transition from silent movies to "talkies". Because Foley refers to a person, the term is usually capitalized. However, because it is a person's name and not the trademark of a machine or process, no or symbol is used. The Universal Studios Hollywood theme park presented a demonstration in its "World of Cinemagic" feature.

How some effects are made

Effect Galloping horses Kissing Punching someone High heels Bone-breaking blow Footsteps in snow Star Trek sliding doors Star Wars sliding doors Bird flapping its wings Grass or leaves crunching How It's Made Banging empty coconut shells together Kissing back of hand Thumping watermelons Artist walks in high heels on wooden platform Breaking celery or bamboo or twisting a head of lettuce Squeezing a box of corn starch Pulling a piece of paper from an envelope Flare gun plus sneakers squeak Flapping a pair of gloves Balling up audio tape

9.MIDI Basics
The Musical Instrument Digital Interface (MIDI) protocol provides a standardized and efficient means of conveying musical performance information as electronic data. MIDI information is transmitted in "MIDI messages", which can be thought of as instructions which tell a music synthesizer how to play a piece of music. The synthesizer receiving the MIDI data must generate the actual sounds. The MIDI 1.0 Detailed Specification provides a complete description of the MIDI protocol. The MIDI data stream is a unidirectional asynchronous bit stream at 31.25 Kbits/sec. with 10 bits transmitted per byte (a start bit, 8 data bits, and one stop bit). The MIDI interface on a MIDI instrument will generally include three different MIDI connectors, labeled IN, OUT, and THRU. The MIDI data stream is usually originated by a MIDI controller, such as a musical instrument keyboard, or by a MIDI sequencer. A MIDI controller is a device which is played as an instrument, 21

and it translates the performance into a MIDI data stream in real time (as it is played). A MIDI sequencer is a device which allows MIDI data sequences to be captured, stored, edited, combined, and replayed. The MIDI data output from a MIDI controller or sequencer is transmitted via the devices' MIDI OUT connector. The recipient of this MIDI data stream is commonly a MIDI sound generator or sound module, which will receive MIDI messages at its MIDI IN connector, and respond to these messages by playing sounds. Figure 1 shows a simple MIDI system, consisting of a MIDI keyboard controller and a MIDI sound module. Note that many MIDI keyboard instruments include both the keyboard controller and the MIDI sound module functions within the same unit. In these units, there is an internal link between the keyboard and the sound module which may be enabled or disabled by setting the "local control" function of the instrument to ON or OFF respectively. The single physical MIDI Channel is divided into 16 logical channels by the inclusion of a 4 bit Channel number within many of the MIDI messages. A musical instrument keyboard can generally be set to transmit on any one of the sixteen MIDI channels. A MIDI sound source, or sound module, can be set to receive on specific MIDI Channel(s). In the system depicted in Figure 1, the sound module would have to be set to receive the Channel which the keyboard controller is transmitting on in order to play sounds.

Figure 1: A Simple MIDI System Information received on the MIDI IN connector of a MIDI device is transmitted back out (repeated) at the devices' MIDI THRU connector. Several MIDI sound modules can be daisychained by connecting the THRU output of one device to the IN connector of the next device downstream in the chain. Figure 2 shows a more elaborate MIDI system. In this case, a MIDI keyboard controller is used as an input device to a MIDI sequencer, and there are several sound modules connected to the sequencer's MIDI OUT port. A composer might utilize a system like this to write a piece of music consisting of several different parts, where each part is written for a different instrument. The composer would play the individual parts on the keyboard one at a time, and these individual parts would be captured by the sequencer. The sequencer would then play the parts back together through the sound modules. Each part would be played on a different MIDI Channel, and the sound modules would be set to receive different channels. For example, Sound module number 1 might be set to play the part received on Channel 1 using a piano sound, while module 2 plays the information received on Channel 5 using an acoustic bass sound, and the drum machine plays the percussion part received on MIDI Channel 10.

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Figure 2: An Expanded MIDI System In this example, a different sound module is used to play each part. However, sound modules which are "multitimbral" are capable of playing several different parts simultaneously. A single multitimbral sound module might be configured to receive the piano part on Channel 1, the bass part on Channel 5, and the drum part on Channel 10, and would play all three parts simultaneously. Figure 3 depicts a PC-based MIDI system. In this system, the PC is equipped with an internal MIDI interface card which sends MIDI data to an external multitimbral MIDI synthesizer module. Application software, such as Multimedia presentation packages, educational software, or games, sends MIDI data to the MIDI interface card in parallel form over the PC bus. The MIDI interface converts this information into serial MIDI data which is sent to the sound module. Since this is a multitimbral module, it can play many different musical parts, such as piano, bass and drums, at the same time. Sophisticated MIDI sequencer software packages are also available for the PC. With this software running on the PC, a user could connect a MIDI keyboard controller to the MIDI IN port of the MIDI interface card, and have the same music composition capabilities discussed in the last two paragraphs. There are a number of different configurations of PC-based MIDI systems possible. For instance, the MIDI interface and the MIDI sound module might be combined on the PC add-in card. In fact, the Multimedia PC (MPC) Specification requires that all MPC systems include a music synthesizer, and the synthesizer is normally included on the audio adapter card (the "sound card") along with the MIDI interface function. Until recently, most PC sound cards included FM synthesizers with limited capabilities and marginal sound quality. With these systems, an external wavetable synthesizer module might be added to get better sound quality. Recently, more advanced sound cards have been appearing which include high quality wavetable music synthesizers on-board, or as a daughter-card options. With the increasing use of the MIDI protocol in PC applications, this trend is sure to continue.

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Figure 3: A PC-Based MIDI System

MIDI Messages
A MIDI message is made up of an eight-bit status byte which is generally followed by one or two data bytes. There are a number of different types of MIDI messages. At the highest level, MIDI messages are classified as being either Channel Messages or System Messages. Channel messages are those which apply to a specific Channel, and the Channel number is included in the status byte for these messages. System messages are not Channel specific, and no Channel number is indicated in their status bytes. Channel Messages may be further classified as being either Channel Voice Messages, or Mode Messages. Channel Voice Messages carry musical performance data, and these messages comprise most of the traffic in a typical MIDI data stream. Channel Mode messages affect the way a receiving instrument will respond to the Channel Voice messages. Channel Voice Messages Channel Voice Messages are used to send musical performance information. The messages in this category are the Note On, Note Off, Polyphonic Key Pressure, Channel Pressure, Pitch Bend Change, Program Change, and the Control Change messages. Note On / Note Off / Velocity In MIDI systems, the activation of a particular note and the release of the same note are considered as two separate events. When a key is pressed on a MIDI keyboard instrument or MIDI keyboard controller, the keyboard sends a Note On message on the MIDI OUT port. The keyboard may be set to transmit on any one of the sixteen logical MIDI channels, and the status byte for the Note On message will indicate the selected Channel number. The Note On status byte is followed by two data bytes, which specify key number (indicating which key was pressed) and velocity (how hard the key was pressed).

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The key number is used in the receiving synthesizer to select which note should be played, and the velocity is normally used to control the amplitude of the note. When the key is released, the keyboard instrument or controller will send a Note Off message. The Note Off message also includes data bytes for the key number and for the velocity with which the key was released. The Note Off velocity information is normally ignored. Aftertouch Some MIDI keyboard instruments have the ability to sense the amount of pressure which is being applied to the keys while they are depressed. This pressure information, commonly called "aftertouch", may be used to control some aspects of the sound produced by the synthesizer (vibrato, for example). If the keyboard has a pressure sensor for each key, then the resulting "polyphonic aftertouch" information would be sent in the form of Polyphonic Key Pressure messages. These messages include separate data bytes for key number and pressure amount. It is currently more common for keyboard instruments to sense only a single pressure level for the entire keyboard. This "Channel aftertouch" information is sent using the Channel Pressure message, which needs only one data byte to specify the pressure value. Pitch Bend The Pitch Bend Change message is normally sent from a keyboard instrument in response to changes in position of the pitch bend wheel. The pitch bend information is used to modify the pitch of sounds being played on a given Channel. The Pitch Bend message includes two data bytes to specify the pitch bend value. Two bytes are required to allow fine enough resolution to make pitch changes resulting from movement of the pitch bend wheel seem to occur in a continuous manner rather than in steps. Program Change The Program Change message is used to specify the type of instrument which should be used to play sounds on a given Channel. This message needs only one data byte which specifies the new program number. Control Change MIDI Control Change messages are used to control a wide variety of functions in a synthesizer. Control Change messages, like other MIDI Channel messages, should only affect the Channel number indicated in the status byte. The Control Change status byte is followed by one data byte indicating the "controller number", and a second byte which specifies the "control value". The controller number identifies which function of the synthesizer is to be controlled by the message. A complete list of assigned controllers is found in the MIDI 1.0 Detailed Specification. - Bank Select Controller number zero (with 32 as the LSB) is defined as the bank select. The bank select function is used in some synthesizers in conjunction with the MIDI Program Change message to expand the number of different instrument sounds which may be specified (the Program Change message alone allows selection of one of 128 possible program numbers). The additional sounds are selected by preceding the Program Change message with a Control Change message which specifies a new value for Controller zero and Controller 32, allowing 16,384 banks of 128 sound each. 25

Since the MIDI specification does not describe the manner in which a synthesizer's banks are to be mapped to Bank Select messages, there is no standard way for a Bank Select message to select a specific synthesizer bank. Some manufacturers, such as Roland (with "GS") and Yamaha (with "XG") , have adopted their own practices to assure some standardization within their own product lines.

10.Frequency
To compare the range of frequencies in human experience, a video signal - for example - has an upper frequency of around 5 MHz (depending on the particular system in use in your part of the world). The Olympic Games have a frequency of 8 nanohertz (they happen once every four years!).

1 hertz (Hz) means one cycle of vibration per second 1000 Hz = 1 kHz 1,000,000 Hz = 1 Megahertz (1 MHz)

Sound comes in virtually all frequencies but our hearing system only responds to a narrow range. The upper limit of young human ears is usually taken to be 20 kilohertz (kHz) (twenty thousand vibrations per second). This varies from person to person, and decreases with age, but as a guideline its a good compromise. If a sound system can handle frequencies up to 20 kHz then few people will miss anything significant. At the lower end of the range it is difficult to know where the ear stops working and you start to feel vibration in your body. In sound engineering however we put a figure of 20 Hz on the lower end. We can hear, or feel, frequencies lower than this but they are generally taken to be unimportant. Frequency is related to wavelength by the formula: velocity = frequency x wavelength This applies to any wave motion, not just sound. The velocity, or speed, of sound in air is a little under 340 meters per second (m/s). This varies with temperature, humidity and altitude, but 340 m/s is a nice round number and well stick with it. If you work out the math, this means that a 20 Hz sound wave travelling in air has a wavelength of 17 metres! The extreme physical size of low frequency sound waves leads to tremendous problems in soundproofing and acoustic treatment. At the other end of the scale, a 20 kHz sound wave travelling in air has a wavelength of a mere 17 mm. Curiously, the higher the frequency the more difficult it is to handle as an electronic, magnetic or other form of signal, but it is really easy to control as a real-life sound wave travelling in air. Low frequencies are easily dealt with electronically, but are very hard to control acoustically.

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11.The Decibel
Decibels are a convenience that allows us to compare and quantify levels in the same manner through different media. In sound terms, decibels can be used for every medium that can store or transport sound or a sound signal:

Real sound travelling in air Electric signal Magnetic signal Digital signal Optical signal on a film sound track Mechanical signal on a vinyl record

A change in level of 3 dB means exactly the same thing in any of these media. Without decibels we would have to convert from newtons per square metre (sound pressure), volts, nanowebers per square metre, etc. Decibels have another advantage for sound. The ear assesses sound levels logarithmically rather than linearly. So a change in sound pressure of 100 N/m2 (micro-newtons per square meter) would be audibly different if the starting point were quiet (where it would be a significant change in level) then if it were loud (where it would be hardly any change at all). A change of 3 dB is subjectively the same degree of change at any level within the ears range. [Sound pressure is measured in newtons per square meter. You may think of the newton as a measure of weight. One newton is about the weight of a small apple.] An indication of how much one newton weighs An important point to bear in mind is that the decibel is a ratio, not a unit. It is always used to compare two sound levels. To convert to decibels apply the following formula: 20 x log10 (P1 /P2 ) where P1 and P2 are the two sound pressures you want to compare. So if one sound is twice the pressure of another then P1 /P2 = 2. The logarithm of 2 (base 10) is 0.3, and multiplying this by 20 gives 6 dB. Actually its 6.02 dB but we dont worry about the odd 0.02. This is useful because we commonly need to, say, increase a level by 6 dB, but it doesnt actually tell us how loud any particular sound is because the decibel is not a unit. The answer to this is to use a reference level as a zero point. The level chosen is 20 N/m2 (twenty micronewtons per square meter), which is, according to experimental data, the quietest sound the average person can hear. We call this level the threshold of hearing and it can be compared to the rustle of a falling autumn leaf at ten paces. We quantify this as 0 dB SPL (sound pressure level) and now any sound can be compared with this zero level. Loud music comes in at around 100 dB SPL and the ear 27

starts to feel a tickling sensation at around 120 dB SPL, and hurts when levels approach 130 dB SPL. If you are not comfortable with math, it is useful to remember the following, which apply to both sound pressure and voltage (decibels work differently when referring to power):

-80 dB = one ten thousandth -60 dB = one thousandth -40 dB = one hundredth -20 dB = one tenth -12 dB = one quarter -6 dB = one half 0 dB = no change 6 dB = twice 12 dB = four times 20 dB = ten times 40 dB = one hundred times 60 dB = one thousand times 80 dB = ten thousand times Threshold of hearing = 0 dB SPL Threshold of feeling = 120 dB SPL Threshold of pain = 130 dB SPL

Do you need to understand decibels to be a sound engineer? The answer is, Yes - to a point. You need to be able to relate a change in decibels to a fader movement, and from there to an image in your aural imagination of what that change should sound like. In addition to that, youll get producers telling you to raise the level of the vocal a bit. How many decibels equal a bit? Only the experience you will gain in the early years of your career will tell you.

The Inverse Square Law

There is more to find out about the inverse square law. Here is an interesting point... The maximum rate of decay of a sound as you move away from it is 6 decibels per doubling of distance (the sound pressure halves). This is simply due to the spreading-out of sound - the same energy has to cover an ever greater area. If the sound is focused in any way, by a large source or by reflection, then it will fade away at a rate less than 6 dB per doubling of distance. This fact is of great importance to PA system designers. The ultimate focused sound source is the old-fashioned ships speaking tube. Sound is confined within the tube and can travel over 100 meters and hardly fade away at all.

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12.Problems in digital systems

We have already discussed frequency response and signal-to-noise ratio. Distortion in digital systems is very closely linked to noise. Although the distortion is at a very low level in practical systems, digital distortion is highly offensive to the ear. The cure for digital distortion is dither. Dither is a very low level noise signal that is added to the analog signal being digitized. This might seem like a crazy idea, to add noise to the signal. But the problem is that digital distortion is correlated to the signal, and the ear easily picks up on that. Adding noise randomises the distortion so that the noise is smooth and continues. The ear is happy to ignore that. And in a 16-bit system the noise is at a very low level anyway.

Digital signal demonstrating low jitter Another problem in digital systems is jitter. Jitter is caused when a digital signal is transmitted at an uneven rate. This can occur because of faulty, poorly-designed or cheap components. Or by long cable paths and physical problems in the system. Jitter translates as noise in the output. Sometimes a lot of noise. Fortunately, jitter can be completely cured by reclocking the signal so that it is even and regular. For practical purposes, it should be said that you will rarely come across significant problems caused by jitter using properly designed equipment. It is something that is worth knowing about however. One of the features of digital audio is that it uses zeros and ones to transmit and store the signal. So for instance a zero could be represented by a low voltage, a one by a high voltage. It would be very easy to distinguish the difference between a zero and a one. But sometimes there can be excessive noise or interference in the transmission or storage system and zeros and ones can be confused, causing errors. Digital systems therefore are able to detect errors and compensate for them. The first thing to do is to detect whether an error has occurred. The digital code used to store a signal has special data added to make this possible. For example if I devised a numerical code to represent letters of the alphabet, and I specified that only even numbers would be used, if you 29

received a message from me and some of the numbers were odd, you would know that there had been an error. When a digital system detects an error, it will attempt to correct it. Error correction means that the data is as good as it was before and the defect is completely inaudible. This is done by storing extra redundant data that can be used if some of the signal data is damaged. So if there is a scratch on a compact disc, the error handling system will recognize that the data is damaged and will look for the redundant data to replace it. Sometimes however the extent of the damage is too great to do this, or the redundant data is damaged as well. In this case error concealment will take place. The system will make an intelligent guess as to what the data would have been. Audio signals are somewhat predictable, so this is easily possible. The result however will be a slight degradation that might be audible. If the worst comes to the worst and the data cannot be corrected or concealed, the system will mute. At least this is what is supposed to happen. Digital glitches can be very high in level and be unpleasant to the ear and even damage loudspeakers. Clearly though not all equipment that is meant to mute on coming across seriously damaged data actually does that. Error correction and concealment takes place in compact disk, DVD, digital radio and television. It is also used on digital tape, but that isnt found so much these days. In a hard disk, CD-ROM and DVD-ROM a more powerful error correction system is used so that in normal operation the data that comes off the disk is perfect. Banks and financial institutions, that use exactly the same storage systems, would be rather less than happy if this were not so. However, where there is a problem that is beyond the error correction systems ability to cope, then the data is often irretrievably damaged.

Latency
All digital systems exhibit the phenomenon of latency where there is a delay between input and output on even the shortest signal paths. In a large analog mixing console there may be literally a kilometre of cable inside, but a signal takes for all practical purposes no time at all to go all the way through the console from input to output. This is so even in a large analog studio complex. Analog audio has no latency. However in a digital system it takes time to convert from analog to digital, and time to convert back again. So even the shortest signal path has a latency of at least a couple of milliseconds. This is generally not audible, although care has to be taken not to mix a signal with a delayed version of itself, or phase cancellation will take place. Where more processing is involved, the latency will be longer. The worst example would be a computer-based recording system where the signal is processed by the computers main processor. Processors such as this are designed for general data and are not optimised for signals. Also, the computer has to apply its attention to monitoring the keyboard and mouse for input, driving the display and other tasks. In this case the latency could be well into the tens of milliseconds, which is noticeable and sometimes off-putting.

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Clocking
An analog signal starts, ends, and goes its own sweet way in between. It has no external reference to time. An analog recording depends absolutely on the mechanism of the recorder running at a precisely controlled speed during recording and playback. There is no information in the recording to monitor or correct the speed. Digital systems on the other hand are sampled typically 44,100 times per second. This therefore binds digital signals to time references. A single signal on its own can happily work to its own time reference, which will simple be the period between samples. However, when two digital signals are mixed, there is a problem. It is impossible to have two digital clocks that run at the same speed. So one signal might be running a tiny bit faster than 44.1 kHz, the other a tiny bit slower. So if the two signals were mixed by simply adding up the numbers they contain, all might start off well, but sooner or later one of the signals will have to skip a sample to keep pace. This will cause a click, so clearly it is undesirable. A single signal contains its own clock. So if you wanted to copy a signal from one machine to another, then the machine you are copying onto can be set to external clock and it will derive its own clock reference from the incoming signal and run at the same speed. Often this happens automatically so the operator is unaware of it. A two-machine system can run perfectly well in this way. But as soon as you add a third digital device, which might be a mixer or processor and not a recorder, then it becomes difficult deciding which should be the clock master and which devices should synchronize to the master. So in larger digital systems it is common to provide a master clock, which is an independent unit. Everything in the entire system will use this clock source, so everything is sure to run at the same rate.

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