Sound Engineer

Professional Audio

2007-06-30T06:24:00.000-07:00

Professional audio, also 'pro audio', can be used a term to refer to both a type of audio equipment as well as a type of audio engineering application.

Professional audio equipment can be used to describe any audio equipment used or marketed for use as a sound application by or for a professional or professional purpose. This includes, but is not limited to, loudspeakers, microphones, Mixing consoles, amplifiers, recording and playback devices such as dat or turntables, and in some cases telephony devices. Pro Audio equipment typically carries an implied elevation of manufacturing quality and features compared to regular or consumer level audio equipment (as is common with other types of professional equipment.)

Professional audio application is commonly used to refer to professional audio engineering and operations, which can include but is not limited to broadcasting radio, audio mastering, sound reinforcement such as a concert, DJ performances, Audio Sampling , public address, surround sound movie theatres, and in some cases piped music application.

Both terms imply involvement of audio engineering at an industrial(occupational) level as opposed to a personal level. For example, a regular personal use microphone such as one in a mobile phone would have a very limited dynamic range focused on speech, whereas a pro audio microphone would have a much wider dynamic range to capture quiet whispers or loud musical instruments. A regular loudspeaker for home use may handle 100 watts rms at a given signal-to-noise ratio, whereas a pro audio loudspeaker such as one used for concert venues may handle 1000 watts rms or more, or a studio use speaker may operate at a significantly more efficient signal-to-noise rating at the same 100 watts as the home speaker.

Specifications alone do not inherently include or exclude equipment for consideration as professional audio level, but are used by most publications and documentation as a starting point of reference.

Soundproofing

2007-06-30T06:19:00.000-07:00

Soundproofing is any means of reducing the intensity of sound with respect to a specified source and receptor. There are several basic approaches to reducing sound: increasing the distance between source and receiver, using noise barriers to block or absorb the energy of the sound waves, using damping structures such as sound baffles, or using active antinoise sound generators.

Soundproofing affects sound in two different ways: noise reduction and noise absorption. Noise reduction simply blocks the passage of sound waves through the use of distance and intervening objects in the sound path. Noise absorption, on the other hand, operates by transforming the sound wave. Noise absorption involves suppressing echoes, reverberation, resonance and reflection. The damping characteristics of the materials it is made out of are important in noise absorption.

Contents [hide]
1 Distance
2 Damping
3 Room Within A Room
4 Noise cancellation
5 Noise barriers as exterior soundproofing
6 See also
7 References
8 External link

[edit] Distance
The use of distance to dissipate sound is straightforward. The energy density of sound waves decrease as they spread out, so that increasing the distance between the receiver and source results in a progressively lesser intensity of sound at the receiver. In a normal three dimensional setting, the intensity of sound waves will be attenuated according to the inverse square of the distance from the source. Using mass to absorb sound is also quite straightforward, with part of the sound energy being used to vibrate the mass of the intervening object, rather than being transmitted. When this mass consists of air the extra dissipation on top of the distance effect is only significant for typically more than 1000 meters, depending also on the weather and reflections from the soil.[1]

[edit] Damping
Damping is the process by which sonic vibrations are converted into heat over time and distance. This can be achieved in several ways. For example, use of a material such as lead that is both heavy and soft, with the softness allowing it to damp the noise rather than allowing transmission. Making a sound wave transfer through different layers of material with different densities also assists in noise damping. This is the reason why open-celled foam is a good sound damper; the sound waves are forced to travel through multiple foam cells and their cell walls as sound travels through the foam medium. Improperly done, however, structural compliance can make things worse, enabling resonance. This process is analogous to a string holding wind-chimes: the string helps the chimes ring by isolating the vibration instead of damping it. Foam tapes may therefore be undependable in a soundproofing protocol.

[edit] Room Within A Room
A Room Within A Room (RWAR) is one method of isolating sound and stopping it transmitting to the outside world where it may be undesirable.

Most vibration / sound transfer from a room to the outside occurs through mechanical means. The vibration passes directly through the brick, woodwork and other solid structural elements. When it meets with an efficient sound board such as a wall, ceiling, floor or window, the vibration is amplified and heard in the second space. A mechanical transmission is much faster, more efficient and may be more readily amplified than an airborne transmission of the same initial strength.

The use of acoustic foams and other absorbent means are useless against this transmitted vibration. The user is required to break the connection between the room that contains the noise source and the outside world. This is called acoustic de-coupling. Ideal de-coupling involves eliminating vibration transfer in both solid materials and in the air, so air-flow into the room is often controlled. This has safety implications, for example proper ventilation must be assured and gas heaters cannot be used inside de-coupled space.

There are very successful professional products and methods available but such a construction is definitely within the reach of competent do-it-yourselfers.[2] Costs vary depending on the individual space, but it is clear that by doing it oneself, an individual can approximate the same result as professionals and save a substantial amount of money

[edit] Noise cancellation
Noise cancellation generators for active noise control are a relatively modern innovation. A microphone is used to pick up the sound that is then analyzed by a computer; then, sound waves with opposite polarity (not phase) are output through a speaker, causing destructive interference and cancelling much of the noise.

[edit] Noise barriers as exterior soundproofing
Main article: Noise barrier
Since the early 1970s it has become common practice in the United States (followed later by many other industrialized countries) to engineer noise barriers along major highways to protect adjacent residents from intruding roadway noise. The technology exists to predict accurately the optimum geometry for the noise barrier design. Noise barriers may be constructed of masonry, earth or a combination thereof. One of the earliest noise barrier designs was in Arlington, Virginia adjacent to Interstate 66, stemming from interests expressed by the Arlington Coalition on Transportation. Possibly the earliest scientifically designed and published noise barrier construction was in Los Altos, California in 1970.

Studio Monitor

2007-06-30T06:17:00.000-07:00

The goal of most studio monitors is to produce a flat frequency response and a truthful representation of the source material. Unlike consumer speakers, which are often designed to make all audio material sound pleasing to the ear, the studio monitor generally attempts to paint an accurate audio image of the material with no unnatural emphasis or de-emphasis of particular frequencies. This is what it means when a monitor is said to be "flat". and "uncolored" or "transparent". Sound engineers usually require accurate sound reproduction from their speakers especially during audio mixing and mastering. This enables the engineer to mix a track that will sound consistent on high-end audio, low quality radios, in a club, in a car stereo or a home stereo. Some engineers, however, prefer to work with monitors that are far from accurate because they reflect the type of systems that end-users will be listening through.

Studio monitors can be active (they include one or more internal power amplifier(s)), or passive (they need an external power amplifier). Active models are usually bi-amplified.

Contents [hide]
1 History
2 Application
3 Home Audio versus Pro Audio
4 See also
5 References

[edit] History
In the early days of the recording industry, studio monitors were used primarily to check for noise interference and obvious technical problems rather than for making artistic evaluations of the performance and recording. Musicians were recorded live and the producer judged the performance on this basis, relying on simple tried-and-true microphone techniques to ensure that it had been adequately captured; playback through monitors was used simply to check that no obvious technical flaws had spoiled the original recording.

As a result, early monitors tended to be crude. The state of the art loudspeakers of the era were massive horn-loaded systems and were consequently used almost exclusively in cinemas. High-end loudspeaker design grew out of the demands of the motion picture industry and most of the early loudspeaker pioneers worked in Los Angeles where they attempted to solve the problems of cinema sound. Designing monitors for recording studios wasn’t a major priority.

The first high-quality loudspeaker developed expressly as a studio monitor was the Altec 604 Duplex in 1944. This innovative driver has historically been regarded as the work of James B. Lansing who’d previously supplied the drivers for the Shearer Horn in 1936, a speaker that had rapidly become the industry standard in motion-picture sound. He’d also designed the smaller Iconic and this was widely employed at the time as a motion-picture studio monitor. The 604 was a relatively compact coaxial design and within a few years it became the industry standard in the United States, a position it maintained in its various incarnations (the 604 went through eleven model-changes) over the next 25 years. It was common in US studios throughout the 1950s and 60s and remained in continuous production until 1998. In the UK, Tannoy introduced its own coaxial design, the Dual Concentric, and this assumed the same reference role in Europe as the Altec 604 held in the USA.

Monitor usage in the industry was highly conservative, with almost monopolistic reliance on industry “standards”, in spite of the sonic failings of these aging designs. The Altec 604 had a notoriously ragged frequency response but almost all U.S studios continued to use it because virtually every producer and engineer knew its sound intimately and were practiced at listening through its sonic limitations. Recording through unfamiliar monitors, no matter how technically advanced, was hazardous because engineers not used to their sonic signature could make poor production decisions and it was financially unviable to give production staff expensive studio time to familiarize themselves with new monitors. As a result, pretty well every U.S studio had a set of 604’s and every European studio a Tannoy Dual Concentric or two.

However, in 1959, at the height of its industry dominance, Altec made the mistake of replacing the 604 with the 605A Duplex, a design widely regarded as inferior to its predecessor. There was a backlash from some record companies and studios and this allowed Altec’s competitor, JBL (a company originally started by 604 designer James B. Lansing), to make inroads into the pro monitor market. Capitol Records quickly replaced their Altecs with JBL D50 Monitors and a few years later their UK affiliate, EMI, also made the move to JBL’s. Although Altec re-introduced the 604 as the "E" version Super Duplex in response to the criticism, they now had a major industry rival to contend with. Over the next decade most of the developments in studio monitor design originated from JBL.

As recording became less and less “live” and multi-tracking and overdubbing became the norm, the studio monitor became far more crucial to the recording process. When there was no original performance outside what existed on the tape, the monitor became the touchstone of all engineering and production decisions. As a result, accuracy and transparency became paramount and the conservatism evident in the retention of the 604 as the standard for over twenty years began to give way to fresh technological development. Despite this, the 604 continued to be widely used - mainly because many engineers and producers were so familiar with their sonic signature that they were reluctant to change. It wasn’t until 1975 that JBL overtook Altec as the monitor of choice for most studios.

In the late 1960’s JBL introduced two monitors which helped secure them preeminence in the industry. The 4320 was a direct competitor to the Altec 604 but was a more accurate and powerful speaker and it quickly made inroads against the industry standard. However, it was the more compact 4310 that revolutionized monitoring by introducing the idea of close or “nearfield” monitoring. (The sound field very close to a sound source is called the "near-field". By "very close" is meant in the predominantly direct, rather than reflected, soundfield. A near-field speaker is a compact studio monitor designed for listening at close distances (3’-5’), so, in theory, the effects of poor room acoustics are greatly reduced.)

The 4310 was small enough to be placed on the recording console and listened to from much closer distances than the traditional large wall-(or “soffit”) mounted main monitors. As a result, studio-acoustic problems were minimized. Smaller studios found the 4310 ideal and that monitor and its successor, the 4311, became studio fixtures throughout the 1970’s. Ironically, the 4310 had been designed to replicate the sonic idiosyncracies of the Altec 604 but in a smaller package to cater for the technical needs of the time. The 4311 was so popular with professionals that JBL introduced a domestic version for the burgeoning home-audio market. This speaker, the JBL L-100, (or "Century") was a massive success and became the biggest-selling hi-fi speaker ever within a few years.

The major studios continued to use huge designs mounted on the wall which were able to produce prodigious SPL’s and amounts of bass. This trend reached its zenith with The Who’s employment in their studio of a dozen JBL 4350 monitors, each capable of 125 dB and containing two fifteen-inch woofers and a twelve-inch mid-bass driver. Most studios, however, also used more modest monitoring devices to check how recordings would sound through car speakers and cheap home systems. A favourite “grot-box” monitor employed in this way was the Auratone 5C, a crude single-driver device that gave a reasonable facsimile of typical lo-fi sound.

However, a backlash against the Behemoth Monitor was soon to take place. With the advent of Punk, New Wave, Indie, and Lo-Fi, a reaction to high-tech recording and large corporate-style studios set in and do-it-yourself recording methods became the vogue. Smaller, less expensive, recording studios needed smaller, less expensive monitors and the Yamaha NS-10, a design introduced in 1978 ironically for the home audio market, became the monitor of choice for many studios in the 1980’s. A variety of stories, probably apocryphal, abound about why the NS-10 assumed this role but it gradually became an industry adage that “if it sounds good on the NS-10 it’ll sound good on anything”. While its sound-quality has often been derided, even by those who monitor through it, the NS-10 continues in use to this day and many more successful recordings have been produced with its aid over the past twenty five years than with any other monitor.

By the mid-80’s the near-field monitor had become pre-eminent. The larger studios still had large main-monitors mounted in (or on) the wall but they were now mere supplements to the small monitors sitting on the meter-bridge and were viewed as prestige items mainly there to “impress the clients” and occasionally check for low-bass anomalies. Favourite large monitors of the time were the Westlake and Urei 813, designs based on a highly modified version of the almost ageless Altec 604. Fostex "Laboratory Series" monitors were to be found in the finest studios but with increasing costs of manufacture, they became rare. The once dominant JBL fell into disfavour as its designs were identified with 70’s excess. The new studio landscape was bare and stripped-down and the large three or four-way monitors were hardly in keeping with this philosophy.

Yamaha eventually discontinued the NS-10 due to the lack of availability of the wood pulp used in the woofer. Even so, old NS-10’s still dot the studio landscape and at present it seems to be the last of the old style Industry Standards. No single monitor has emerged to become the fixtures that the Altec 604, the JBL 4311, and the Yamaha NS-10 were in their day. It now seems that every producer and engineer has their personal favourite monitor and developments in recording and monitor design have enabled this trend to continue. Personal recording studios have accelerated the move towards customization and individuality as the need for common industry standards is lessened. Monitors have become more and more compact and portable so it’s now feasible for producers to take their personal monitor choices with them to different recording venues.

The main post-NS10 trend has been the almost universal acceptance of powered monitors where the speaker enclosure contains the driving amplifiers. The old style passive monitors required outboard power amplifiers to drive them as well as speaker wire to connect them. Powered monitors, by contrast, are much more convenient and streamlined single units which in addition have a number of technical advantages. The interface between speaker and amplifier is optimized, with greater control and precision, and advances in amplifier design have reduced the size and weight of the electronics significantly. The result has been that passive monitors are now only a sidelight to the powered market and are in danger of being phased out completely.

[edit] Application
The studio monitor may be the single most important piece of equipment in the recording studio. This stems from the fact that every aspect of the recording process can only be heard - and evaluated – by listening through a monitor. Without it, recording is just guesswork.

As a result, production staff need to trust their monitors. The less confidence they have that a monitor is telling them what they need to know to make a correct decision the more random and time-consuming the recording process becomes. If a monitor fails to distinguish between two very different microphones or conceals distortion in the recording chain, poor production decisions are likely to be made. Producers, engineers, and mixers consequently insist on monitors that are either highly accurate and transparent or ones that they’re intimately familiar with. In the latter case, they learn to listen through the flaws, tend to be conservative, and often stick to a specific monitor even when it becomes technically outdated because changing to a new one requires a period of adjustment.

There are a variety of approaches to monitor choice and much “subjective” disagreement hinges on which of these methods production staff adopt. Some rigorously pursue accuracy and transparency, on the grounds that only an accurate monitor can tell the unvarnished truth and enable them to make completely trustworthy decisions. Others may insist that the systems used by the consumers are themselves far from accurate and that it’s more important that monitors reflect what these lo-fi stereos are doing to the recordings rather than pretend that it's being reproduced with perfect accuracy. Others believe that monitors should exaggerate recording flaws and make them work harder to make them sound good. Many more stick to monitors they know well, even though they’re fully aware of their limitations. Arguments about the “right” monitor to use still rage on, although there’s a growing acceptance that “what works for you” is the best adage to follow when choosing monitors. Producers and engineers have differing goals; some are producing high-quality recordings for expensive home systems while others are catering mainly for boomboxes and car radios. Their choices of monitor invariably take account of these facts.

Monitor use is thus in most cases geared to preparing the recording for end use: for making it sound good to the consumer. As a result, while accuracy has traditionally been considered to be the sine qua non of the studio monitor, in practice producers, engineers, and mixers are less concerned with accuracy per se than with how well the monitor translates, i.e., how well recordings made with its aid sound through a variety of playback systems, ranging from audiophile esoterica to boom-boxes. The link between accuracy and translatability is unclear, and the fact that the three most popular monitors of all time, the Altec 604, the JBL 4311, and the Yamaha NS-10, were all far from accurate may not be entirely coincidental.

[edit] Home Audio versus Pro Audio
While no rigid distinction exists between speakers intended for consumer use and those designed as studio monitors, there has been an increasing gulf between the two markets in practice. Whereas in the 1970’s the JBL 4311’s domestic equivalent, the L-100, was used in a large number of homes, and the Yamaha NS-10 also served both domestically and professionally during the 1980’s, there are no present-day equivalents. Professional companies such as Fostex, Genelec, Adam Audio, KRK Systems, Mackie, Klein and Hummel, Quested, PMC, and M & K sell almost exclusively to the professional monitor market while most of the consumer audio manufacturers confine themselves to supplying speakers for the home. Even companies that straddle both worlds, like Tannoy, Focal/JM Labs, Dynaudio, and JBL, tend to clearly separate their pro and consumer lines.

There are a number of reasons for this:

Domestic speakers are generally less rugged and unable to cope with the often extreme conditions encountered in the recording studio;
pro monitors are generally designed to be listened to from much shorter distances than home speakers;
pro monitors are generally powered while domestic speakers are almost always passive;
pro monitors are voiced to be less flattering to the source than domestic speakers are.
An illuminating indication of the difference between the two markets is the fact that the observation that “it makes everything sound great” is seen as a criticism in the studio monitor world! Monitors are selected because they ostensibly don’t flatter the material played through them and offer a “warts and all” presentation that makes it less likely for producers/engineers to approve unsatisfactory productions. Monitors are intended to err, if at all, on the side of harshness and aggressiveness rather than on papering over recording flaws whereas domestic speakers are often designed to make even mediocre material sound palatable.

For some reason, domestic speakers haven’t followed the professional move towards the active and powered. In audiophile circles, this is probably due to the fact that powered speakers tend to emphasize sonic qualities they find uncongenial, as well as a desire to select separate components rather than simplify the audio chain. For this reason, the seemingly inevitable move to domestic powered speakers is more likely to come at the lower end of the consumer market.

Stereo Imaging

2007-06-30T06:16:00.000-07:00

Stereo imaging is the audio jargon term used for that aspect of sound recording and reproduction concerning spatial locations of the performers, both laterally and in depth. An image is 'good' if the performers can be effortlessly located; 'bad' if there is no hope of doing so. A well-made stereo recording, properly reproduced, can provide good imaging within the front quadrant; a well-made Ambisonic recording, properly reproduced, can offer good imaging all around the listener and even including height information.

For many listeners, good imaging adds markedly to the pleasure of reproduced music. One may speculate that this is due to the evolutionary importance to humans of knowing where sounds are coming from, and that imaging may therefore be more important than some purely esthetic considerations in satisfying the listener. Listeners do exist who have difficulty paying attention to the musical content of a recording if the imaging is not good.

The quality of the imaging arriving at the listener's ear depends on numerous factors, of which the most important is the original "miking", that is, the choice and arrangement of the recording microphones (where "choice" refers here not to the brands chosen, but to the size and shape of the microphone diaphragms, and "arrangement" refers to microphone placement and orientation relative to other microphones). This is partly because miking simply affects imaging more than any other factor, and because, if the miking spoils the imaging, nothing later in the chain can recover it.

If miking is done well, then quality of imaging can be used to evaluate components in the record/playback chain (remembering that once the imaging is destroyed, it cannot be recovered).

It is worth noting that only a handful of recordings are miked for optimal imaging, and what usually passes for stereo, while being two-channel recording, is not true stereo because the imaging information is incoherent.

Imaging is usually thought of in the context of recording with two or more channels, though single-channel recording may convey depth information convincingly, and at least one expert thinks it can convey information about lateral placement, also.

SAE Institute

2007-06-30T06:12:00.000-07:00

SAE Institute
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The SAE Institute (SAE in short, formerly also known as the School of Audio Engineering and the SAE Technology College) is a private college founded in 1976 by Sound Engineer/Record Producer Tom Misner. The first school was opened 1977 in Sydney, (Australia). SAE Institute offers courses in Audio engineering, 3D Animation, Multimedia and Digital Filmmaking. It is the largest college worldwide in these fields, and currently has campuses / facilities in 46 cities in 21 countries. Since 1998 SAE is additionally offering full university degrees through its global partnership with Middlesex University, London, United Kingdom, and in Australia already since 1996 through its co-operation with Southern Cross University.

[edit] History
1976 The School of Audio Engineering (SAE) is set up in June 1976 by Tom Misner[1] who, in so doing, develops the first practical/theoretical curriculum.
1977 The first 9 month course commences in February, in Byron Bay Australia with a Sony 4-track tape recorder and a custom made 12 channel mixing console.
1978 SAE Melbourne is established with a small 8 track studio and some editing tape machines.
1980 SAE Brisbane is established and SAE's first commercial studio Central Recorder is opened in Sydney.
1981 SAE Sydney commences with the acceptance of overseas students and gains the first form of government recognition - a public service grading.
1982 SAE Adelaide commences operation early in the year. SAE Perth is established in August.
1984 SAE Coffs Harbour (Australia) is set up for one year only to conduct a course on behalf of the local television and radio stations, but the college stays open for a further year.
1985 Resulting from a business trip to London, Tom Misner establishs SAE London, the first overseas college. Operation commences in March.
1986 SAE Munich, the first foreign language SAE school, opens in Germany. SAE Frankfurt commences late in the year.
1987 SAE Vienna the first college in Austria is opened in February. The first custom-designed program is conducted by SAE Germany for radio station 'Radio Free Europe'.
1988 SAE Berlin the third German college is opened late in the year with special assistance from the German government. A new course 'Live Sound Engineering' is offered by SAE.
1989 SAE introduces the 6 months part-time 'Studio Sound Certificate' course.
1990 SAE Auckland is established and is granted full government approval (New Zealand Qualifications Authority). SAE introduces the 'DJ and Sampling' course in London. SAE Glasgow commences operation late in the year.
1991 SAE Amsterdam, the new European operational head office is opened. One of the largest orders of Neve audio editing consoles ever placed (11 VR series consoles). SAE Singapore commences in September as the first audio college in Asia.
1992 The audio engineering programme is extended to 15 months part-time in all SAE colleges. SAE Kuala Lumpur opens in October.
1993 SAE Paris opens. Partial government approval and funding for students is given to SAE by the Government of France. SAE Hamburg opens and commences operation as the first SAE college to teach extensive digital practice. The studios are based upon the Soundtracs Jade console, Sony APR multitrack and ProTools III.
1994 SAE Kuala Lumpur gains government approval (Ministry of Education). The first SAE Book (Practical Studio Techniques by Tom Misner) is published. SAE forms an official link with the Australian Southern Cross University to offer a joint degree program (BA Music Production). The School of Audio Engineering changes its name to SAE Technology College. Tom Misner opens the only new large commercial recording studio in Australia to be built in the 90's, Mirage Studios.
1995 The first SAE-ProSchool is established in London teaching the Digidesign ProTools system. SAE Stockholm (Sweden) commences. SAE Zürich has now been established in the Technopark industrial complex. SAE Hobart, the 6th college in Australia, opens. SAE Cologne, the fifth German College, opens later in that year and offers both the audio and multimedia programmes. SAE Singapore receives government approval and is able to accept overseas students.
1996 The multimedia program is expanded to Zürich and Singapore. The first full university degree programme is launched by SAE Sydney with the co-operation of the Southern Cross University. SAE Frankfurt now offers the first live sound program in Germany. SAE Milano opens in Italy.
1997 Expansion of several SAE campuses. SAE forms the SAE Entertainment Company for professional production of CD ROM, CD extra, CD audio and internet homepages.
1998 SAE New York City is licensed. SAE Athens, Greece opens late in the year. SAE enters into a collaborative arrangement with Middlesex University, England and the first BA (Hons) degree programmes are run at the London, Munich and Sydney campuses.
1999 SAE Nashville starts operation. SAE purchases recording facility Studio 301 in Australia. SAE Munich starts the Digital Film Programme in November 1999, with Cologne, Hamburg and Vienna to follow in spring.
2000 SAE Munich starts Digital Film Arts Degree. SAE Hamburg starts the Digital Film Program. Four franchise schools are established in India. SAE Frankfurt, Amsterdam, Stuttgart, and Berlin areexpanded.
2001 sees the opening of SAE Miami, Liverpool and Madrid. Tom Misner purchases the largest recording studio in Germany, which is now part of the Studios 301 Group.
2002 SAE Adelaide and SAE Perth turn 20. The new Digital Film Making Program is starting in Australia and Europe. SAE Thiruvananthapuram, India, commences operation. SAE Berlin and SAE Athens are approved as Degree Centres by Middlesex University, England. SAE Bangkok starts operation.
2003 The SAE Alumni Association is founded. SAE Brussels, Belgium, SAE SAE Institute Bangkok and SAE Yangon, Myanmar, are opened. SAE Berlin is offering the first Bachelor courses. The new headquarters in Byron Bay, Australia opens.
2004 SAE Munich, Amsterdam, Melbourne, Stuttgart and Hamburg all move to new, bigger and improved premises. SAE Leipzig opens. SAE Barcelona opens. SAE acquires QANTM, Australia's leading production, new media and training company.
2005 Tom Misner takes over console maker AMS Neve. SAE Dubai, first college in the Middle East, opens. SAE Los Angeles, fourth college in the US, opens. First SAE Alumni worldwide conference is held in Frankfurt.
2006 SAE opens its second Middle East campus in Kuwait.

[edit] External links
SAE Institute for Audio Engineering, Filmmaking, Animation and Multimedia & Web Design Courses
SAE Institute Byron Bay - International Headquarters offering Audio Engineering, Filmmaking and Multimedia & Web Design Courses
SAE Institute UK
SAE Institute Germany
SAE Institute Austria
SAE Institute Italia
SAE Institute France
SAE Institute Spain
SAE Institute Bangkok
SAE Institute Malaysia
SAE Institute Kuwait
SAE Institute Singapore
SAE Institute The Netherlands

Microphone

2007-06-30T06:07:00.000-07:00

A microphone, sometimes referred to as a mike or mic (both IPA pronunciation: [maɪk]), is an acoustic to electric transducer or sensor that converts sound into an electrical signal.

A Neumann U87 capacitor microphoneMicrophones are used in many applications such as telephones, tape recorders, hearing aids, motion picture production, live and recorded audio engineering, in radio and television broadcasting and in computers for recording voice, VoIP, and for non-acoustic purposes such as ultrasonic checking.

Contents [hide]
1 History
2 Principle of operation
3 Microphone varieties
3.1 Condenser, capacitor or electrostatic microphones
3.1.1 Technology
3.1.1.1 DC-biased microphone operating principle
3.1.1.2 RF condenser microphone operating principle
3.1.2 Usage
3.1.3 Electret condenser microphones
3.2 Dynamic microphones
3.2.1 Moving coil microphones
3.2.1.1 Technology
3.2.2 Ribbon microphones
3.3 Carbon microphones
3.4 Piezo microphones
3.4.1 Technology
3.4.2 Usage
3.5 Laser microphones
3.5.1 Usage
3.6 Liquid microphones
3.6.1 Technology
3.6.2 Usage
3.7 MEMS microphones
3.8 Speakers as microphones
4 Capsule design and directivity
5 Microphone polar patterns
6 Application-specific microphone designs
7 Connectivity
7.1 Connectors
7.2 Impedance matching
8 Measurements and specifications
9 Measurement microphones
9.1 Microphone calibration techniques
9.1.1 Pistonphone apparatus
9.1.2 Reciprocal method
10 Microphone array and array microphones
11 See also
12 Microphone manufacturers
13 External links
14 References

[edit] History
Several early inventors built primitive microphones (then called transmitters) prior to Alexander Bell, but the first commercially practical microphone was the carbon microphone conceived in October 1876 by Thomas Edison. Many early developments in microphone design took place at Bell Laboratories. See also Timeline of the telephone.

[edit] Principle of operation

An Oktava condenser microphone.A microphone is a device made to capture waves in air, water (hydrophone) or hard material and translate them into an electrical signal. The most common method is via a thin membrane producing some proportional electrical signal. Most microphones in use today for audio use electromagnetic generation (dynamic microphones), capacitance change (condenser microphones) or piezoelectric generation to produce the signal from mechanical vibration. The piezoelectric microphone is now largely obsolete. However, piezoelectric pickups are still the most common device for amplifying acoustic guitars, usually placed under the guitar's saddle or embedded in the bridge.

[edit] Microphone varieties

[edit] Condenser, capacitor or electrostatic microphones

Inside the Oktava 319 condenser microphone.
[edit] Technology
In a condenser microphone, also known as a capacitor microphone, the diaphragm acts as one plate of a capacitor, and the vibrations produce changes in the distance between the plates.

There are two methods of extracting an audio output from the transducer thus formed. They are known as DC biased and RF (or HF) condenser microphones.

[edit] DC-biased microphone operating principle
The plates are biased with a fixed charge (Q). The voltage maintained across the capacitor plates changes with the vibrations in the air, according to the capacitance equation:

where Q = charge in coulombs, C = capacitance in farads and V = potential difference in volts. The capacitance of the plates is inversely proportional to the distance between them for a parallel-plate capacitor. (See capacitance for details.)

A nearly constant charge is maintained on the capacitor. As the capacitance changes, the charge across the capacitor does change very slightly, but at audible frequencies it is sensibly constant. The capacitance of the capsule and the value of the bias resistor form a filter which is highpass for the audio signal, and lowpass for the bias voltage. Note that the time constant of a RC circuit equals the product of the resistance and capacitance.

Within the time-frame of the capacitance change (on the order of 100 μs), the charge thus appears practically constant and the voltage across the capacitor adjusts itself instantaneously to reflect the change in capacitance. The voltage across the capacitor varies above and below the bias voltage. The voltage difference between the bias and the capacitor is seen across the series resistor. The voltage across the resistor is amplified for performance or recording.

[edit] RF condenser microphone operating principle
In a DC-biased condenser microphone, a high capsule polarisation voltage is necessary. In contrast, RF condenser microphones use a comparatively low RF voltage, generated by a low-noise oscillator. The oscillator is frequency modulated by the capacitance changes produced by the sound waves moving the capsule diaphragm. Demodulation yields a low-noise audio frequency signal with a very low source impedance. This technique achieves better low frequency response - in fact it will theoretically operate down to DC.

The RF biasing process results in a lower electrical impedance capsule, a useful byproduct of which is that RF condenser microphones can be operated in damp weather conditions which would effectively short out a DC biased microphone. The Sennheiser "MKH" series of microphones use the RF biased technique.

[edit] Usage
Condenser microphones span the range from cheap throw-aways to high-fidelity quality instruments. They generally produce a high-quality audio signal and are now the popular choice in laboratory and studio recording applications. They require a power source, provided either from microphone inputs as phantom power or from a small battery. Professional microphones often sport an external power supply for reasons of quality perception. Power is necessary for establishing the capacitor plate voltage, and is also needed for internal amplification of the signal to a useful output level. Condenser microphones are also available with two diaphragms, the signals from which can be electrically connected such as to provide a range of polar patterns (see below), such as cardioid, omnidirectional and figure-eight. It is also possible to vary the pattern smoothly with some microphones, for example the Røde NT2000.

[edit] Electret condenser microphones
Main article: Electret microphone
An electret microphone is a relatively new type of capacitor microphone invented at Bell laboratories in 1962 by Gerhard Sessler and Jim West[1]. An electret is a dielectric material that has been permanently electrically charged or polarized. The name comes from electrostatic and magnet; a static charge is embedded in an electret by alignment of the static charges in the material, much the way a magnet is made by aligning the magnetic domains in a piece of iron. They are used in many applications, from high-quality recording and lavalier use to built-in microphones in small sound recording devices and telephones. Though electret microphones were once low-cost and considered low quality, the best ones can now rival capacitor microphones in every respect and can even offer the long-term stability and ultra-flat response needed for a measuring microphone. Unlike other capacitor microphones, they require no polarizing voltage, but normally contain an integrated preamplifier which does require power (often incorrectly called polarizing power or bias). This preamp is frequently phantom powered in sound reinforcement and studio applications. While few electret microphones rival the best DC-polarized units in terms of noise level, this is not due to any inherent limitation of the electret. Rather, mass production techniques needed to produce electrets cheaply don't lend themselves to the precision needed to produce the highest quality microphones.

[edit] Dynamic microphones
Dynamic microphones work via electromagnetic induction. They are robust, relatively inexpensive and resistant to moisture, and for this reason they are widely used on-stage by singers. Dynamic microphones are velocity receivers. There are two basic types: the moving coil microphone and the ribbon microphone.

[edit] Moving coil microphones

The Shure SM57 and Beta 57A dynamic microphones
[edit] Technology
A small movable induction coil, positioned in the magnetic field of a permanent magnet, is attached to the diaphragm. When sound enters through the windscreen of the microphone, the sound wave moves the diaphragm. When the diaphragm vibrates, the coil moves in the magnetic field, producing a varying current in the coil through electromagnetic induction. A single dynamic membrane will not respond linearly to all audio frequencies. Some microphones for this reason utilize multiple membranes for the different parts of the audio spectrum and then combine the resulting signals. Combining the multiple signals correctly is difficult and designs that do this are rare and tend to be expensive. There are on the other hand several designs that are more specifically aimed towards isolated parts of the audio spectrum. AKG D112 is for example designed for bass content rather than treble. In audio engineering several kinds of microphones are often used at the same time to get the best result.

The dynamic principle is exactly the same as in a loudspeaker, only reversed.

[edit] Ribbon microphones
Main article: Ribbon microphone
In ribbon microphones a thin, usually corrugated metal ribbon is suspended in a magnetic field. The ribbon is electrically connected to the microphone's output, and its vibration within the magnetic field generates the electrical signal. Ribbon microphones are similar to moving coil microphones in the sense that both produce sound by means of magnetic induction. Basic ribbon microphones detect sound in a bidirectional (also called figure-eight) pattern because the ribbon, which is open to sound both front and back, responds to the pressure gradient rather than the sound pressure. Though the symmetrical front and rear pickup can be a nuisance in normal stereo recording, the high side rejection can be used to advantage by positioning a ribbon microphone horizontally, for example above cymbals, so that the rear lobe picks up only sound from the cymbals. Other directional patterns are produced by enclosing one side of the ribbon in an acoustic trap or baffle, allowing sound to reach only one side. Ribbon microphones give very high quality sound reproduction, and were once valued for this reason, but a good low-frequency response can be obtained only if the ribbon is suspended very loosely, and this makes them fragile. Protective wind screens can reduce the danger of damaging the ribbon, but will somewhat reduce the treble response.

Ribbon microphones don't require phantom power; in fact, this voltage can damage these microphones.

[edit] Carbon microphones
Main article: Carbon microphone
A carbon microphone, formerly used in telephone handsets, is a capsule containing carbon granules pressed between two metal plates. A voltage is applied across the metal plates, causing a small current to flow through the carbon. One of the plates, the diaphragm, vibrates in sympathy with incident sound waves, applying a varying pressure to the carbon. The changing pressure deforms the granules, causing the contact area between each pair of adjacent granules to change, and this causes the electrical resistance of the mass of granules to change. The changes in resistance cause a corresponding change in the voltage across the two plates, and hence in the current flowing through the microphone, producing the electrical signal. Carbon microphones were once commonly used in telephones; they have extremely low-quality sound reproduction and a very limited frequency response range, but are very robust devices.

Unlike other microphone types, the carbon microphone can also be used as a type of amplifier, using a small amount of sound energy to produce a larger amount of electrical energy. Carbon microphones found use as early telephone repeaters, making long distance phone calls possible in the era before vacuum tubes. These repeaters worked by mechanically coupling a magnetic telephone receiver to a carbon microphone: the faint signal from the receiver was transferred to the microphone, with a resulting stronger electrical signal to send down the line. (One illustration of this amplifier effect was the oscillation caused by feedback, resulting in an audible squeal from the old "candlestick" telephone if its earphone was placed near the carbon microphone.)

[edit] Piezo microphones

[edit] Technology
A piezo microphone uses the phenomenon of piezoelectricity—the ability of some materials to produce a voltage when subjected to pressure—to convert vibrations into an electrical signal. An example of this is Rochelle salt (potassium sodium tartrate), which is a piezoelectric crystal that works as a transducer, both as a microphone and as a slimline loudspeaker component.

[edit] Usage
Piezo transducers are often used as contact microphones to amplify sound from acoustic musical instruments, or to record sounds in unusual environments (underwater, for instance). Saddle mounted pickups on acoustic guitars are generally piezos that are mechanically connected to the strings through the saddle. This type of microphone is not to be confused with magnetic coil pickups commonly visible on typical electric guitars.

[edit] Laser microphones

[edit] Usage
Laser microphones are new, very rare and expensive, and are most commonly portrayed in movies as spying devices.

[edit] Liquid microphones

[edit] Technology
Early microphones did not produce intelligible speech, until Alexander Graham Bell made a set of improvements. Bell’s liquid transmitter consisted of a metal cup filled with dilute sulfuric acid. A sound wave caused the diaphragm to move, forcing a brass tube to move up and down in the liquid. The electrical resistance between the wire and the cup was then inversely proportional to the length of wire submerged. Elisha Gray filed a patent for a version using a needle instead of the brass tube. Other minor variations and improvements were made to the liquid microphone by Majoranna, Chambers, Vanni, Sykes, and Elisha Gray, and one version was even patented by Reginald Fessenden in 1903.

[edit] Usage
These were the first working microphones, but they were not practical for commercial application and are utterly obsolete now. It was with a liquid transmitter that the famous first phone conversation between Bell and Watson took place. Other inventors soon devised superior devices.

[edit] MEMS microphones
The MEMS microphone is also called a microphone chip or silicon microphone. The pressure-sensitive diaphragm is etched directly on a silicon chip by MEMS techniques[citation needed], and is usually accompanied with integrated preamplifier. Most MEMS microphones are modern embodiments of the standard condenser microphone. Often MEMS mics have a built in ADC on the same CMOS chip making the chip a digital microphone and easily integrated into modern digital products. Major manufacturers using MEMS manufacturing for silicon microphones are Akustica (AKU200x), Infineon (SMM310 product), Knowles Electronics and Sonion MEMS.

[edit] Speakers as microphones
A loudspeaker, a transducer that turns an electrical signal into sound waves, is the functional opposite of a microphone. Since a conventional speaker is constructed much like a dynamic microphone (with a diaphragm, coil and magnet), speakers can actually work "in reverse" as microphones. The result, though, is a microphone with poor quality, limited frequency response (particularly at the high end), and poor sensitivity.

In practical use, speakers are sometimes used as microphones in such applications as intercoms or walkie-talkies, where high quality and sensitivity are not needed. However, there is at least one other novel application of this principle; using a medium-size woofer placed closely in front of a "kick" (bass drum) in a drum set to act as a microphone. This has been commercialized with the Yamaha "Subkick".[1]

[edit] Capsule design and directivity
The shape of the microphone defines its directivity. Inner elements are of major importance and concerns the structural shape of the capsule, outer elements may be the interference tube.

A pressure gradient microphone is a microphone in which both sides of the diaphragm are exposed to the incident sound and the microphone is therefore responsive to the pressure differential (gradient) between the two sides of the membrane. Sound incident parallel to the plane of the diaphragm produces no pressure differential, giving pressure-gradient microphones their characteristic figure-eight directional patterns.

The capsule of a pressure microphone however is closed on one side, which results in an omnidirectional pattern.

[edit] Microphone polar patterns
Common polar patterns for microphones (Microphone facing top of page in diagram, parallel to page):

Omnidirectional

Subcardioid

Cardioid

Supercardioid

Hypercardioid

Bi-directional

Shotgun

A microphone's directionality or polar pattern indicates how sensitive it is to sounds arriving at different angles about its central axis. The above polar patterns represent the locus of points that produce the same signal level output in the microphone if a given sound pressure level is generated from that point. How the physical body of the microphone is oriented relative to the diagrams depends on the microphone design. For large-membrane microphones such as in the Oktava (pictured above), the upward direction in the polar diagram is usually perpendicular to the microphone body, commonly known as "side fire". For small diaphragm microphones such as the Shure (also pictured above), it usually extends from the axis of the microphone commonly known as "end fire". Some microphone designs combine several principles in creating the desired polar pattern. This ranges from shielding (meaning diffraction/dissipation/absorption) by the housing itself to electronically combining dual membranes.

An omnidirectional microphone's response is generally considered to be a perfect sphere in three dimensions. In the real world, this is not the case. As with directional microphones, the polar pattern for an "omnidirectional" microphone is a function of frequency. The body of the microphone is not infinitely small and, as a consequence, it tends to get in its own way with respect to sounds arriving from the rear, causing a slight flattening of the polar response. This flattening increases as the diameter of the microphone (assuming it's cylindrical) reaches the wavelength of the frequency in question. Therefore, the smallest diameter microphone will give the best omnidirectional characteristics at high frequencies. The wavelength of sound at 10 kHz is little over an inch (3.4 cm) so the smallest measuring microphones are often 1/4" (6 mm) in diameter, which practically eliminates directionality even up to the highest frequencies. Omnidirectional microphones, unlike cardioids, do not employ resonant cavities as delays, and so can be considered the "purest" microphones in terms of low coloration; they add very little to the original sound. Being pressure-sensitive they can also have a very flat low-frequency response down to 20 Hz or below. Pressure-sensitive microphones also respond much less to wind noise than directional (velocity sensitive) microphones.

A unidirectional microphone is sensitive to sounds from only one direction. The diagram above illustrates a number of these patterns. The microphone faces upwards in each diagram. The sound intensity for a particular frequency is plotted for angles radially from 0 to 360°. (Professional diagrams show these scales and include multiple plots at different frequencies. These diagrams just provide an overview of the typical shapes and their names.)

The most common unidirectional microphone is a cardioid microphone, so named because the sensitivity pattern is heart-shaped (see cardioid). A hyper-cardioid is similar but with a tighter area of front sensitivity and a tiny lobe of rear sensitivity. These two patterns are commonly used as vocal or speech microphones, since they are good at rejecting sounds from other directions. Because they employ internal cavities to provide front-back delay, directional microphones tend to have more coloration than omnis, and they also suffer from low-frequency roll-off. These problems are overcome to a large extent by careful design, but only the best cardioids can begin to approach the performance of a tiny low-cost omni in terms of absolute accuracy. This is not always recognised, but is the price paid for directionality, often needed to exclude ambient reverberation wherever very close placement is impossible.

Figure 8 or bi-directional microphones receive sound from both the front and back of the element. Most ribbon microphones are of this pattern.

An Audio-Technica shotgun microphoneShotgun microphones are the most highly directional. They have small lobes of sensitivity to the left, right, and rear but are significantly more sensitive to the front. This results from placing the element inside a tube with slots cut along the side; wave-cancellation eliminates most of the off-axis noise. Shotgun microphones are commonly used on TV and film sets, and for field recording of wildlife.

An omnidirectional microphone is a pressure transducer; the output voltage is proportional to the air pressure at a given time.

On the other hand, a figure-8 pattern is a pressure gradient transducer; the output voltage is proportional to the difference in pressure on the front and on the back side. A sound wave arriving from the back will lead to a signal with a polarity opposite to that of an identical sound wave from the front. Moreover, shorter wavelengths (higher frequencies) are picked up more effectively than lower frequencies.

A cardioid microphone is effectively a superposition of an omnidirectional and a figure-8 microphone; for sound waves coming from the back, the negative signal from the figure-8 cancels the positive signal from the omnidirectional element, whereas for sound waves coming from the front, the two add to each other. A hypercardioid microphone is similar, but with a slightly larger figure-8 contribution.

Since pressure gradient transducer microphones are to some extent directional, their frequency response is dependent on the distance to the sound source. This is known as the proximity effect, resulting in a bass boost at distances of a few centimeters[citation needed].

[edit] Application-specific microphone designs
A lavalier microphone is made for hands-free operation. These small microphones are worn on the body and held in place either with a lanyard worn around the neck or a clip fastened to clothing. The cord may be hidden by clothes and either run to an RF transmitter in a pocket or clipped to a belt (for mobile use), or run directly to the mixer (for stationary applications).

A wireless microphone is one which does not use a cable. It usually transmits its signal using a small FM radio transmitter to a nearby receiver connected to the sound system, but it can also use infrared light if the transmitter and receiver are within sight of each other.

A contact microphone is designed to pick up vibrations directly from a solid surface or object, as opposed to sound vibrations carried through air. One use for this is to detect sounds of a very low level, such as those from small objects or insects. The microphone commonly consists of a magnetic (moving coil) transducer, contact plate and contact pin. The contact plate is placed against the object from which vibrations are to be picked up; the contact pin transfers these vibrations to the coil of the transducer. Contact microphones have been used to pick up the sound of a snail's heartbeat and the footsteps of ants. A portable version of this microphone has recently been developed.

A throat microphone is a variant of the contact microphone, used to pick up speech directly from the throat, around which it is strapped. This allows the device to be used in areas with ambient sounds that would otherwise make the speaker inaudible.

A parabolic microphone uses a parabolic reflector to collect and focus sound waves onto a microphone receiver, in much the same way that a parabolic antenna (e.g. satellite dish) does with radio waves. Typical uses of this microphone, which has unusually focused front sensitivity and can pick up sounds from many meters away, include nature recording, outdoor sporting events, eavesdropping, law enforcement, and even espionage. Parabolic microphones are not typically used for standard recording applications, because they tend to have poor low-frequency response as a side effect of their design.

[edit] Connectivity

[edit] Connectors
The most common connectors used by microphones are:

Male XLR connector on professional microphones
¼ inch mono phone plug on less expensive consumer microphones
3.5 mm (Commonly referred to as 1/8 inch mini) mono mini phone plug on very inexpensive and computer microphones
Some microphones use other connectors, such as 1/4 inch TRS (tip ring sleeve), 5-pin XLR, or stereo mini phone plug (1/8 inch TRS) on some stereo microphones. Some lavalier microphones use a proprietary connector for connection to a wireless transmitter. Since 2005, professional-quality microphones with USB connections have begun to appear, designed for direct recording into computer-based software studios.

[edit] Impedance matching
Microphones have an electrical characteristic called impedance, measured in ohms (Ω) that depends on the design. Typically, the rated impedance is stated.[2] Low impedance is considered under 600 Ω. Medium impedance is considered between 600 Ω and 10 kΩ. High impedance is above 10 kΩ. Most professional microphones are low impedance, about 200 Ω or lower. Low-impedance microphones are preferred over high impedance for two reasons: one is that using a high-impedance microphone with a long cable will result in loss of high frequency signal due to the capacitance of the cable; the other is that long high-impedance cables tend to pick up more hum (and possibly radio-frequency interference (RFI) as well). However, some equipment, such as vacuum tube guitar amplifiers, has an input impedance that is inherently high, requiring the use of a high impedance microphone or a matching transformer. Nothing will be damaged if the impedance between microphone and other equipment is mismatched; the worst that will happen is a reduction in signal or change in frequency response.

To get the best sound in most cases, the impedance of the microphone must be distinctly lower (by a factor of at least five) than that of the equipment to which it is connected. Most microphones are designed not to have their impedance "matched" by the load to which they are connected; doing so can alter their frequency response and cause distortion, especially at high sound pressure levels. There are transformers (confusingly called matching transformers) that adapt impedances for special cases such as connecting microphones to DI units or connecting low-impedance microphones to the high-impedance inputs of certain amplifiers, but microphone connections generally follow the principle of bridging (voltage transfer), not matching (power transfer). In general, any XLR microphone can usually be connected to any mixer with XLR microphone inputs, and any plug microphone can usually be connected to any jack that is marked as a microphone input, but not to a line input. This is because the signal level of a microphone is typically 40-60 dB lower (a factor of 100 to 1000) than a line input. Microphone inputs include the necessary amplification circuitry to deal with these very low level signals. The exception to these comments is in the case of certain ribbon and dynamic microphones which are most linear when operated into a load of known impedance [3]

[edit] Measurements and specifications

A comparison of the far field on-axis frequency response of the Oktava 319 and the Shure SM58Because of differences in their construction, microphones have their own characteristic responses to sound. This difference in response produces non-uniform phase and frequency responses. In addition, microphones are not uniformly sensitive to sound pressure, and can accept differing levels without distorting. Although for scientific applications microphones with a more uniform response are desirable, this is often not the case for music recording, as the non-uniform response of a microphone can produce a desirable coloration of the sound. There is an international standard for microphone specifications,[4] but few manufacturers adhere to it. As a result, comparison of published data from different manufacturers is difficult because different measurement techniques are used. The Microphone Data Website has collated the technical specifications complete with pictures, response curves and technical data from the microphone manufacturers for every currently listed microphone, and even a few obsolete models, and shows the data for them all in one common format for ease of comparison.[2]. Caution should be used in drawing any solid conclusions from this or any other published data, however, unless it is known that the manufacturer has supplied specifications in accordance with IEC 60268-4.

A frequency response diagram plots the microphone sensitivity in decibels over a range of frequencies (typically at least 0–20 kHz), generally for perfectly on-axis sound (sound arriving at 0° to the capsule). Frequency response may be less informatively stated textually like so: "30 Hz–16 kHz ±3 dB". This is interpreted as a (mostly) linear plot between the stated frequencies, with variations in amplitude of no more than plus or minus 3 dB. However, one cannot determine from this information how smooth the variations are, nor in what parts of the spectrum they occur. Note that commonly-made statements such as "20 Hz–20 kHz" are meaningless without a decibel measure of tolerance. Directional microphones' frequency response varies greatly with distance from the sound source, and with the geometry of the sound source. IEC 60268-4 specifies that frequency response should be measured in plane progressive wave conditions (very far away from the source) but this is seldom practical. Close talking microphones may be measured with different sound sources and distances, but there is no standard and therefore no way to compare data from different models unless the measurement technique is described.

The self-noise or equivalent noise level is the sound level that creates the same output voltage as the microphone does in the absence of sound. This represents the lowest point of the microphone's dynamic range, and is particularly important should you wish to record sounds that are quiet. The measure is often stated in dB(A), which is the equivalent loudness of the noise on a decibel scale frequency-weighted for how the ear hears, for example: "15 dBA SPL" (SPL means sound pressure level relative to 20 micropascals). The lower the number the better. Some microphone manufacturers state the noise level using ITU-R 468 noise weighting, which more accurately represents the way we hear noise, but gives a figure some 11 to 14 dB higher. A quiet microphone will measure typically 20 dBA SPL or 32 dB SPL 468-weighted.

The maximum SPL (sound pressure level) the microphone can accept is measured for particular values of total harmonic distortion (THD), typically 1%. This is generally inaudible, so one can safely use the microphone at this level without harming the recording. Example: "142 dB SPL peak (<1% THD)". The higher the value, the better, although microphones with a very high maximum SPL also have a higher self-noise.

The clipping level is perhaps a better indicator of maximum usable level, as the 1% THD figure usually quoted under max SPL is really a very mild level of distortion, quite inaudible especially on brief high peaks. Harmonic distortion from microphones is usually of low-order (mostly third harmonic) type, and hence not very audible even at 3-5%. Clipping, on the other hand, usually caused by the diaphragm reaching its absolute displacement limit (or by the preamplifier), will produce a very harsh sound on peaks, and should be avoided if at all possible. For some microphones the clipping level may be much higher than the max SPL.

The dynamic range of a microphone is the difference in SPL between the noise floor and the maximum SPL. If stated on its own, for example "120 dB", it conveys significantly less information than having the self-noise and maximum SPL figures individually.

Sensitivity indicates how well the microphone converts acoustic pressure to output voltage. A high sensitivity microphone creates more voltage and so will need less amplification at the mixer or recording device. This is a practical concern but is not directly an indication of the mic's quality, and in fact the term sensitivity is something of a misnomer, 'transduction gain' being perhaps more meaningful, (or just "output level") because true sensitivity will generally be set by the noise floor, and too much "sensitivity" in terms of output level will compromise the clipping level. There are two common measures. The (preferred) international standard is made in millivolts per pascal at 1 kHz. A higher value indicates greater sensitivity. The older American method is referred to a 1 V/Pa standard and measured in plain decibels, resulting in a negative value. Again, a higher value indicates greater sensitivity, so −60 dB is more sensitive than −70 dB.

[edit] Measurement microphones
Some microphones are intended for use as standard measuring microphones for the testing of speakers and checking noise levels etc. These are calibrated transducers and will usually be supplied with a calibration certificate stating absolute sensitivity against frequency.

[edit] Microphone calibration techniques

[edit] Pistonphone apparatus
A pistonphone is an acoustical calibrator (sound source) using a closed coupler to generate a precise sound pressure for the calibration of instrumentation microphones. The principle relies on a piston mechanically driven to move at a specified rate on a fixed volume of air to which the microphone under test is exposed. The air is assumed to be compressed adiabatically and the SPL in the chamber can be calculated from PV = const. The pistonphone method only works at low frequencies, but it can be accurate and yields an easily calculable sound pressure level. The standard test frequency is usually around 250 Hz.

[edit] Reciprocal method
This method relies on the reciprocity of one or more microphones in a group of 3 to be calibrated. It can still be used when only one of the microphones is reciprocal (exhibits equal response when used as a microphone or as a loudspeaker).

Loudspeaker

2007-06-30T06:02:00.001-07:00

A loudspeaker, speaker, or speaker system is an electromechanical transducer which converts an electrical signal into sound. The term loudspeaker is currently used for both individual devices and for complete systems consisting of one or more drivers (as the individual transducers are often called) in an enclosure, often with a crossover circuit. Their cost may range from pennies in a cheap radio to high-fidelity speaker systems costing many thousands of dollars. Loudspeakers are the most variable elements in any audio system, regardless of cost, and are responsible for marked audible differences between otherwise identical sound systems.

Full-range speaker systems are typically multi-driver systems, particularly when high SPL output or high accuracy are required. "Multi driver" means a speaker system containing two or more drive units, possibly including woofers, midranges, tweeters, or supertweeters. In loudspeaker specifications, systems are often classified as "N-way speakers", where N indicates the number of separate frequency bands, usually separated by an electrical filter called a crossover. A 2-way system will have woofer and tweeter sections; a 3-way system a combination of woofer, tweeter, and mid-range speakers, and so on.

Contents [hide]
1 History
2 Driver design
2.1 Driver types
2.1.1 Full range drivers
2.1.2 Subwoofer
2.1.3 Crossover
2.2 Enclosures
2.2.1 Wiring connections
3 Specifications
4 Electrical characteristics of a dynamic loudspeaker
4.1 Electromechanical measurements
4.2 Efficiency vs. Sensitivity
5 Interaction with the listening environment
5.1 Loudspeaker placement
6 Loudspeaker directivity
6.1 Point sources
6.2 Line sources
7 Other driver designs
7.1 Horn loudspeakers
7.2 Piezoelectric speakers
7.3 Electrostatic loudspeakers
7.4 Heil air motion transducers
7.5 Plasma arc speakers
7.6 Digital speakers
8 References
9 See also
10 External links

[edit] History
Alexander Graham Bell patented the first loudspeaker as part of his telephone in 1876. This was soon followed by an improved version from Ernst Siemens in Germany and England (1878). Nikola Tesla is believed to have created a similar device in 1881[1]. The modern design of moving-coil drivers was established by Oliver Lodge in (1898)[2]. The moving coil principle was patented in 1924 by Chester W. Rice and Edward W. Kellogg.

These first loudspeakers used electromagnets because large, powerful permanent magnets were not available at reasonable cost. The coil of an electromagnet, called a field coil, was energized by direct current through a second pair of connections to the driver. This winding usually served a dual role, acting also as a choke coil filtering the power supply of the amplifier to which the loudspeaker was connected.

The quality of loudspeaker systems until the 1950s was, to modern ears, poor. Continuous developments in enclosure design and materials have led to significant audible improvements. The most notable improvements in modern speakers are improvements in cone materials, the introduction of higher temperature adhesives, improved permanent magnet materials, improved measurement techniques, computer aided design and finite element analysis.

[edit] Driver design

Cut-away view of a dynamic loudspeakerThe most common type of driver uses a lightweight diaphragm connected to a rigid basket, or frame, via flexible suspension which constrains a coil of fine wire to move axially through a cylindrical magnetic gap. When an electrical signal is applied to the voice coil, a magnetic field is created by the electric current in the coil which thus becomes an electromagnet. The coil and the driver's magnetic system interact, generating a mechanical force which causes the coil, and so the attached cone, to move back and forth and so reproduce sound under the control of the applied electrical signal coming from the amplifier. The following is a brief discussion of the individual components of this most common type of loudspeaker.

The diaphragm is usually manufactured in a cone or dome shaped profile. Numerous materials may be used, but the most common are paper, plastic and metal. The ideal material would be stiff, light and well damped. In practice, all three of these criteria cannot be met, and thus driver design involves tradeoffs. Paper is light and well damped, but not stiff. Metal can be made stiff and light, but it is not well damped. Plastic can be light, but typically the stiffer it is made, the less well-damped it is. As a result, many cones are made of some sort of composite. This can either be a sandwich construction or simply a coating to stiffen or damp a cone.

The basket or frame must be designed for rigidity to avoid deformation which could cause the voice coil to rub against the magnet structure. Baskets are typically cast or stamped metal, although molded plastic baskets are becoming common, especially for inexpensive drivers. The frame plays a secondary role in conducting heat away from the coil.

The suspension system keeps the coil centered in the gap and provides a restoring force to make the speaker cone return to a neutral position after moving. A typical suspension system consists of two parts: the "spider", which connects the diaphragm or voice coil to the frame and provides the majority of the restoring force; and the "surround", which helps center the coil and allows free movement. The spider is usually made of a corrugated fabric disk. The surround can be a roll of rubber or foam or a corrugated fabric, attached to the outer circumference of the cone and to the frame.

The voice coil wire is usually copper, though aluminum, or rarely silver, may be used. Voice coil wire can be round, rectangular, or hexagonal, giving varying amounts of wire volume coverage in the available magnetic gap. The coil is oriented coaxially inside the gap, a small circular volume (a hole, slot, or groove) in the magnetic structure within which it can move back and forth. The gap establishes a concentrated magnetic field between the two poles of a permanent magnet; the outside of the gap being one pole and the center post (a.k.a. pole-piece) being the other. The center post and back-plate are sometimes a single piece called the yoke.

Modern driver magnets are almost always permanent and made of ceramic, ferrite, Alnico, or, more recently, rare earth. The size and type of magnet and the magnetic circuit differ depending on design goals. A current trend in design, due to increases in transportation costs and a desire for smaller, lighter devices (as in many home theater multi-speaker installations), is the use of rare earth magnet instead of ferrite types.

Driver design, and the combination of one or more drivers into an enclosure to make a speaker system, is both an art and science. Adjusting a design to improve performance is done using magnetic and material science theory, high precision measurements, as well as experienced listeners. Designers can use an anechoic chamber to ensure the speaker can be measured independently of room effects, or any of several electronic techniques. Some developers eschew anechoic chambers in favor of specific standardized room set-ups intended to simulate real-life listening conditions. Some of the issues speaker designers must confront are lobing, phase effects, off axis response, crossover complications, and psychoacoustics.

Most loudspeaker drivers are currently manufactured in China. The fabrication of finished loudspeaker systems is segmented, depending largely on price point. High-end speaker systems are usually made in the same region as their target markets and can command prices of $10,000 per pair and up. The lowest-priced speaker systems are mostly manufactured in China or other low-cost manufacturing locations. Although the manufacture of drivers has become essentially commoditized, the fabrication and subsequent sale of finished speaker systems still carry high profit margins. Partly for this reason, manufacturers are increasingly combining power amplifier electronics (a typically lower profit item) with finished speaker systems to create "powered speakers" with an overall higher market value.

[edit] Driver types

Exploded view of a dome tweeterA woofer is a driver capable of reproducing low (bass) frequencies. The usable frequency range varies widely according to design. Some woofers can cover the audio band from lowest bass to 3 kHz, while others only work up to 1 kHz or less. Some woofers are capable of very deep bass performance in an enclosure that is large enough and properly braced. Others woofers become unusable or highly distorting below 50 or 60 Hz, and so listeners who want to listen to music with very deep bass may need a subwoofer (see below).

A tweeter is a driver capable of reproducing the higher end of the audio spectrum, usually from around 3-5 kHz up to 20 kHz and beyond.

A mid-range speaker, also called a squawker, is designed to cover the middle of the audio spectrum, typically from a few hundred Hertz to about 4-5 kHz. Midranges are used when the other drivers are incapable of adequately covering the full audio range without them. They also increase system maximum output, as tweeters in 3-way systems can be spared the difficult requirement to reproduce lower frequencies; this increases their maximum sound output before damage.

[edit] Full range drivers
A full-range driver is designed to have as wide a frequency response as possible. These drivers are small, typically 2 to 6 inches (5 to 16 cm) in diameter to permit reasonably high frequency response, but this means they often have limited low distortion sound output at low frequencies and limited power handling capacity (due to a small voice coil).

They often employ an additional cone called a whizzer, a small, light cone attached to the woofer's apex near the dust cap, to extend the high frequency response and broaden the high frequency directivity. The main cone is built so as to flex more in this region at high frequencies than the rest of the cone. The result is that the main cone delivers the low frequencies and the whizzer cone contributes most of the higher frequencies. Since the whizzer cone is smaller than the main diaphragm, dispersion at high frequencies is improved over a driver with a single larger diaphragm. Full range drivers are one approach to avoiding the possible audible effects of multiple driver systems caused by non-coincident driver location and crossover issues.

[edit] Subwoofer
A subwoofer is a woofer driver used only for the lowest part of the audio spectrum. A typical subwoofer only reproduces sounds below perhaps 120 Hz; some can go lower than 20 Hz. Because the intended range of frequencies is limited, subwoofer design is usually simpler, often consisting of a single, subwoofer enclosed in a suitable (often bass reflex) cabinet. To accurately reproduce very low bass notes without unwanted resonance, subwoofers have to be large enough and properly braced. Subwoofers are often supplied with power amplifiers and electronic filters, with additional controls relevant to low frequency reproduction, such as phase switches built directly into the cabinet. These subwoofers are known as "active subwoofers". Some subwoofer systems also include sophisticated systems utilizing accelerometers or back EMF sensors to sense cone movement. The actual motion of the cone is compared to the input signal many times per second and the feedback circuitry applies continuous correction to the drive signal to enable the woofer to reproduce the input signal with less distortion. These last are often called "servo" or "motional feedback" subwoofers.

[edit] Crossover
Main article: Audio_crossover
In a multiple driver (i.e. 2-way, 3-way, etc...) loudspeaker system, some means must be provided to separate the frequency band into sections so that each driver will produce the frequency range it is designed for, and to reduce the interference between the drivers. This separation of the frequencies is accomplished using a type of filter circuit called a crossover. The ideal crossover would have no overlap, but this is not achievable in practice with standard analog filters. The vast majority of loudspeakers use a passive crossover circuit. Passive crossover circuits use only capacitors, inductors, and resistors, which are known as passive components. Active crossovers use extra amplification stages to divide the frequency range before the signal is amplified. These require a separate amplifier for each frequency range. There are some inherent advantages to active crossovers, but the added expense and complexity makes them most prevalent in professional sound applications.

[edit] Enclosures
Main article: Loudspeaker enclosure

A 4-way speaker system. The cabinet is narrow to reduce a diffraction effect called the 'baffle step'.Most loudspeaker systems consist of drivers mounted in an enclosure, or cabinet. The main physical role of the enclosure is provide a place to mount the drivers. Perhaps the simplest enclosure is a baffle, just a flat board with the drivers mounted to it. This simple enclosure has the disadvantage that at frequencies with a wavelength longer than the baffle dimensions the antiphase radiation from the rear of the cone is free to interfere with the front radiation and will cause uneven response and a loss of bass. If the baffle is made infinitely large, this problem goes away.

Since infinite baffles are impractical, most enclosures function by containing the rear radiation from the cone. The simplest is a sealed box. The sealed enclosure prevents transmission of the sound emitted from the rear of the loudspeaker to the listening space by ideally being rigid and airtight. Techniques used to reduce transmission of sound through the walls of the cabinet include thicker cabinet walls, lossy wall material, internal bracing, curved cabinet walls or more rarely visco-elastic materials or thin lead sheeting applied to interior enclosure walls.

However, this rigid enclosure will then induce internal reflection of sound which can then be retransmitted through the loudspeaker cone; again resulting in degradation of sound quality. This is reduced through internal absorption through the use of absorptive materials (often called "damping") such as fiberglass, wool or synthetic fiber batting within the enclosure. The internal shape of the enclosure can be designed to reduce this by reflecting sounds away from the loudspeaker where they may then be absorbed.

Many other enclosure types exist which attempt to modify the rear radiation, which is half of the energy radiated by the driver, so that it may add constructively to the output from the front of the cone. Many designs which do this (Bass reflex, passive radiator, transmission line, etc...) are often used to extend the low frequency response of the speaker system.

In an attempt to make the transition between drivers as seamless as possible, system designers have also attempted in recent years to time-align or phase adjust the drivers, which often involves moving one or more drivers forward or back, so that the acoustic centers of the drivers is in the same vertical plane. This sometimes involves tilting the face of a floor-mounted speaker back, or providing separate enclosure mounting for the drivers, or, less commonly, using electronic techniques to achieve the same effect. These attempts account for some of the unusual cabinet arrangements in speaker systems.

Another issue designers must manage is sound wave diffraction caused by the surfaces (face plate, cabinet, etc.) in which a driver is mounted. This is usually a problem at higher frequencies, as those wavelengths are similar to, or smaller than, cabinet dimensions. The problem is addressed by rounding the front edges of the cabinet or by using a smaller or narrower enclosure, or by strategic arrangement of the drivers. Sometimes, an absorptive layer such as felt is added to the mounting surface around a driver to reduce such effects.

[edit] Wiring connections

Five-way binding posts on a loudspeaker connected using banana plugs.Most loudspeakers use two wiring points to connect to the source of the signal (for example, to the audio amplifier or receiver). This is usually done using binding posts, or spring clips on the back of the enclosure. If the wires for left and right speakers (in a stereo setup) are not connected in phase with each other (the + and - connections on the speaker and amplifier should be connected to each other) the loudspeakers will be out of phase and destructive sound wave interference will occur when a common signal is sent to each speaker. In this case, any motion one cone (usually the woofer) makes will be opposite to the other. This type of wiring error doesn't damage speakers but does create inverse sound waves that partially cancel those from the other speaker. Due to the spacing of the speakers, the bass frequencies are where this phenomenon is most apparent.

[edit] Specifications

Specifications label on a loudspeakerSpeaker specifications generally include:

Speaker or driver type (individual units only) – Full-range, woofer, tweeter or mid-range.
Rated Power – Nominal or continuous power and peak or maximum short-term power that the loudspeaker can handle (that is, maximum allowed input power without thermally destroying the loudspeaker. It is not the power that the passive loudspeaker produces). Under special conditions, such as overdriving at very low frequencies or via sine wave input at higher frequencies, a loudspeaker may be damaged at much less than rated power.
Impedance – typically 4 Ω (ohms), 8 Ω, etc.
Baffle or enclosure type (enclosed systems only) – Sealed, bass reflex, etc.
Number of drivers (complete speaker systems only) – 2-way, 3-way, etc.
and optionally:

Crossover frequency(ies) (complete multi-driver systems only) – The frequency or frequencies where electrical filtering occurs.
Frequency response – The measured or specified variance in sound pressure level to a constant input over a specified range of frequencies, often including a variance such as within +/- 2.5 dB.
Thiele/Small parameters (individual drivers only) – these include the driver's Fs (resonance frequency), Qts (the driver's Q or damping factor at resonance), Vas (the equivalent air compliance volume of the driver), etc.
Sensitivity – The sound pressure level produced by a loudspeaker, usually specified in dB, measured at 1 meter with an input of 1 Watt or 2.83 Volts. This rating is often inflated by manufacturers.

[edit] Electrical characteristics of a dynamic loudspeaker
Main article: Electrical characteristics of a dynamic loudspeaker
The load a driver presents to an amplifier consists of a complex electrical impedance, a combination of resistance, and both capacitive and inductive reactance, reflecting the properties of the driver, its mechanical motion, the effects of crossover componenents (if any), and the effects of air loading on the driver as modified by the enclosure. Most amplifiers (amps) output specifications are given at a specific power into an ideal resistive load. However, a loudspeaker with a nominal impedance of 8 Ω does not really have a constant resistance. Instead, the voice coil is inductive, the enclosure changes the characteristics of the driver, and a passive crossover between the drivers and the amplifier contributes its own variations.

[edit] Electromechanical measurements
Fully characterizing the sound output of a loudspeaker in detail is difficult (for example, phase characteristics vs. frequency, impulse response at various frequencies, directivity vs. frequency, distortion vs. SPL output (eg, harmonic, intermodulation vs SPL output, compression, etc), stored energy (that is, ringing) vs. frequency and output level, small signal vs. large signal performance, etc.), but the raw sound pressure level output is rather easier to measure. The sound pressure level (SPL) a loudspeaker produces is measured in decibels (dBspl).

[edit] Efficiency vs. Sensitivity
Loudspeaker efficiency is defined as the sound power output divided by the electrical power input. Most loudspeakers are actually very inefficient transducers. Only about 1% of the electrical energy sent by an amplifier to a typical home loudspeaker is converted to the acoustic energy we can hear -- the remainder is converted to heat, typically in the voice coil and magnet assembly. The main reason for this is the difficulty of achieving proper impedance matching between the acoustic impedance of the drive unit and that of the air into which it is radiating.

Driver ratings based on the SPL for a given input voltage (corresponding closely to power input for a particular driver impedance) are known as sensitivity ratings and are, approximately, equivalent to efficiency. Sensitivity is usually defined as so many dB at 1 W electrical input, measured at 1 meter. The voltage used is often 2.83 VRMS, which happens to be 1 watt into an 8 Ω (nominal) speaker impedance (nominally true for many speaker systems). Measurements taken with this reference are quoted as dB with 2.83 V @ 1 m.

The sound pressure is measured at (or scaled to be equivalent to a measurement taken at) one meter from the loudspeaker and on-axis or directly in front of it under the conditions that the loudspeaker is radiating into an infinitely large space and mounted on an infinite baffle. Clearly then, sensitivity does not correlate precisely with efficiency as it also depends on the directivity of the driver being tested and the acoustic environment in front of the actually deployed loudspeaker. As a simple example, a cheerleader's horn makes more sound output in the direction it is pointed than the cheerleader could by herself, but the horn did not improve or increase the cheerleader's total sound power output much, it just focused it into a smaller space.

Typical home loudspeakers have sensitivities of about 85 to 95 dB for 1 W @ 1 m - an efficiency of 0.5-4%.
Sound reinforcement and public address loudspeakers have sensitivities of perhaps 95 to 102 dB for 1 W @ 1 m - an efficiency of 4-10%.
Rock concert, stadium PA, marine hailing, etc speakers all have higher sensitivities -- maybe 103 to 110 dB for 1 W @ 1 m - an efficiency of 10-20%.
A driver with a higher maximum power rating cannot necessarily be driven to louder levels than a lower rated one, since sensitivity and power handling are independent. In the examples which follow, assume for simplicity that the drivers being compared have the same electrical impedance, are operated at the same frequency which is within both driver's respective pass bands, and that power compression is and distortion are low. For the first example, a speaker 3 dB more sensitive than another will produce double the sound pressure level (or be 3 dB louder) for the same power input. Thus a 100 W driver ("A") rated at 92 dB for 1 W @ 1 m sensitivity will output twice as much acoustic power as a 200 W driver ("B") rated at 89 dB for 1 W @ 1 m when both are driven with 100 W of input power. For this particular example, when driven at 100 W, speaker A will produce the same SPL, or loudness, speaker B would produce with 200 W input. Thus a 3 dB increase in sensitivity of the speaker means that it will need half the amplifier power to achieve a given SPL; this translates into a smaller, less complex power amplifier and, often, to reduced overall cost.

It is not possible to combine high efficiency, especially at low frequencies, with compact enclosure size, and adequate low frequency response. One can, more or less, only choose two of the three parameters when designing a speaker system. So, for example, if extended low frequency performance and a small box size are important, one must accept low efficiency. This rule of thumb is sometimes called Hoffman's Iron Law (after J. A. Hoffman, the H in KLH).

[edit] Interaction with the listening environment
The interaction of a loudspeaker system with its environment is complex and is largely out of the designer's control. Most listening rooms present a more or less reflective environment, depending on size, shape, volume, and furnishings. This means the sound reaching a listener's ears consists not only of direct sound, but also of that sound delayed by traveling to and from (and being modified by) reflections from one or more surfaces. These reflected sound waves, when added to the direct sound, cause cancellation and addition at certain frequencies, changing the timbre and character of the signal being reproduced. Our brains are very sensitive to these small variations. This is part of the reason why a loudspeaker system sounds different at different listening positions or in different rooms.

A significant factor in the sound of a loudspeaker system is the amount of absorption and diffusion present in the environment. Clapping one's hands in an empty room, without draperies or carpet, will produce a zippy fluttery echo which is due both to a lack of absorption and to reverberation (that is, repeated echoes). The addition of hard surfaced furniture, wall hangings, and shelving will change the echoes, due primarily to the diffusion caused by somewhat reflective objects with shapes and textures having sizes on the order of the sound wavelengths being diffused. This somewhat breaks up the simple reflections otherwise caused by flat walls, floors and ceilings, and spreads the reflected energy of an incident wave over a larger angle on reflection.

Adding carpet, curtains, tapestries, people, or soft surfaced furniture will further change the interaction of a loudspeaker with the room by absorbing sound at various frequencies and reducing reflections at those frequencies. By and large, the thinner a material is, the less likely it will have an effect at low frequencies. An overabundance of absorption at high frequencies can be caused by large areas of absorptive materials and can cause a speaker system to sound deficient at higher frequencies, and likewise minimal absorption can cause an otherwise adequate loudspeaker to sound too bright or sibilant at those frequencies.

[edit] Loudspeaker placement
For good sound in a home environment, a listening room should have a balance of diffusion and absorption. Most systems will sound best when the speakers are set up more or less symmetrically with respect to the listener and also to room boundaries. Early reflections (the first reflection of a particular sound) do the most to color the sound (due to the so-called Haas effect in psychoacoustics), so placing speakers too near the rear or side walls is generally something to be avoided, although judicious use of absorbing or diffusing materials can somewhat moderate an otherwise poor placement location. Mounting a speaker in a wall (or in a bookshelf with books flush with the baffle) somewhat removes diffractive boundary concerns, but limits placement flexibility. In professional applications, placement is largely controlled by the location of the listening audience, required appearance (for example, prominence or invisibility), and available space. Fine adjustment is often not possible.

Another factor in room acoustics is a phenomena called standing waves. A one dimensional example is sound bouncing between two reflective boundaries. Sound resonates, or repeatedly reflects at particular frequencies, if the distance between the boundaries corresponds to an integral number of half wavelengths. Since sound travels at ~345 m/s, a pair of reflective boundaries separated by 5 meters will cause resonances at 34.5 Hz, 69 Hz, 103.5 Hz ..., recalling that wavelength is the speed of sound divided by the frequency. It is best, if possible, to arrange that no room wall length or height is simply related to any other. A cubical listening room would be most resonant since all dimensions are identical, with walls, floor and ceiling parallel, thus reinforcing the resonance modes. One approach is to ensure that each room dimension is related to another by the Golden Mean, which will ensure that the unavoidable reflections between walls are not reinforced by any others.

In a typical rectangular listening room, this resonant phenomenon happens in three dimensions, and there are even more complex interactions that involve four or even all six boundary surfaces. It is primarily an issue for low frequencies which are not much affected by such things as furniture or its placement. In addition, the location of the loudspeakers, and the listener, with respect to room boundaries affect how strongly the resonances are excited. Many people are familiar with certain locations in a room, club, or building which have much more, or less, bass - most usually near room walls or corners. This is because standing wave patterns are most pronounced in these locations and at lower frequencies, below the Schroeder frequency - typically around 200-300 Hz, depending on room size.

[edit] Loudspeaker directivity
This is an important issue because it affects the frequency balance of sound a listener hears, and also the interaction of the speaker system with the room.

In general, a sound source will radiate of one of four basic ways: as a point source, a line source, a planar source or a 3D source.

An extremely small point source is often considered ideal, because it radiates all frequencies equally in all directions in a spherical radiation pattern, and thus favors none. A theoretical line sources may be finite or effectively infinite, and will radiate sound in a cylindrical pattern. These first two source types are not actually practical, although real sound sources may approximate them, especially at some frequencies. In real life, most speaker systems and individual drivers are actually complex 3D shapes such as cones and domes.

Some drivers, and some enclosures (e.g. horns) take advantage of directivity in that, rather than radiating in a wide pattern, they focus sound into a constrained pattern. This is desirable for large areas such as theaters, concert halls, arenas and outdoor areas where the listener(s) may be a great distance from the sound source and yet should still hear well.

Less well understood are the psychoacoustic consequences of particular kinds of directivity. For instance, Amar Bose delivered a paper in 1968 to an Audio Engineering Society conference entitled "On the Design, Measurement and Evaluation of Loudspeakers" (reprints are available from the AES: http://www.aes.org/e-lib/browse.cfm?elib=1390), in which he discussed the issue of reflected versus direct sound, and that in live performance most, if not all, listeners are located in the sound field for which reflected sound dominates. Only a tiny fraction of the sound reaching the pinna of the ear arrives as direct sound from the musical instruments; in a typical home listening environment, speaker systems at that time delivered the higher frequencies directly at the ears of the listener. Does directionality matter in this context (i.e. reproduction of recorded sound)? Physics and engineering cannot resolve this issue; it lies in the domain of psychoacoustic and recording practice. The experiments Amar Bose conducted at MIT in the 1960s, reported in this paper, convinced him that a dominance of reverberant sound is important to the perception of quality in sound reproduction, IF one's standard of quality is informed by the experience of live performances.

[edit] Point sources
The neutrality of this article or section is disputed.
Please see the discussion on the talk page.

Point sources can be approximated (at least at low frequencies) as a planar form that creates a sound wave which becomes more directional as frequency increases, because the wavelength of the sound wave becomes small compared to the size of the diaphragm. That is, the intensity of the sound produced varies depending on the listener's angle relative to the central axis of the speaker.

A common variation on the dynamic loudspeaker cone design uses a dome as the moving part instead of the familiar inverted cone. Contrary to intuition, making the moving surface a dome rather than an inverted cone does not always help to direct sound evenly in a half-spherical space. The dome is used primarily because, in the case of a tweeter, its radiating surface is smaller than the voice coil and because a conical shape is difficult to fit in the tweeter structure -- unless specially modified, the magnet system's pole piece will mechanically interfere with cone motion at high amplitudes. Some tweeters do have inverted domes, however, but the pole piece is specially configured to accommodate the dome's shape. Some tweeters (TDL and other manufacturer use this shape) use bullet-shaped domes instead of domes with a constant radius. The intent is to reduce or eliminate bending in the center of the dome (consider that an egg shape is harder to push in at the pointed end than at the other). Finally, some manufacturers leave out the center of the dome altogether and only use the outer ring (called a ring radiator) to altogether avoid distortions of the inner part of the dome due to bending effects (for example, some models from Scan-Speak, Kea Audio, Vifa etc). A ring radiator also has better directivity (that is, is less directive) than a dome. A typical one inch dome tweeter begins to be directive at about 8000 Hz, below this frequency it approximates a point source, above this frequency, it becomes increasingly directional. At distances more than ~7 times the diameter of the cone or dome, the response is essentially that of a flat plane, at closer distances, the exact shape of the diaphragm becomes increasingly important.

Two dimensional and three dimensional sound sources can be monopolar, dipolar or bipolar. Most planar (that is, flat diaphragm) drivers are dipolar, which means that sound from the rear of the diaphragm is permitted to freely radiate. When the rear radiation is absorbed or trapped in a box, the diaphragm becomes a monopole radiator. Bipolar speakers, made by mounting in-phase monopoles on opposite sides of a box, are a method of approximating a point source or pulsating sphere.

Various manufacturers use assorted driver mounting arrangements, and the resulting radiation patterns, to more closely simulate the way sound is produced by real instruments, or to mimic one of the ideal sound source types, or simply to create a controlled energy distribution. Most professional audio speaker systems use horns or other dispersion control techniques because broad dispersion is a liability in many commercial situations such as concert sound or public address contexts.

The Manger bending wave transducer uses a bending wave scheme, in which vibration waves start from the center of a round flat diaphragm and travel to the outside. The rigidity of the material increases from the center to the outside. Short wavelength sound therefore radiate primarily from the inner area, while longer waves reach the edge of the speaker. To prevent reflections, long waves are absorbed by a surrounding damper. The Manger transducer covers the frequency range from 80 Hz to 35,000 Hz, and is close to an ideal point sound source. The Walsh loudspeaker systems from Ohm Acoustics have been quite similar in their bending scheme, though different in numerous details.

Coaxial speakers have been made commercially since the 1930s. These approximate a point source by moving the radiating axes of the various drivers close to the same point, usually with benefits in polar response. Coaxial mounting eliminates crossover lobing (that is, interference between drivers caused by non coincident placement). The woofer cone often acts as a horn in many respects. The technique of using concentric radiating elements for a multiway system has been used by several manufacturers, notably Technics. Cabasse recently published a paper analyzing 3-way and even 4-way coaxial speakers using concentric ring-shaped radiators. Several manufacturers (for example, Tannoy, Eminence, etc.) still build 2-way coaxial drivers in which the tweeter fires through a horn that passes through the woofer pole piece, and several (for example, KEF, SEAS, Kea-Audio, Tannoy etc.) build coaxial units in which the tweeter is mounted on the woofer pole piece. The small form factor this last approach requires has been made more effective by recent developments in rare earth magnets.[3]

Several manufacturers have attempted to simulate a point source by approximating a pulsating sphere. In the 1960s, Amar Bose (an MIT Professor) designed a one-eighth sphere loudspeaker system covered in small full-range drivers for room corner placement. The 1801 produced a wavefront very like that of an ideal sphere when wall reflections were included. Few were built and the system was not a commercial success, but it gave rise to commercially successful speaker system designs (the 901, most importantly) which also use multiple small drivers pointed in various directions to create a mixture of direct and reflected sound claimed to approximate that of a concert hall. In the 1801 and the 901, the small drivers involved were not actually inherently full-range and required considerable equalization to provide adequate low frequency performance and to compensate for decreasing high frequency performance. Especially at low frequencies, this approach demanded rather more amplifier power than competing speakers of the time. Both techniques have remained somewhat controversial.

The Ohm speaker drivers, whose principle was invented by Lincoln Walsh, use a single voice coil/cone mounted vertically, firing downwards into the top of the cabinet, but instead of the normal almost flat cone, has an extended cone entirely exposed at the top of the speaker. The usual problem with designing a cone driver is how to keep the cone as stiff as possible (without adding too much mass) so that it moves as a unit, and does not support traveling waves nor distort during cone breakup. The Walsh driver was so designed that the entire purpose of the cone's motion was to generate traveling waves down the cone from the magnetic motor (that is, voice coil and magnet structure) at the top. As the waves moved down the cone, the effect was to reproduce a 360 degree wavefront at all frequencies, more or less like a cylinder. This created a very effective omni-directional radiator (although it suffered the same "planarity" effect as ribbon tweeters for higher-frequency sounds ) and eliminated all problems of multiple drivers, such as crossover issues, phase anomalies between drivers, etc. However, in practice it was found necessary to use a very complex and expensive cone made of various materials along its length.

High fidelity speaker systems of this design are still being produced by Ohm in the US, and in Germany, by German-Physik and, as a variant, by Manger. This approach has not been used in professional sound reinforcement, most likely due to the delicacy of the physically large cone structure and the inherent cylindrical directivity.

[edit] Line sources
A ribbon speaker consists of a thin metal-film ribbon suspended in a magnetic field. The electrical signal is applied to the ribbon which moves with it, thus creating the sound. The advantage of a ribbon driver is that the ribbon has very little mass; thus, it can accelerate very quickly, yielding very good high-frequency response. Ribbon loudspeakers are often very fragile -- some can be torn by a strong puff of air. Most ribbon tweeters emit sound in a dipole pattern; a very few have backings which limit the dipole radiation pattern. Above and below the ends of the more or less rectangular ribbon, there is less audible output due to phase cancellation, but the precise amount of directivity depends on ribbon length. Ribbon designs generally require exceptionally powerful magnets which make them costly to manufacture. Ribbons have a very low resistance that most amplifiers cannot drive directly. A step down transformer is therefore typically used to increase the current through the ribbon. The amplifier "sees" a load that is the ribbon's resistance times the transformer turns ratio squared. The transformer must be carefully designed so that its frequency response and parasitic losses do not degrade the sound, further increasing cost and complication relative to conventional designs.

Planar magnetic speakers (having printed or embedded conductors on a flat diaphragm) are sometimes described as "ribbons", but are not truly ribbon speakers. The term planar is generally reserved for speakers which have roughly rectangular shaped flat radiating surfaces. Planar magnetic speakers consist of a flexible membrane with a voice coil printed or mounted on them. The current flowing through the coil interacts with the magnetic field of carefully placed magnets on either side of the diaphragm, causing the membrane to vibrate more or less uniformly and without much bending or wrinkling. The driving force covers a large percentage of the membrane surface and reduces resonance problems inherent in coil-driven flat diaphragms. Many designs touted as "ribbons" are in fact planar magnetic. Many of these designs have small cavities between the magnet structures and the diaphragm. This is not ideal and it sometimes creates a "cavity resonance" response peak that requires corrective filtration. Failure to correct this cavity resonance is a cause of the steely or shrill sound sometimes attributed to these designs.

There have also been many attempts to reduce the size of speaker systems, or alternatively to make them less obvious. One such attempt was the development of voice coil driven 'exciters' mounted to flat panels to act as sound sources. These can then be made in a neutral color and hung on walls where they will be less noticeable than many speakers, or can be deliberately painted with patterns in which case they can function decoratively. An example is Wharfedale Pro's 'Loudpanel' series. There are two related problems with flat panel techniques: first, a flat panel is necessarily more flexible than a cone shape in the same material, and therefore will move as a single unit even less, and second, resonances in the panel are difficult to control, leading to considerable distortions. Some progress has been made using such lightweight, rigid, yet damped, materials as Styrofoam, and there have been several flat panel systems commercially produced in recent years.

A newer implementation of the flat panel speaker system involves an intentionally flexible panel and an "exciter", mounted off-center in a location such that it excites the panel to vibrate, but with minimal resonances. Speakers using NXT techniques design methods can reproduce sound with a wide directivity pattern (paradoxically somewhat like a point source) and have been used in some computer speaker designs and a few small 'shelf systems' from such manufacturers as TEAC and Philips.

[edit] Other driver designs
Other types of drivers which depart from the most commonly used electro-dynamic driver mounted in an enclosure include:

[edit] Horn loudspeakers

Painting of Nipper, used by Gramophone Ltd and Victor Talking Machine, UK, & then RCA, USHorn speakers have been designed and built since the late 19th century; one is prominent in the RCA logo with Nipper the dog listening to His Master's Voice. Horns using modern electrodynamic drivers are a more recent development beginning shortly after the First World War. The increasing cross-sectional area of the horn allows for a greater mechanical advantage of the driver against the resistance of air, increasing the efficiency of the driving element. An efficient home loudspeaker system has a sensitivity of around 90 dB @ 2.83 volts (1 watt @ 8 Ohms) @ 1 Meter distance, while several home-use horn loaded speakers are rated as high as 100 dB @ 2.83 volts (1 watt @ 8 Ohms) @ 1 Meter. This is a tenfold increase in output at one watt, resulting in an output level which would require 10 watts from the speaker rated at 90 dB sensitivity, and is invaluable in some applications. The length and cross sectional mouth area required to create a bass or sub-bass horn may necessitate a horn several feet long. Due to the large volume that such a horn occupies, it is often necessary to fold the horn in order to allow it to fit its environment. Formerly, largely after WWII and before the stereo era, horns whose mouths took up much of a room wall were not uncommon amongst hi-fi fans. Such installations became much less acceptable when two were required, and entirely unthinkable in modern, multi-channel home systems.

Few full-range horns are being commercially produced for home use, and those which are have very high prices. But there is an active DIY horn building community around the world which has produced some visually striking enclosures, some claimed to be audibly excellent as well. More common are 'short horns' (of a practical, though still large, size) used for professional sound work. These are typically bass reflex enclosures usually with two large drivers (12" or 15") firing into a common horn with a very large throat. The horn in these cases is more used for dispersion control than acoustic loading at low frequencies. The Altec Lansing Voice of the Theater model is an example, first used in movie theater sound five decades ago.

[edit] Piezoelectric speakers
Piezoelectric speakers are frequently used as beepers in watches and other electronic devices, and are sometimes used as tweeters in less-expensive speaker systems, such as computer speakers and portable radios. Piezoelectric speakers have several advantages over conventional loudspeakers:

they have no voice-coil, therefore there is no electrical inductance to manage
it is easy to couple high-frequency electrical energy into the piezoelectric transducer since the transducers are resistant to overloads which would normally destroy the voice coil of a conventional loudspeaker.
they are an inherently capacitive electrical load so they usually do not require an external cross-over network. They can simply be placed in parallel with conventional inductive voice coil drivers.
There are also disadvantages:

their frequency response, in most cases, is inferior to that of other technologies. This is why they are generally used in single frequency (beeper) or non-critical applications
some amplifiers cannot drive capacitive loads well; this can cause high frequency oscillation, which results in distortion or damage to the amplifier.

[edit] Electrostatic loudspeakers
Electrostatic loudspeakers give a more linear response than electromechanical voice coils, though for considerably reduced maximum motion amplitude. The entire diaphragm is driven by electrostatic charges and is very closely controlled. For many years electrostatic loudspeakers had a reputation as an unreliable and occasionally dangerous product. A primary disadvantage was that the signal must be converted to a very high voltage at low current, which was problematic for reliability and maintenance. High voltage charges attract dust, and many of these the speakers developed a tendency to arc, particularly where the dust provided a partial discharge path. Arthur Janszen was granted U.S. Patent 2,631,196 in 1953 for the practical electrostatic design.

Full range electrostatic loudspeakers are large by nature. An early model, the KLH Model 9, was taller than most people. In addition, electrostatics are inherently dipole radiators and cannot, in practice, be used in enclosures to increase their efficiency as with common cone drives. In electrostatic loudspeakers, diaphragm excursion is limited to fractions of a millimeter whereas more ordinary dynamic cone loudspeakers can usually move many millimeters, even up to centimeters in some instances. This means that the membrane of a full range electrostatic loudspeaker must be larger than an equivalent dynamic loudspeaker to produce even marginally acceptable low frequency performance and output level. Electrostatic tweeters have proven to be more practical and are more widely used.

[edit] Heil air motion transducers
Dr Oscar Heil invented this design in the 1960s. ESS, a California manufacturer, licensed it, employed Dr Heil, and produced a range of speaker systems using them as tweeters during the 1970s and 1980s. Radio Shack, a large US retail store chain, also sold speaker systems using them as tweeters for a time.

In this approach, a pleated diaphragm is mounted in a magnetic field and forced to close and open under control of a music signal. Air is forced from between the pleats in accordance with the imposed signal, generating sound. The drivers are less fragile than ribbons and considerably more efficient (and able to produce higher absolute output levels) than ribbon, electrostatic, or planar magnetic tweeter designs. At present, there are two manufacturers of these drivers, both in Germany, one of which produces a range of high end professional speakers using tweeters and midrange drivers based on the technology.

[edit] Plasma arc speakers
Plasma arc loudspeakers use electrical plasma as a driver. Since plasma has minimal mass, but is charged and therefore can be manipulated by an electric field, the result is a very linear output at frequencies far higher than the audible range. Problems of maintenance and reliability for this approach tend to make it unsuitable for mass market use. In 1978 Dr. Alan Hill of the Los Alamos National Laboratory designed the Hill Plasmatronics, an $8000 monster whose plasma was generated from compressed helium gas.[4] This avoided the ozone and nitrous oxide produced by RF decomposition of air in an earlier generation of plasma tweeters made by the pioneering DuKane Corporation, who produced the Ionovac (marketed as the Ionofane in the UK) during the 1950s. Currently, there remain a few manufacturers, all in Germany it seems, and a do it yourself design has been published.

A less expensive variation on this theme is the use of a flame for the driver, as flames contain ionized (electrically charged) gases.[5]

[edit] Digital speakers
This section may stray from the topic of the article.
Please help improve this section or discuss this issue on the talk page. (help)

Digital speakers are an experimental but venerable technology, having been the subject of experiments by Bell Labs as far back as the 1920s. The design is simple; each bit drives a tiny speaker driver. A value of "1" causes that driver to be driven to full amplitude; a value of "0" causes it to be completely shut off. Increasingly significant bits drive speakers of twice the area of the previous (often in a ring around the previous driver).

The name is sometimes confused with speakers used with digital equipment, such as computer sound systems; they are not actually digital in this sense, being examples of typical if usually small speaker designs.

There are two problems with this design which has led to it being abandoned as impractical for the present. For a reasonable number of bits (required for adequate sound reproduction quality), the size of the system becomes very large. For example, a 16 bit signal compatible with the 16 bit audio CD standard, starting with a 2 square inch (13 cm²) driver for the least significant bit, would require a total area for the drivers of over 900 square feet (85 m²). Secondly, due to analog digital conversion, the effect of aliasing is unavoidable, so that the audio output is "reflected" at equal amplitude in the frequency domain, on the other side of the sampling frequency. Even accounting for the vastly lower efficiency of speaker drivers at such high frequencies, the result generates an unacceptably high level of ultrasonics accompanying the desired output.

Live Sound Mixing

2007-06-30T05:54:00.001-07:00

Live sound mixing is the art of combining and processing a number of audio signals together to create a "mix" that the audience or performers at a live show hear. There are two types of live sound mixing: Front of House (FOH) and Monitor mixing.

Whenever sound reinforcement is needed for a live performance of either music, theater, or spoken word, a sound system is set up to provide this reinforcement. This sound system generally comprises a number of microphones on the stage, a mixing board, a number of speakers, often a number of audio processing devices, and the cabling to connect all of these components. For smaller venues and sound systems, the performer(s) often do not need a live sound engineer to operate the system during their performance. But when the venue and complexity of the sound system reaches a certain size, at least one live sound engineer is needed to operate the system. A live sound engineer refers to a person that is experienced in the set up and operation of a sound reinforcement system.

For mid sized venues and sound systems, usually only one live sound engineer is needed to mix the sound. When only one engineer is present, both the Front of House mix and the Monitor mix are done by the one engineer with one mixing board at the Front of House position. For larger sound systems and venues, at least two engineers and a number of technicians are required to run the system. The two primary engineers are the Front of House engineer and the Monitor Engineer. The Front of House engineer mixes the sound that the audience hears in the house and the Monitor engineer mixes the sound that the performers hear on stage.

Contents [hide]
1 The monitor engineer
2 The front of house engineer
3 Set up, tear down, and techs
4 See also
5 External links

[edit] The monitor engineer

A monitor engineer and console at an outdoor eventThe Monitor engineer's role is most essential at music events as opposed to spoken word events. In most cases, each performer on stage has their own individual mix that is custom tailored by the monitor engineer to suit their audio needs. The monitor engineer is then faced with the challenge of pleasing anywhere from 4, 10, or maybe even more musicians with a good mix. Though monitor speakers are still in use today, the newest and highest quality monitor system is what is known as an In Ear Monitor (IEM) system. In Ear Monitors are those hearing aid type devices that you see your favorite rock stars wearing at their performances. These are basically a pair of headphones that are custom molded for the musicians individuals ears and therefore greatly reduces the outside noise that the performer hears. This isolation protects the musicians ears from getting damaged from the long durations of high volumes that they are subjected to on a large stage. It also allows them to hear their individual mix with more clarity. At the largest and highest budgeted of concert events, each musician is hearing their own individual in ear mix. This involves much more than simply mixing the sound, but requires a great deal of additional audio processing to increase the quality of the performer's mix and therefore encourage them to perform at their best.

[edit] The front of house engineer

An FOH console at an outdoor eventWhilst all the chaos is going on backstage with the monitor engineer, the Front of House engineer mixes the house sound for the audience at the back of the venue known as the "Front of House Position." The Front of House engineer (commonly known as the "noise boy" or "sound guy") uses a variety of processors and effects to tailor a musical and high quality mix of the performance that is being done on stage. Just as the monitor engineer is, they are constantly adjusting the volume of each instrument or voice on stage and are constantly adding and adjusting various effects for the musical requirements of the song. The efforts of the Front of House engineer often go unnoticed when the sound is good due to the fact that a well mixed show will sound natural to the general public and will not sound like anything is being done other than simple amplification of the music on stage, which is not at all the only thing going on.

[edit] Set up, tear down, and techs
The other duty that the live sound engineer serves is the set up and tear down of these sound reinforcement systems. For large tours and events, this is a long (sometimes multiple day) and strenuous process. It usually involves getting to the venue much much earlier than the performers with a semi truck load of gear and unloading and setting up this often heavy equipment quickly. Of course, the two engineers could never do this alone, as they are assisted by a number of audio techs that are responsible for maintaining the system during the show whilst the engineers focus on mixing the show and getting the best sounding mix that they can. After the show is done, the live sound engineers and techs must tear down and put away this large sound system and the reload it into the truck for the next show on the tour. The tear down process always ends up taking much less time than the set up process and usually only takes a few hours. Of course, as all this is going on with the sound system, there are many other aspects of the show going on to consider such as the concert lighting, backline, catering, artist management, merchandising, security, and audience direction just to list a few.

Live sound mixing is an artform in its own as there are a number of different ways that the mix can be done and a number of different ways that it can sound. The live sound engineer usually has a music sense of some sort so that they can make the proper decisions on how to mix different types of music and different types of songs at a concert. It is truly a field that is often overlooked by the general public yet without it, concerts would never be able to approach the size that they have reached today.

Music Sequencer

2007-06-30T05:51:00.001-07:00

In the field of electronic music, a sequencer is a device or piece of software that allows the user to record, play back and edit MIDI data. Sequencers do not record audio, only the control information for synthesizers to recreate the composition. Though the term 'sequencer' is today used primarily for software, some hardware synthesizers and almost all music workstations include a built-in MIDI sequencer, while drum machines generally have a step sequencer built in. There are still also standalone hardware MIDI sequencers, though the market demand for those has diminished greatly in the last ten years.

Many sequencers have features for limited music notation, and most are able to show music in a piano roll notation. (For software designed specifically for music notation, see scorewriter.)

Music can also be sequenced using trackers such as ModPlug Tracker, and some of those are able to sequence MIDI events too.

Contents [hide]
1 History
2 Step sequencers
3 List of software sequencers / DAWs with sequencing features
4 Hardware music sequencers
5 External links

[edit] History
Early analog music sequencers used a control voltage/trigger interface, but were replaced by digital hardware- or software-based MIDI sequencers, which play back MIDI events and MIDI control information at a specified number of beats per minute.

As computer speeds increased in the 1990s, audio recording, audio editing, and sample triggering features were added to the software. Software so enhanced is called a digital audio workstation (DAW) to distinguish from sequencers and multitrack recording programs. DAWs almost always include sequencing features but, strictly speaking, go beyond what a sequencer is.

[edit] Step sequencers
A special case of sequencers are step sequencers. Instead of recording played notes or drawing notes by hand on the piano roll, the user composes patterns using a grid of (usually) 16 buttons, or steps, each step being 1/16th of a measure. Step sequencer patterns are monophonic by nature, but usually a single pattern may contain individual subpatterns for a number of different instruments. These patterns are then chained together to form longer compositions. Step sequencers are mostly used in drum machines and grooveboxes.

[edit] List of software sequencers / DAWs with sequencing features

Cakewalk's Home Studio 2002.
Steinberg's Cubase VST.For a list of trackers, see the tracker article.
Also see: List of MIDI editors and sequencers.

Commercial Sequencers:

Acid and Cinescore from Sony
Cubase and Nuendo from Steinberg
Digital Performer and AudioDesk from MOTU
FL Studio from Image Line Software
Live from Ableton
Logic Pro, Logic Express and Garage Band from Apple
Pro Tools from Digidesign
Reason from Propellerhead
Samplitude, Sequoia, Music Maker and Music Studio from Magix
SAWStudio from RML Labs [1]
Sonar, Project5 and Home Studio from Cakewalk
Storm from Arturia
Tracktion from Mackie
Open Source Sequencers:

Jazz++ [2]
LMMS [3]
MusE
Rosegarden
Seq24 [4]
Hydrogen

[edit] Hardware music sequencers
In alphabetical order (and by no means exhaustive):

AKAI MPC series
Alesis MMT-8
Clavivox, keyboard synth patented in 1956 by Raymond Scott
Doepfer MAQ 16-3
Doepfer Schaltwerk
Doepfer Regelwerk
Ensoniq ASR-10
Ensoniq ESQ-1
Ensoniq EPS-16
Fairlight CMI
Frostwave Fat Controller
genoQs Octopus
Infection Music Phaedra
Infection Music Zeit
Latronic Notron
Kawai Q-80
Korg SQ-8
Korg SQD-1
Korg SQD-8
Manikin Schrittmacher
Moog 960 Sequential Controller -- part of the Moog modular synthesizer system, and possibly the earliest sequencer.
Radikal Technologies Spectralis
RCA Mark II Sound Synthesizer (Victor). Room-filling device built in 1957 for a half-million dollars. Included a 4-polyphony synth with 12 oscillators, a sequencer fed with paper tape, and a shellac record lathe for output.
Roland MC-4
Roland MC-8
Roland MC-300
Roland MC-303
Roland MC-327
Roland MC-50
Roland MC-50 Mk2
Roland MC-500 Microcomposer
Roland MC-505
Roland MC-808
Roland MC-909
Roland MV-30
Roland MV-8000
Roland SB-55
Roland TB-303
Sequential Circuits PolySequencer
Sequentix P3
Yamaha PSR-3000
Yamaha QX1
Yamaha QX3
Yamaha QX5
Yamaha QX7
Yamaha QX21
Yamaha QY10
Yamaha QY300
Yamaha QY700
Yamaha QY100
Yamaha RM1x
Yamaha RS7000
Zyklus MPS

Musical Instrument Digital Interface (MIDI)

2007-06-30T04:50:00.001-07:00

MIDI (Musical Instrument Digital Interface; IPA: /ˈmɪdi/) is an industry-standard electronic communications protocol that enables electronic musical instruments, computers and other equipment to communicate, control and synchronize with each other in real time.

Note names and MIDI note numbers.MIDI does not transmit an audio signal or media — it simply transmits digital data "event messages" such as the pitch and intensity of musical notes to play, control signals for parameters such as volume, vibrato and panning, cues and clock signals to set the tempo. As an electronic protocol, it is notable for its success, both in its widespread adoption throughout the industry, and in remaining essentially unchanged in the face of technological developments since its introduction in 1983. Also see: Category:MIDI standards

Contents [hide]
1 History
2 Overview
3 MIDI interfaces
4 MIDI message interoperability
5 How MIDI channel messages work
6 How MIDI Show Control works
7 The MIDI 1.0 Protocol
7.1 Hardware Transport (Electrical and Mechanical Connections)
7.2 Message Format
7.2.1 Low bandwidth
8 MIDI file formats
8.1 Standard MIDI File (SMF) Format
8.2 MIDI Karaoke File (.KAR) Format
8.3 XMF File Formats
8.4 RMI File Format
9 MIDI usage and applications
9.1 Extensions of the MIDI standard
9.1.1 General MIDI
9.1.2 General MIDI 2
9.1.3 SP-MIDI
9.1.4 Alternate Hardware Transports
9.1.5 Alternate Tunings
9.2 Other applications of MIDI
9.3 MIDI controllers: hardware, software, datastream
10 Beyond MIDI 1.0
10.1 OSC
10.2 mLAN
10.3 HD-MIDI
11 MIDI software
12 Sound samples
13 See also
14 External links
14.1 Official MIDI Standards Organizations
14.2 Unofficial Sources
14.3 MIDI Search engines
14.4 Other resources

[edit] History
By the end of the 1970s, electronic musical devices were becoming increasingly common and affordable. However, devices from different manufacturers were generally not compatible with each other and could not be interconnected. Different interfacing models included:

analog control voltages at various standards (such as 1 volt per octave, or the logarithmic "hertz per volt")
analog clock, trigger and "gate" signals (both positive "V-trig" and negative "S-trig" varieties, between -15V to +15V)
proprietary digital interfaces such as Roland Corporation's DCB (digital control bus) and Yamaha's "keycode" system.
In an attempt to find a way forward from this situation, audio engineer and synthesizer designer Dave Smith of Sequential Circuits, Inc. proposed the MIDI standard in 1981 in a paper to the Audio Engineering Society. The proposal received widespread enthusiasm within the industry, and the MIDI Specification 1.0 was published in August 1983. Today, Dave Smith is generally regarded as the "Father of MIDI" and MIDI technology has been standardized and is maintained by the MIDI Manufacturers Association (MMA).

[edit] Overview
All official MIDI standards are jointly developed and published by the MIDI Manufacturers Association (MMA) in Los Angeles, California, USA (http://www.midi.org), and for Japan, the MIDI Committee of the Association of Musical Electronic Industry (AMEI) in Tokyo (http://www.amei.or.jp). The primary reference for MIDI is The Complete MIDI 1.0 Detailed Specification, document version 96.1, available only directly from MMA in English, or from AMEI in Japanese.

The MIDI Show Control (MSC) protocol (in the Real Time System Exclusive subset) is an industry standard ratified by the MIDI Manufacturers Association in 1991 which allows all types of media control devices to talk with each other and with computers to perform show control functions in live and canned entertainment applications. Just like musical MIDI (above), MSC does not transmit the actual show media — it simply transmits digital data providing information such as the type, timing and numbering of technical cues called during a multimedia or live theatre performance.

Almost all music recordings today use MIDI devices. In addition, MIDI is also used to control hardware including recording devices and live performance equipment such as stage lights and effects pedals.

MIDI allows computers, synthesizers, MIDI controllers, sound cards, samplers and drum machines to control one another, and to exchange system data.

MIDI was a major factor in bringing an end to the "wall of synthesizers" phenomenon in 1970s-80s rock music concerts, when keyboard instrument performers were sometimes hidden behind banks of various instruments. Following the advent of MIDI, many synthesizers were released in rack-mount versions, enabling performers to control multiple instruments from a single keyboard.

Another important result of MIDI has been the development of hardware and computer-based sequencers, which can be used to record, edit and play back performances. In the years immediately after the 1983 ratification of the MIDI specification, MIDI interfaces were released for both the Apple Macintosh computer and the Windows platform, allowing for the development of a market for powerful, inexpensive, and now-widespread computer-based MIDI sequencers.

Synchronization of MIDI sequences is made possible by the use of MIDI timecode, an implementation of the SMPTE time code standard using MIDI messages, and MIDI timecode has become the standard for digital music synchronization.

A number of music file formats have been based on the MIDI bytestream. These formats are very compact; a file as small as 10 KB can produce a full minute of music. This is advantageous for applications such as mobile phone ringtones, and some video games.

The term "MIDI sound" has often been used as a synonym for "bad sounding computer music", but this reflects a misunderstanding: MIDI does not define the sound, only the control protocol. This is probably a result of the poor quality sound sythesis provided by many early sound cards, which relied on FM synthesis instead of wavetables to produce audio.

[edit] MIDI interfaces
All MIDI In and MIDI Out connectors are part of a MIDI interface. A MIDI interface moves internal binary data to the MIDI Out connector for transmission to another device's MIDI In connector, in MIDI message form. It also receives incoming MIDI messages arriving on the MIDI In connector (from another device's MIDI Out connector) into internal binary data. All MIDI compatible instruments have a built-in MIDI interface. Some computers' sound cards have a built-in MIDI Interface, whereas others require an external MIDI Interface which is usually connected to the computer via USB or FireWire.

[edit] MIDI message interoperability
All MIDI compatible controllers, musical instruments, and MIDI-compatible software follow the same MIDI 1.0 specification, and thus interpret any given MIDI message the same way, and so can communicate with and understand each other. For example, if a note is played on a MIDI controller, it will sound at the right pitch on any MIDI instrument whose MIDI In connector is connected to the controller's MIDI Out connector. Often, the joystick port doubles as a midi port.

[edit] How MIDI channel messages work
When a musical performance is played on an MIDI instrument (or controller) it transmits MIDI channel messages from its MIDI Out connector. A typical MIDI channel message sequence corresponding to a key being struck and released on a keyboard is:

The user presses the middle C key with a specific velocity (which is usually translated into the volume of the note but can also be used by the synthesiser to set the timbre as well). ---> The instrument sends one Note On message.
The user changes the pressure applied on the key while holding it down - a technique called aftertouch (can be repeated, optional). ---> The instrument sends one or more Aftertouch messages.
The user releases the middle C key, again with the possibility of velocity of release controlling some parameters. ---> The instrument sends one Note Off message.
Note On, Aftertouch, and Note Off are all channel messages. For the Note On and Note Off messages, the MIDI specification defines a number (from 0–127) for every possible note pitch (C, C#, D etc.), and this number is included in the message. For example, the Middle C note played on any MIDI compatible musical instrument will always transmit the same MIDI channel message from its MIDI Out connector.

Other performance parameters can be transmitted with channel messages, too. For example, if the user turns the pitch wheel on the instrument, that gesture is transmitted over MIDI using a series of Pitch Bend messages (also a channel message). The musical instrument generates the messages autonomously; all the musician has to do is play the notes (or make some other gesture that produces MIDI messages). This consistent, automated abstraction of the musical gesture could be considered the core of the MIDI standard.

[edit] How MIDI Show Control works
Main article: MIDI Show Control.

When any cue is called by a user (typically a Stage Manager) and/or preprogrammed timeline in a show control software application, the show controller transmits one or more Real Time System Exclusive messages from its 'MIDI Out' port. A typical MSC message sequence is:

the user just called a cue
the cue is for lighting device 3
the cue is number 45.8
the cue is in cue list 7

[edit] The MIDI 1.0 Protocol
Main article: The MIDI 1.0 Protocol
IMPORTANT: Some of the information in this section diverges from the official MMA/AMEI MIDI specifications in terminology and in technical detail. Developers interested in maximizing interoperability are encouraged to work directly from the official MMA/AMEI specifications.

There are two sides to MIDI 1.0: the hardware transport specification describing the electrical and mechanical connection, and the message format specification.

[edit] Hardware Transport (Electrical and Mechanical Connections)

MIDI ports and cable.The MIDI standard consists of a communications messaging protocol designed for use with musical instruments, as well as a physical interface standard. It consists physically of a one-way (simplex) digital current loop serial communications electrical connection signaling at 31,250 bits per second. One start bit (must be 0), eight data bits, no parity bit and one stop bit (must be 1) is used.

[edit] Message Format
Every MIDI connection is a one-way connection from the MIDI Out connector of the sending device to the MIDI In connector of the receiving device. Each such connection can carry a stream of MIDI messages, with most messages representing a common musical performance event or gesture such as note-on, note-off, controller value change (including volume, pedal, modulation signals, etc.), pitch bend, program change, aftertouch, channel pressure. All of those messages include channel number. There are 16 possible channels in the protocol. The channels are used to separate "voices" or "instruments", somewhat like tracks in a multi-track mixer.

[edit] Low bandwidth
MIDI messages are extremely compact, due to the low bandwidth of the connection, and the need for real-time accuracy. Most messages consist of a status byte (channel number in the low 4 bits, and an opcode in the high 4 bits), followed by one or two data bytes. However, the serial nature of MIDI messages means that long strings of MIDI messages take an appreciable time to send, at times even causing audible delays, especially when dealing with dense musical information or when many channels are particularly active.

To further optimize the data stream, "Running status", a convention that allows the status byte to be omitted if it would be the same as that of the previous message, helps to mitigate bandwidth issues somewhat.

[edit] MIDI file formats

[edit] Standard MIDI File (SMF) Format
MIDI messages (along with timing information) can be collected and stored in a computer file system, in what is commonly called a MIDI file, or more formally, a Standard MIDI File (SMF). The SMF specification was developed by, and is maintained by, the MIDI Manufacturers Association (MMA). MIDI files are typically created using computer-based sequencing software (or sometimes a hardware-based MIDI instrument or workstation) that organizes MIDI messages into one or more parallel "tracks" for independent recording and editing. In most but not all sequencers, each track is assigned to a specific MIDI channel and/or a specific General MIDI instrument patch. Although most current MIDI sequencer software uses proprietary "session file" formats rather than SMF, almost all sequencers provide export or "Save As..." support for the SMF format.

An SMF consists of one header chunk and one or more track chunks. There are three SMF formats; the format is encoded in the file header. Format 0 contains a single track and represents a single song performance. Format 1 may contain any number of tracks, enabling preservation of the sequencer track structure, and also represents a single song performance. Format 2 may have any number of tracks, each representing a separate song performance. Sequencers do not commonly support Format 2.

Large collections of SMFs can be found on the web, most commonly with the extension .mid. These files are most frequently authored with the assumption that they will be played on General MIDI players.

[edit] MIDI Karaoke File (.KAR) Format
MIDI-Karaoke (which uses the ".kar" file extension) files are an "unofficial" extension of MIDI files, used to add synchronized lyrics to standard MIDI files. SMF players play the music as they would a .mid file but do not display these lyrics unless they have specific support for .kar messages. These often display the lyrics synchronized with the music in "follow-the-bouncing-ball" fashion, essentially turning any PC into a karaoke machine.

MIDI-Karaoke file formats are not maintained by any standardization body.

[edit] XMF File Formats
The MMA has also defined (and AMEI has approved) a new family of file formats, XMF (eXtensible Music File), some of which package SMF chunks with instrument data in DLS format (Downloadable Sounds, also an MMA/AMEI specification), to much the same effect as the MOD file format. The XMF container is a binary format (not XML-based, although the file extensions are similar). See the main article Extensible Music Format (XMF).

[edit] RMI File Format
On Microsoft Windows, the system itself uses RIFF-based MIDI files with the .rmi extension. Note, Standard MIDI Files per se are not RIFF-compliant. An RMI file, however, is simply a Standard MIDI File wrapped in a RIFF header. If the RIFF header is thrown away, the result should be a regular Standard MIDI File.

The RMI file format is not maintained by any standardization body.

[edit] MIDI usage and applications
Main article: MIDI usage and applications

[edit] Extensions of the MIDI standard
Many extensions of the original official MIDI 1.0 spec have been standardized by MMA/AMEI. Only a few of them are described here; for more comprehensive information, see the MMA web site.

[edit] General MIDI
The General MIDI (GM) and General MIDI 2 (GM2) standards define a MIDI instrument's response to the receipt of a defined set of MIDI messages. As such, they allow a given, conformant MIDI stream to be played on any conformant instrument. Although dependent on the basic MIDI 1.0 specification, the GM and GM2 specifications are each separate from it. As such, it is not generally safe to assume that any given MIDI message stream or MIDI file is intended to drive GM-compliant or GM2-compliant MIDI instruments. General Midi 1 was introduced in 1991.

[edit] General MIDI 2
Later, companies in Japan's Association of Musical Electronics Industry (sic) (AMEI) developed General MIDI Level 2 (GM2), incorporating aspects of the Yamaha XG and Roland GS formats, extending the instrument palette, specifying more message responses in detail, and defining new messages for custom tuning scales and more. The GM2 specs are maintained and published by the MMA and AMEI.

General MIDI 2 was introduced in 1992.

[edit] SP-MIDI
Later still, GM2 became the basis of the instrument selection mechanism in Scalable Polyphony MIDI (SP-MIDI), a MIDI variant for mobile applications where different players may have different numbers of musical voices. SP-MIDI is a component of the 3GPP mobile phone terminal multimedia architecture, starting from release 5.

GM, GM2, and SP-MIDI are also the basis for selecting player-provided instruments in several of the MMA/AMEI XMF file formats (XMF Type 0, Type 1, and Mobile XMF), which allow extending the instrument palette with custom instruments in the Downloadable Sound (DLS) formats, addressing another major GM shortcoming.

[edit] Alternate Hardware Transports
In addition to the original 31.25 kBaud current-loop, 5-pin DIN transport, transmission of MIDI streams over USB, IEEE 1394 AKA FireWire, and ethernet is now common. Perhaps in the long run the IETF's RTP MIDI specification for transport of MIDI streams over ethernet and internet may completely supersede the original DIN transport, since RTP MIDI is capable of providing the high-bandwidth channel that earlier alternatives to MIDI (such as ZIPI) were intended to bring. See external links below for further information.

[edit] Alternate Tunings
By convention, instruments that receive MIDI generally use the conventional 12-pitch per octave equal temperament tuning system. Unfortunately this tuning system makes many types of music inaccessible because the music depends on a different intonation system. To address this issue in a standardized manner, in 1992 the MMA ratified the MIDI Tuning Standard, or MTS. This standard allow MIDI instruments that support MTS to be tuned in any way desired, through the use of a MIDI Non-Real Time System Exclusive message.

MTS uses three bytes, which can be thought of as a three-digit number base 128, to specify a pitch in logarithmic form. The following formula gives the byte values needed to encode a given frequency in Hertz:

For a note in A440 equal temperament, this formula delivers the standard MIDI note number. Any other frequencies fill the space evenly.

While support for MTS is not particularly widespread in commercial hardware instruments, it is nonetheless supported by some instruments and software, for example the free software programs TiMidity and Scala (program), as well as other microtuners.

[edit] Other applications of MIDI
MIDI is also used every day as a control protocol in applications other than music, including:

show control
theatre lighting
special effects
sound design
recording system synchronization
audio processor control
computer networking, as demonstrated by the early first-person shooter game MIDI Maze, 1987
animatronic figure control
Such non-musical applications of MIDI are possible because any device built with a standard MIDI Out connector should in theory be able to control any other device with a MIDI In port, just as long as the developers of both devices have the same understanding about the semantic meaning of all the MIDI messages the sending device emits. This agreement can come either because both follow the published MIDI specifications, or else in the case of any non-standard functionality, because the message meanings are agreed upon by the two manufacturers.

[edit] MIDI controllers: hardware, software, datastream
The term MIDI controller is used in two different ways.

In one sense, a MIDI controller is a hardware or software entity able to transmit MIDI messages via a MIDI Out connector to other devices with MIDI In connectors.
In the other (more technical) sense, a MIDI controller is any parameter in a device with a MIDI In connector that can be set with the MIDI Control Change message. For example, a synthesizer may use controller number 18 for a low-pass filter's frequency; to open and close that filter with a physical slider, a user would assign the slider to transmit controller number 18. Then, all changes in the slider position will be transmitted as MIDI Control Change messages with the controller number field set to 18; when the synthesizer receives the messages, the filter frequency will change accordingly.

[edit] Beyond MIDI 1.0
Although traditional MIDI connections work well for most purposes, a number of newer message protocols and hardware transports have been proposed over the years to try to take the idea to the next level. Some of the more notable efforts include:

[edit] OSC
The Open Sound Control (OSC) protocol was at CNMAT. OSC has been implemented in the well-known software synthesizer Reaktor and in other projects including SuperCollider, Pure Data, Isadora, Max/MSP, Csound, VVVV and ChucK. The Lemur Input Device, a customizable touch panel with MIDI controller-type functions, also uses OSC. OSC differs from MIDI over traditional 5-pin DIN in that it can run at broadband speeds when sent over Ethernet connections. Unfortunately few mainstream musical applications and no standalone instruments support the protocol so far, making whole-studio interoperability problematic. OSC is not owned by any private company, however it is also not maintained by any standards organization.

[edit] mLAN
Yamaha has its mLAN[1] protocol, which is a based on the IEEE 1394 transport (also known as FireWire) and carries multiple MIDI message channels and multiple audio channels. mLAN is not maintained by a standards organization as it is a proprietary protocol. mLAN is open for licensing.

Audio Signal Processing

2007-06-30T04:48:00.000-07:00

Audio signal processing
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Audio signal processing, sometimes referred to as audio processing, is the processing of a representation of auditory signals, or sound. The representation can be digital or analog.

The focus in audio signal processing is most typically a mathematical analysis of which parts of the signal are audible. For example, a signal can be modified for different purposes such that the modification is controlled in the auditory domain.

The parts of the signal are heard and which are not, is not decided merely by physiology of the human hearing system, but very much by psychological properties. These properties are analysed within the field of psychoacoustics

Contents [hide]
1 History of audio processing
2 Analog signals
3 Digital signals
4 Application areas
4.1 Audio Broadcasting

[edit] History of audio processing
Audio processessing was necessary for early radio broadcasting -- as there were many problems with studio to transmitter links.

[edit] Analog signals
An analog representation is usually electrical; a voltage level represents the air pressure waveform of the sound.

[edit] Digital signals
A digital representation expresses the pressure wave-form as a sequence of symbols, usually binary numbers, which permits digital signal processing. It must be noted that all real world audio signals are continuous-time analog signals. Therefore, sampling and quantization must be applied to convert the continuous-time analog signal to a discrete-time digital representation. While such a conversion is lossy, most modern audio systems use this approach as the techniques of digital signal processing are much more powerful and efficient than analog domain signal processing.

[edit] Application areas
Processing methods and application areas include storage, level compression, data compression, transmission, enhancement (e.g., equalization, filtering, noise cancellation, echo or reverb removal or addition, etc.)

[edit] Audio Broadcasting
Audio broadcasting (be it for television or audio broadcasting) is perhaps the biggest market segement (and user area) for audio processing products -- globally.

Traditioanlly the most important audio processing (in audio brodcating) takes place just before the transmitter. Studio audio processing is limited in the modern era due to digital audio systems (mixers, routers) being pervasive in the studio.

In audio broadcasting, the audio processer must

prevent overmodulation, and minimize it when it occours
maximize overall loudness
compensate for non-lineral transmitters, more common with mediumwave and shortwave broadcasting

Audio Compression

2007-06-30T04:47:00.001-07:00

Audio compression can mean two things:

Audio data compression - in which the amount of data in a recorded waveform is reduced for transmission. This is used in CD and MP3 encoding, internet radio, and the like.
Audio level compression - in which the dynamic range (difference between loud and quiet) of an audio waveform is reduced. This is used in guitar effects racks, recording studios, etc.

Digital Audio

2007-06-30T04:44:00.000-07:00

A sound wave, in gray, represented digitally, in red (after a zero-order hold but before filtering)Digital audio uses digital signals for sound reproduction. This includes analog-to-digital conversion, digital-to-analog conversion, storage, and transmission.

Digital audio has emerged because of its usefulness in the recording, manipulation, mass-production and distribution of sound. Modern distribution of music across the internet through on-line stores depends on digital recording, and digital compression algorithms. Distribution of audio as data files rather than as physical objects has significantly reduced costs of distribution. However, it has brought about a rise in music sharing through peer to peer networks, which is illegal in many countries as copyright infringement. The Recording Industry Association of America and other organizations claim that music sharing severely harms the profitability of their business.

From the wax cylinder, to the compact cassette, analogue audio music storage and reproduction have been based on the same principles upon which human hearing are based.

In an analogue audio system, sounds begin as physical waveforms in the air, are transformed into an electrical representation of the waveform, via a transducer (for example, a microphone), and are stored or transmitted. To be re-created into sound, the process is reversed, through amplification and then conversion back into physical waveforms via a loudspeaker. Although its nature may change, its fundamental wave-like characteristics remain unchanged during its storage, transformation, duplication, amplification. All analogue audio signals are susceptible to noise and distortion, due to the inherent noise present in electronic circuits.

On the other hand, the digital audio chain begins when an analogue audio signal is converted into electrical signals — ‘on/off’ pulses — rather than electro-mechanical signals. This signal is then re-encoded (rather like a spy might use a code book), in order to combat any errors that might occur in the storage or transmission of the signal. It is this "channel coding" that is essential to the ability of the digital system to recreate the analogue signal upon replay. An example of a channel code is Eight to Fourteen Bit Modulation as used in the audio Compact Disc.

Contents [hide]
1 Overview of digital audio
2 Subjective evaluation
3 History of digital audio use in commercial recording
4 Digital audio technologies
5 Digital audio interfaces
6 References
7 See also

[edit] Overview of digital audio

Sampling and 4-bit quantization of an analogue signal (red) using Pulse Code Modulation.Digital audio is the method of representing audio in digital form.

An analog signal is converted to a digital signal at a given sampling rate and bit resolution; it may contain multiple channels (2 channels for stereo or more for surround sound). Generally speaking: the higher the sampling rate and bit resolution the more fidelity. Both systems introduce noise at the capturing stage, in analogue recording this is due to the noise floor of the circuit, and in digital recording due to quantization noise.

Quantization Noise (file info) — play in browser (beta)
An example of audio with progressively worsening quantization noise.
Problems listening to the file? See media help.

Much like an analog audio system, a digital audio system strives to reproduce the audio perfectly but neither can ultimately prevail. Analog systems have inherent capacitance and inductance which limit the bandwidth of the system and resistance limits the amplitude. Digital systems' sampling rate limits the bandwidth and bit resolution limits the dynamic range (resolution of amplitude creation). Both systems require increased cost and attention to achieve higher fidelity.

A digital audio signal starts with an analog-to-digital converter (ADC) that converts an analog signal to a digital signal. The ADC runs at a sampling rate and converts at a known bit resolution. For example, CD audio has a sampling rate of 44.1 kHz (44,100 samples per second) and 16-bit resolution for each channel (stereo). If the analog signal is not already bandlimited then an anti-aliasing filter is necessary before conversion, to prevent aliasing in the digital signal. (Aliasing occurs when frequencies above the Nyquist frequency have not been band limited, and instead appear as audible artifacts in the lower frequencies).

Some audio signals such as those created by digital synthesis originate entirely in the digital domain, in which case analog to digital conversion does not take place.

After being sampled with the ADC, the digital signal may then be altered in a process which is called digital signal processing where it may be filtered or have effects applied.

The digital audio signal may then be stored or transmitted. Digital audio storage can be on a CD, an iPod, a hard drive, USB flash drive, CompactFlash, or any other digital data storage device. Audio data compression techniques — such as MP3, Ogg Vorbis, or AAC — are commonly employed to reduce the size. Digital audio can be streamed to other devices.

The last step for digital audio is to be converted back to an analog signal with a digital-to-analog converter (DAC). Like ADCs, DACs run at a specific sampling rate and bit resolution but through the processes of oversampling, upsampling, and downsampling, this sampling rate may not be the same as the initial sampling rate.

[edit] Subjective evaluation
Fidelity evaluation is a long-standing issue with audio systems in general and introduction of lossy compression algorithms and psychoacoustic models have only increased debate.

Audio can be measured and analyzed to more exacting measures than can be done by ear, but what this technical measurement and analysis lacks is the ability to determine if it sounds "good" or "bad" to any given listener.[dubious — see talk page] Like any other human opinion, there are numerous parameters that widely vary between people that affect their subjective evaluation of what is good or bad. Such things that pertain to audio include hearing capabilities, personal preferences, location with respect to the speakers, and the room's physical properties.

This is not to say that subjective evaluation is unique to digital audio, digital audio can add to the fervor of discussion because it does introduce more things (e.g., lossy compression, psychoacoustic models) that can be debated.

[edit] History of digital audio use in commercial recording
Commercial digital recording of classical and jazz music began in the early 1970s, pioneered by Japanese companies such as Denon, the BBC, and British record label Decca (who in the mid-70s developed digital audio recorders of their own design for mastering of their albums), although experimental recordings exist from the 1960s. The first 16-bit PCM recording in the United States was made by Thomas Stockham at the Santa Fe Opera in 1976 on a Soundstream recorder. In most cases there was no mixing stage involved; a stereo digital recording was made and used unaltered as the master tape for subsequent commercial release. These unmixed digital recordings are still described as DDD since the technology involved is purely digital. (Unmixed analogue recordings are likewise usually described as ADD to denote a single generation of analogue recording.)

The first entirely digitally recorded (DDD) popular music album was Ry Cooder's Bop Till You Drop, recorded in late 1978. It was unmixed, being recorded straight to a two-track 3M digital recorder in the studio. Many other top recording artists were early adherents of digital recording. Others, such as former Beatles producer George Martin, felt that the multitrack digital recording technology of the early 1980s had not reached the sophistication of analogue systems. Martin used digital mixing, however, to reduce the distortion and noise that an analogue master tape would introduce (thus ADD). An early example of an analogue recording that was digitally mixed is Fleetwood Mac's 1979 release Tusk.

[edit] Digital audio technologies
DAB (Digital Audio Broadcasting)
Digital audio workstation
Digital audio player
Storage technologies:

Digital Audio Tape (DAT)
Compact disc (CD)
DVD DVD-A
MiniDisc
Super Audio CD
various audio file formats

[edit] Digital audio interfaces
Audio-specific interfaces include:

AC97 (Audio Codec 1997) interface between Integrated circuits on PC motherboards
ADAT interface
AES/EBU interface with XLR connectors
AES47, Professional AES3 digital audio over Asynchronous Transfer Mode networks
I²S (Inter-IC sound) interface between Integrated circuits in consumer electronics
MIDI low-bandwidth interconnect for carrying instrument data; cannot carry sound
S/PDIF, either over coaxial cable or TOSLINK
TDIF, Tascam proprietary format with D-sub cable
Naturally, any digital bus (e.g., USB, FireWire, and PCI) can carry digital audio.

Audio File Format

2007-06-30T04:42:00.000-07:00

An audio file format is a container format for storing audio data on a computer system. There are numerous file formats for storing audio data.

The general approach towards storing digital audio is to sample the audio voltage (which on playback, would correspond to a certain position of the membrane in a speaker) of the individual chanels with a certain resolution (the number of bits per sample) in regular intervals (forming the sample rate). This data can then be stored uncompressed or compressed to reduce the file size.

Contents [hide]
1 Types of formats
1.1 Uncompressed audio format
1.2 Lossless audio formats
1.3 Free and Open File Formats
1.4 Open File Formats
1.5 Proprietary Formats

[edit] Types of formats
It is important to distinguish between a file format and a codec. A codec performs the encoding and decoding of the raw audio data while the data itself is stored in a file with a specific audio file format. Though most audio file formats support only one audio codec, a file format may support multiple codecs, as AVI does.

There are three major groups of audio file formats:

Uncompressed audio formats, such as WAV, AIFF and AU;
formats with lossless compression, such as FLAC, Monkey's Audio (filename extension APE), WavPack (filename extension WV), Shorten, TTA, Apple Lossless and lossless Windows Media Audio (WMA); and
formats with lossy compression, such as MP3, Vorbis, lossy Windows Media Audio (WMA) and AAC.

[edit] Uncompressed audio format
There is one major uncompressed audio format, PCM, which is usually stored as a .wav on Windows or as .aiff on Mac OS. WAV is a flexible file format designed to store more or less any combination of sampling rates or bitrates. This makes it an adequate file format for storing and archiving an original recording. A lossless compressed format would require more processing for the same time recorded, but would be more efficient in terms of space used. WAV, like any other uncompressed format, encodes all sounds, whether they are complex sounds or absolute silence, with the same number of bits per unit of time.

Let's take an example. A file contains a minute of a symphonic orchestra playing beautifully followed by a minute of silence. If the sound were stored in WAV, the same amount of data would be used for each half. If data were encoded with TTA, the first minute would be a bit smaller than in the WAV file, and the silent half would take almost no disc space at all. But then, recording in the TTA format would require a lot more processing than the WAV.

The WAV format is based on the RIFF file format, which is similar to the IFF format.

BWF (Broadcast Wave Format) is a standard audio format created by the European Broadcasting Union as a successor to WAV. BWF allows metadata to be stored in the file. See: European Broadcasting Union: Specification of the Broadcast Wave Format - A format for audio data files in broadcasting. EBU Technical document 3285, July 1997. This format is the primary recording format used in many professional Audio Workstations used in the Television and Film industry. Stand-alone file based multi-track recorders from Sound Devices, Zaxcom, HHB USA, Fostex, and Aaton all use BWF as their preferred file format for recording multi-track audio files with SMPTE Time Code reference. This standardized Time Stamp in the Broadcast Wave File allows for easy synchronization with a separate picture element.

[edit] Lossless audio formats
Lossless audio formats (such as TTA and FLAC) provide a compression ratio of about 2:1.

[edit] Free and Open File Formats
wav - standard audio file format used mainly in Windows PCs. Commonly used for storing uncompressed (PCM), CD-quality sound files, which means that they can be large in size - around 10MB per minute of music. It is less well known that wave files can also be encoded with a variety of codecs to reduce the file size (for example the GSM or mp3 codecs). Wav files use a RIFF structure.
ogg - a free, open source container format supporting a variety of codecs, the most popular of which is the audio codec Vorbis. Vorbis offers better compression than MP3 but is less popular.
flac - a lossless compression codec. You can think of lossless compression as like zip but for audio. If you compress a PCM file to flac and then restore it again it will be a perfect copy of the original. (All the other codecs discussed here are lossy which means a small part of the quality is lost). The cost of this losslessness is that the compression ratio is not good. Flac is recommended for archiving PCM files where quality is important (eg. broadcast or music use).
aiff - the standard audio file format used by Apple. It is like a wav file for the Mac.
raw - a raw file can contain audio in any codec but is usually used with PCM audio data. It is rarely used except for technical tests.
au - the standard audio file format used by Sun, Unix and Java. The audio in au files can be PCM or compressed with the ulaw, alaw or G729 codecs.

[edit] Open File Formats
mp3 - the MPEG Layer-3 format is the most popular format for downloading and storing music. By eliminating portions of the audio file that are essentially inaudible, mp3 files are compressed to roughly one-tenth the size of an equivalent PCM file while maintaining good audio quality. The mp3 format is recommended for music storage. It is not that good for voice storage.
gsm - designed for telphony use in Europe, gsm is a very practical format for telephone quality voice. It makes a good compromise between file size and quality. Note that wav files can also be encoded with the gsm codec.
dct - A variable codec format designed for dictation. It has dictation header information and can be encrypted (often required by medical confidentiality laws).
vox - the vox format most commonly uses the Dialogic ADPCM (Adaptive Differential Pulse Code Modulation) codec. Similar to other ADPCM formats, it compresses to 4-bits. Vox format files are similar to wave files except that the vox files contain no information about the file itself so the codec sample rate and number of channels must first be specified in order to play a vox file.
aac - the Advanced Audio Coding format is based on the MPEG2 and MPEG4 standards. aac files are usually ADTS or ADIF containers.
mp4/m4a - MPEG-4 audio most often AAC but sometimes MP2/MP3

[edit] Proprietary Formats
wma - the popular Windows Media Audio format owned by Microsoft. Designed with Digital Rights Management (DRM) abilities for copy protection.
atrac (.wav) - the older style Sony ATRAC format. It always has a .wav file extension. To open these files simply install the ATRAC3 drivers.
ra - a Real Audio format designed for streaming audio over the Internet. The .ra format allows files to be stored in a self-contained fashion on a computer, with all of the audio data contained inside the file itself.
ram - a text file that contains a link to the Internet address where the Real Audio file is stored. The .ram file contains no audio data itself.
dss - Digital Speech Standard files are an Olympus proprietary format. It is a fairly old and poor codec. Prefer gsm or mp3 where the recorder allows.
msv - a Sony proprietary format for Memory Stick compressed voice files.
dvf - a Sony proprietary format for compressed voice files; commonly used by Sony dictation recorders.
m4p - A proprietary version of AAC in MP4 with Digital Rights Management developed by Apple for use in music downloaded from their iTunes Music Store.
Retrieved from "http://en.wikipedia.org/wiki/Audio_file_format"

Audio Format

2007-06-30T04:40:00.000-07:00

An audio format is a medium for storing sound and music. The term is applied to both the physical recording media and the recording formats of the audio content – in computer science it is often limited to the audio file format, but its wider use usually refers to the physical method used to store the data.

Music is recorded and distributed using a variety of audio formats, some of which store additional information.

[edit] Timeline of audio format developments
Year Media formats Recording formats
1877 Phonograph cylinder Mechanical analog; "hill-and-dale" grooves, vertical stylus motion
1883 Music roll Mechanical digital (automated musical instruments)
1895 Gramophone record Mechanical analog; lateral grooves, horizontal stylus motion
1898 Wire recording Analog; magnetization; no "bias"
1925 Electrical cut record Mechanical analog; electrically cut from amplified microphone signal, lateral grooves, horizontal stylus motion, discs at 7", 10", 12", most at 78 rpm
1930s Reel-to-Reel, Magnetic Tape Analog; magnetization; "bias" dramatically increases linearity/fidelity, tape speed at 30 ips, later 15 ips with NAB equalization; refined speeds: 7 1/2 ips, 3 3/4 ips, 1 7/8 ips
1930s Electrical transcriptions Mechanical analog; electrically cut from amplified microphone signal, high fidelity sound, lateral or vertical grooves, horizontal or vertical stylus motion, most discs 16" at 33 1/3 rpm
1948 (Commercial release) Vinyl Record Analog, with preemphasis and other equalization techniques (LP, RIAA); lateral grooves, horizontal stylus motion; discs at 7" (most 45 rpm), 10" and 12" (most 33 1/3 rpm)
1957 Stereophonic Vinyl Record Analog, with preemphasis and other equalization techniques. Combination lateral/vertical stylus motion with each channel encoded 45 degrees to the vertical.
1962 4-Track Analog, 1/4 inch wide tape, 3 3/4 inches/sec, endless loop cartridge.
1963 Compact Cassette Analog, with bias, preemphasis, 0.15 inch wide tape, 17/8 inches/sec. 1970: introduced Dolby noise reduction.
1965 8-Track Analog, 1/4 inch wide tape, 3 3/4 inches/sec, endless loop cartridge.
1969 Microcassette Analog, 1/8 inch wide tape, used generally for notetaking, mostly mono, some stereo. 2.4 cm/s or 1.2 cm/s.
1969 Minicassette Analog, 1/8 inch wide tape, used generally for notetaking, 1.2 cm/s
1970 Quadraphonic 8-Track (Q8) Analog, 1/4 inch wide tape, 3 3/4 inches/sec, 4 Channel Stereo, endless loop cartridge.
1971 Quadraphonic Vinyl Record (CD-4) (SQ Matrix)
1975 Betamax Digital Audio 'Dolby Stereo' cinema surround sound
1976 Elcaset
1978 Laserdisc
1982 Compact Disc (CD-DA) PCM
1985 Audio Interchange File Format (AIFF)
1985 Sound Designer (by Digidesign)
1987 Digital Audio Tape (DAT)
1991 MiniDisc (MD) ATRAC
1992 Digital Compact Cassette (DCC)
1992 WAVEform (WAV)
Dolby Digital surround cinema sound

1993 Digital Theatre System (DTS)
Sony Dynamic Digital Sound (SDDS)

1995 MP3
1997 DVD Dolby Digital
1997 DTS-CD DTS Audio
1999 DVD-Audio
1999 Super Audio CD (SACD)
1999 Windows Media Audio (WMA)
1999 The True Audio Lossless Codec (TTA)
2000 Free Lossless Audio Codec (FLAC)
2001 Advanced audio coding (AAC)
2002 Ogg Vorbis
2003 DualDisc
2004 Apple Lossless (ALE or ALAC)
2005 HD-DVD
2006 Blu-Ray

Home Cinema

2007-06-30T04:34:00.001-07:00

A 3 metres/119 inch projection screen with a high-definition television image.
A home theater with video projector mounted in a box on the ceiling.
Built-in shelves provide a place for movie decor, DVDs, and equipment. Note the component stack on the right, where the audio receiver, DVD player, secondary monitor, and video game system are located.
The same projection screen as at top, without image.Home cinema, also called home theatre, seeks to reproduce cinema quality video and audio in the home.

Technically, a home cinema could be as basic as a simple arrangement of a television, DVD, and a set of speakers. It is therefore difficult to specify exactly what distinguishes a "home cinema" from a "television and stereo". Most people in the consumer electronics industry would agree that a "home theater" is really the integration of a relatively high-quality video output with surround sound.

Contents [hide]
1 Design
2 Component systems vs. Theater-in-a-Box
3 Dedicated home theaters
4 Backyard theater
5 History
5.1 1950s and 1960s home movies
5.2 1980s home cinema
5.3 1990s home cinema
6 See also
7 References

[edit] Design
Today, "home cinema" implies a real "cinema experience" and therefore a higher quality set of components than the average television provides. A typical home theater includes the following parts:

Input Devices: One or more audio/video sources. High quality formats such as HD DVD or Blu-ray are preferred, though they often include a VHS player or Video Game Systems. Some home theatres now include a home theater PC to act as a library for video and music content.
Processing Devices: Input devices are processed by either a standalone AV receiver or a Preamplifier and Sound Processor for complex surround sound formats. The user selects the input at this point before it is forwarded to the output.
Audio Output: Systems consist of at least 2 speakers, but can have up to 11 with additional subwoofer.
Video Output: A large HDTV display. Options include Liquid crystal display television, video projector, plasma TV, rear-projection TV, or a traditional CRT TV.
Atmosphere: Comfortable seating and organization to improve the cinema feel. Higher end home theaters commonly also have sound insulation to prevent noise from escaping the room, and a specialized coating to ensure correct absorption of the sound in the room.

[edit] Component systems vs. Theater-in-a-Box
High-quality home cinemas are assembled from component pieces purchased separately to provide the best combination of equipment for the cost. It is possible to purchase home theater in a box kits that include a set of speakers for surround sound, an amplifier/tuner for adjusting volume and selecting video sources, and sometimes a DVD player. Though these kits often pale in comparison to a custom-built home cinema, they are inexpensive and easy to set up; one needs only to add a television and some movies in order to create a simple home theater.

[edit] Dedicated home theaters
Some home cinema enthusiasts go so far as to build a dedicated room in the home for the theater. These more advanced installations often include sophisticated acoustic design elements, including "room-in-a-room" construction that isolates sound and provides the potential for a nearly ideal listening environment. These installations are often designated as "screening rooms" to differentiate from simpler installations. This idea can go as far as completely recreating an actual cinema, with a projector enclosed in a projection booth, specialized furniture, a piano or theatre organ, curtains in front of the projection screen, movie posters, or a popcorn or snack machine. More commonly, real dedicated home theatres pursue this to a lesser degree.

[edit] Backyard theater
In places that have the proper outdoor atmosphere, it is possible for people to set up a home theater in their backyard. Depending on the space available, it may simply be a temporary version with foldable screen, a projector and couple of speakers, or a permanent fixture with huge screens and dedicated audio set up poolside. Due to the outdoor nature, it is quite popular with BBQ parties and pool parties.

Some people have built upon the idea, and constructed mobile drive-in theaters that can play movies in public open spaces. Usually, these require a powerful projector, a laptop or DVD player, outdoor speakers and/or an FM transmitter to broadcast the audio to other car radios.[1][2]

[edit] History

[edit] 1950s and 1960s home movies
In the 1950s, home movies became popular in the United States and elsewhere as Kodak 8 mm film (Pathé 9.5 mm in France) and camera and projector equipment became affordable. Projected with a small, portable movie projector onto a portable screen, often without sound, this system became the first practical home theater. They were generally used to show home movies of family travels and celebrations but also doubled as a means of showing private stag films. Dedicated home cinemas were called screening rooms at the time and were outfitted with 16 mm or even 35 mm projectors for showing commercial films. These were found almost exclusively in the homes of the very wealthy, especially those in the movie industry.

Portable home cinemas improved over time with color film, Kodak Super 8 mm film film cartridges, and monaural sound but remained awkward and somewhat expensive. The rise of home video in the late 1970s almost completely killed the consumer market for 8 mm film cameras and projectors, as VCRs connected to ordinary televisions provided a simpler and more flexible substitute.

[edit] 1980s home cinema
The development of multi-channel audio systems and laserdisc in the 1980s created a new paradigm for home cinema. The first known home cinema system was installed as a sales tool by Steve LaFontaine at Kirshmans furniture store 5800 Veterans Memorial Highway. In Metairie, Louisiana, in 1974 where he built a special sound room which incorporated the earliest quadraphonic audio systems with his video projection systems he invented and hand built by modifying Sony trinitron televisions for projecting the image. Many systems were sold in the New Orleans area in the ensuing years before the first public demonstration of this integration occurred in 1982 at the Summer Consumer Electronics Show in Chicago, Illinois. Peter Tribeman of NAD (USA) organized and presented a demonstration made possible by the collaborative effort of NAD, Proton, ADS, Lucasfilm and Dolby Labs who contributed their technologies to demonstrate what a home cinema would "look and sound" like.

Over the course of three days, retailers, manufacturers, and members of the consumer electronics press were exposed to the first "home like" experience of combining a high quality video source with multi-channel surround sound. That one demonstration is credited with being the impetus for developing what is now a multi-billion dollar business.

[edit] 1990s home cinema
In the late 1990s, the development of DVD, 5-channel audio, and high-quality video projectors that provide a cinema experience at a price that rivals a big-screen HDTVs sparked a new wave of home cinema interest.

High Fidelity (Hi-Fi)

2007-06-30T04:32:00.000-07:00

This article is about audiophile sound systems. For other uses, see High Fidelity.
High fidelity or hi-fi reproduction is a term used by home stereo listeners and home audio enthusiasts (audiophiles) to refer to high-quality reproduction of sound or images that is very faithful to the original master recording. High fidelity equipment has minimal or unnoticeable amounts of noise and distortion and an accurate frequency response as set out in 1973 by the German Deutsches Institut für Normung (DIN) standard DIN 45500. The term was most widely used in this strict sense in the 1950s and 1960s; in subsequent decades, the term was applied more loosely to any mid-level stereo system. In the 2000s, the term "hi-fi" for expensive high quality home audio electronics was largely replaced with the term "high-end audio".

Contents [hide]
1 History
2 Ascertaining high fidelity: double-blind tests
3 Semblance of realism
4 Modularity
5 Modern equipment
6 See also
7 External links

[edit] History
The 1920s saw the introduction of electronic amplification, microphones, and the application of quantitative engineering principles to the reproduction of sound. Much of the pioneering work was done at Bell Laboratories and commercialized by Western Electric. Acoustically-recorded disc records with capriciously peaky frequency response were replaced with electrically recorded records. The Victor Orthophonic phonograph, although purely acoustic, was created by engineers who applied waveguide technology to the design of the interior folded horn to produce a smooth frequency response which complemented and equalled that of the electrically recorded Victor Orthophonic records.

Meanwhile, the rise of radio meant increased popularity for loudspeakers and tube amplifiers, so there was an anomaly of a period of time during which radio receivers commonly used loudspeakers and electronic amplifiers to produce sound, while phonographs were still commonly purely mechanical and acoustic. Later, electronic phonographs became available, as stand-alone units or designed to play through consumer's radios. The now ubiquitous RCA connector was first introduced by the Radio Corporation of America for this purpose.

The development of Sound film in the 1930s led motion picture companies to develop amplification and loudspeaker systems to fill movie theaters with good quality sound at a reasonable volume. To achieve this result, they employed loudspeakers with separate sections for low and high frequencies ("woofers" and "tweeters"), connected via an audio crossover network, and more carefully engineered enclosures. This development exposed the public to better fidelity than home equipment was capable of at the time. Some movie stars purchased movie theater sound equipment for use in their homes but the cost was out of reach for anyone of modest means.

After World War II, several innovations created the conditions for a major improvement of home-audio quality:

Reel-to-reel audio tape recording, based on technology found in Germany after the war, helped musical artists such as Bing Crosby make and distribute recordings with better fidelity.
the advent of the 33⅓ RPM Long Play (LP) microgroove vinyl record, with low surface noise and quantitatively-specified equalization curves. Classical music fans, who were opinion leaders in the audio market quickly adopted LPs because, unlike with older records, most classical works would fit on a single LP.
FM radio, with wider audio bandwidth and less susceptibility to signal interference and fading than AM radio.
better amplifier designs, with more attention to frequency response and much higher power output capability, allowing audio peaks to be reproduced without distortion.
In the 1950s, the term high fidelity began to be used by audio manufacturers as a marketing term to describe records and equipment which were intended to provide faithful sound reproduction. While some consumers simply interpreted high fidelity as fancy and expensive equipment, many found the difference in quality between "hi-fi" and the then standard AM radios and 78 RPM records readily apparent and bought 33 LPs, such as RCA's New Orthophonics and London's ffrrs, and high-fidelity phonographs. Audiophiles paid attention to technical characteristics and bought individual components, such as separate turntables, radio tuners, preamplifiers, power amplifiers and loudspeakers. Some enthusiasts assembled their own loudspeaker systems. In the 1950s, hi-fi became a generic term, to some extent displacing phonograph and record player. Rather than "playing a record on the phonograph", people would "play it on the hi-fi".

In the late 1950s and early 1960s, the development of the Westrex single-groove stereophonic record led to the next wave of home-audio improvement, and in common parlance, stereo displaced hi-fi. Records were now played on a stereo. In the world of the audiophile, however, high fidelity continued and continues to refer to the goal of highly-accurate sound reproduction and to the technological resources available for approaching that goal. A very popular type of system for reproducing music from the 1970s onwards is the integrated music centre--the successor to the older stereogram or radiogram. Purists will generally avoid referring to these systems as high fidelity, though some are capable of very good quality sound reproduction.

[edit] Ascertaining high fidelity: double-blind tests
Double-blind testing has been required in the approval of new medicines since about 1960. Although single-blind testing of loudspeakers had been used for a number of years by Floyd E. Toole at the National Research Council of Canada, the double-blind audio listening test of amplifiers was first described in the United States by Daniel J. Shanefield in November of 1974 in the newsletter of the Boston Audio Society. This was later reported to the general public in High Fidelity magazine, March 1980. The double-blind listening comparison is now a standard procedure with almost all audio professionals respected in their field. For marketing purposes, a few manufacturers of very expensive audio equipment dispute the need for this test. A commonly-used variant of this test is the ABX test. This involves comparing two known audio sources (A and B) with either one of these when it has been randomly selected (X). The test and its associated equipment was developed by the Southeastern Michigan Woofer and Tweeter Marching Society (SMWTMS) — a semi-professional organization in Detroit that is very active in the double-blind testing of new audio components.

An alternative view is that such testing is stressful, and perhaps because of this, is unable to distinguish the fine subtleties of top equipment; that only long-term listening will allow one to get to grips with its true sound — furthermore that proponents of double-blind testing have an agenda to discredit that such subtle differences exist, to claim that critics of double-blind testing are purely self-delusionary and victims of advertising hype. However, there is still another level of argument that maintains that all serious listening comparisons can be stressful. Also, listeners who paid an unusually large price for playback equipment might have a subconscious tendency to favor it. Therefore most professional audio testing uses double-blind comparisons.

Nevertheless, the double-blind methodology does not rule out a long-term test conducted at leisure in comfortable situations.

[edit] Semblance of realism
When high fidelity was limited to monophonic sound reproduction, a realistic approximation to what the listener would experience in a concert hall was limited. Researchers early realized that the ideal way to experience music played back on audio equipment was through multiple transmission channels, but the technology was not available at that time. It was, for example, discovered that a realistic representation of the separation between performers in an orchestra from an ideal listening position in the concert hall would require at least three loudspeakers for the front channels. For the reproduction of the reverberation, at least two loudspeakers placed behind or to the sides of the listener were required.

Stereophonic sound provided a partial solution to the problem of creating some semblance of the illusion of performers performing in an orchestra by creating a phantom middle channel when the listener sits exactly in the middle of the two front loudspeakers. When the listener moves slightly to the side, however, this phantom channel disappears or is greatly reduced. An attempt to provide for the reproduction of the reverberation was tried in the 1970s through quadraphonic sound but, again, the technology at that time was insufficient for the task. Consumers did not want to pay the additional costs required in money and space for the marginal improvements in realism. With the rise in popularity of home theatre, however, multi-channel playback systems became affordable, and consumers were willing to tolerate the six to eight channels required in a home theatre. The advances made in signal processors to synthesize an approximation of a good concert hall can now provide a somewhat more realistic illusion of listening in a concert hall.

In addition to spatial realism, the playback of music must be subjectively free from noise to achieve realism. The compact disc (CD) provides at least 90 decibels of dynamic range, which is about as much as most people can tolerate in an average living room. This therefore requires the playback equipment to provide a signal-to-noise ratio of at least 90 decibels. Many people can hear up to, at most, 15 kHz and for a few, up to 20 kHz. There is relatively little music below 50 Hz, loud bass below 30 Hz is rare, music below 16 Hz is almost non-existent, and music below 5 Hz is probably non-existent. (Incidentally, the cannons in Telarc's recording of Pyotr Tchaikovsky's 1812 Overture are said to go down to 5 Hz.) The equipment must also provide no noticeable distortion of the signal or emphasis or de-emphasis of any frequency in this frequency range. Except for spatial realism, good modern equipment can easily satisfy all of these requirements at a relatively moderate cost.

[edit] Modularity
Integrated, midi, or lifestyle systems contain one or more sources such as a CD player, a tuner, or a cassette deck together with a preamplifier and a power amplifier in one box. (Midi has no connection with MIDI technology in electronic instruments.) Such products are generally disparaged by audiophiles, although some high-end manufacturers do produce integrated systems. The traditional hi-fi enthusiast, however, will build a system from separates, often with each item from a different manufacturer specialising in a particular component. This provides the most flexibility for piece-by-piece upgrades.

For slightly less flexibility in upgrades, a preamplifier and a power amplifier in one box is called an integrated amplifier; with a tuner, it is a receiver. A monophonic power amplifier , which is called a monoblock, is often used for powering a subwoofer. Other modules in the system may include components like cartridges, tonearms, turntables, Digital Media Players, DVD players that play a wide variety of discs including CDs, CD recorders, MiniDisc recorders, hi-fi video-cassette recorders (VCRs), reel-to-reel recorders, equalizers, signal processors, and subwoofers.

This modularity allows the enthusiast to spend as little or as much as he wants on a component that suits his specific needs. In a system built from separates, sometimes a failure on one component still allows partial use of the rest of the system. A repair of an integrated system, though, means complete lack of use of the system. Another advantage of modularity is the ability to spend one's money on only a few core components at first and then later add additional components to one's system. Because of all these advantages to the modular way of building a high-fidelity system instead of buying an integrated system, audiophiles almost always assemble their system from separates. Some of the obvious disadvantages of this approach are increased cost, complexity, and space required for the components, not to mention the possibility of introducing noise via the interconnects between components.

[edit] Modern equipment
Modern hi-fi equipment usually includes digital audio signal sources such as CD players, Digital Audio Tape (DAT) and Digital Audio Broadcasting (DAB) or HD Radio tuners, an amplifier, and loudspeakers. Some modern hi-fi equipment can be digitally connected using fiber optic TOSLINK cables, universal serial bus (USB) ports (including one to play MP3 and Ogg files in an USB flash drive), or WiFi support.

One modern component that is making fast gains in acceptance is the music server consisting of one or more computer hard drives that hold music in the form of computer files. When the music is stored in an audio file format that is lossless (such as FLAC or Monkey Audio), unlike lossy file formats such as MP3, WMA, and Ogg (which all suffer from fidelity-degradation), the computer playback of recorded audio can indeed serve as an audiophile-quality source for a hi-fi system. However, it should be noted that lossy audio formats are not hi-fi in the stricter sense of the term. Some non-lossy file formats are AIFF and WAV; both are suitable for high fidelity audio, in a variety of resolutions. Resolutions which exceed CD quality are capable with such files and appropriate playback equipment (professional or semipro digital to analog converters).

If the hi-fi system includes components such as a projector, television, satellite decoder, DVD player, surround sound amplification and multi-channel loudspeakers, then it is often called home cinema or a home theatre system.

MP3 CD

2007-06-30T04:31:00.000-07:00

MP3 CD is a term used to refer to compact discs (CD-R or CD-RW) that contain MP3 files. Discs are burned in data mode, as opposed to Red Book format as with standard audio CDs.

MP3 CDs are supported by several modern CD players. There are also CD players capable of playing wma and Ogg files, and on Sony branded players, audio encoded to their ATRAC format.

The BBC has released some of its CD back-catalogue on MP3 CDs, including compilations of Doctor Who audiobooks and television soundtracks and the entire Hitchhiker's Guide to the Galaxy radio series. These discs retail at a higher price than typical audio CDs, but contain up to 25 hours of audio on a single disc.

Digital Video Disc (DVD)

2007-06-30T04:29:00.000-07:00

Media type: optical disc
Capacity: 4.7 GB (single layer), 8.5 GB (dual layer)
Usage: Data storage, video

DVD ("Digital Versatile Disc" or "Digital Video Disc") is an optical disc storage media format that can be used for data storage, including movies with high video and sound quality. DVDs resemble Compact Discs in that they have the exact appearance (i.e. diameter: 120mm or 4.72in., occasionally 80mm or 3.15in.) and both are optical storage media so similar that a DVD reader or writer can usually read CDs, but DVDs are encoded in a different format of much greater density, allowing a data storage capacity 8 times greater (single-layer, single-sided).

All read-only DVD discs, regardless of type, are DVD-ROM discs. This includes replicated (factory pressed), recorded (burned), video, audio, and data DVDs. A DVD with properly formatted and structured video content is a DVD-Video disc. DVDs with properly formatted and structured audio content are DVD-Audio discs. Everything else, (including other types of DVD discs with video content) is referred to as a DVD-Data disc. Consumers use the term "DVD-ROM" to refer to pressed data discs only, but that is grammatically incorrect, moreover, the term DVD also is applied generically in describing newer video disc formats, Blu-ray Disc and HD DVD.

Contents [hide]
1 History
1.1 Etymology
2 DVD disc capacity
2.1 Capacity nomenclature
3 DVD recordable and rewriteable
4 Dual layer recording
5 DVD-Video
6 DVD-Audio
6.1 Security
7 Competitors and successors
8 See also
9 References
10 External links
10.1 Official
10.2 Quality guide
10.3 Knowledge

[edit] History

Optical disc authoring
Optical disc
Optical disc image
Recorder hardware
Authoring software
Recording technologies
Recording modes
Packet writing

Optical media types
Laserdisc
Compact Disc/CD-ROM: CD-R, CD-RW
MiniDisc
DVD: DVD-R, DVD-R DL, DVD+R,
DVD+R DL, DVD-RW, DVD+RW,
DVD-RW DL, DVD+RW DL, DVD-RAM
Blu-ray Disc: BD-R, BD-RE
HD DVD: HD DVD-R: HD DVD-RAM
UDO
UMD
Holographic data storage
3D optical data storage
History of optical storage media

Standards
Rainbow Books
File systems
ISO 9660
Joliet
Rock Ridge
Amiga Rock Ridge extensions
El Torito
Apple ISO9660 Extensions
Universal Disk Format
Mount Rainier

Size comparison: A 12cm Sony DVD+RW and a 19cm Dixon Ticonderoga pencil.In the early 1990s two high-density optical storage standards were being developed; one was the MultiMedia Compact Disc, backed by Philips and Sony, and the other was the Super Density disc, supported by Toshiba, Time-Warner, Matsushita Electric, Hitachi, Mitsubishi Electric, Pioneer, Thomson, and JVC. IBM's president, Lou Gerstner, acting as a matchmaker, led an effort to unite the two camps behind a single standard, anticipating a repeat of the costly videotape format war between VHS, Betamax and Video 2000 in the 1980s.

Philips and Sony abandoned their MultiMedia Compact Disc and fully agreed upon Toshiba's SuperDensity Disc with only one modification, namely changing to EFMPlus modulation. EFMPlus was chosen as it has a great resilience against disc damage such as scratches and fingerprints. EFMPlus, created by Kees Immink, who also designed EFM, is 6% less efficient than the modulation technique originally used by Toshiba, which resulted in a capacity of 4.7 GB as opposed to the original 5 GB. The result was the DVD specification, finalized for the DVD movie player and DVD-ROM computer applications in December of 1995.[1] In May 1997, the DVD Consortium was replaced by the DVD Forum, which is open to all other companies.

[edit] Etymology
"DVD" was originally used as an initialism for the unofficial term "digital videodisk".[2] It was reported in 1995, at the time of the specification finalization, that the letters officially stood for "digital versatile disc" (due to non-video applications)[3], however, the text of the press release announcing the specification finalization only refers to the technology as "DVD", making no mention of what (if anything) the letters stood for.[1] A newsgroup FAQ written by Jim Taylor (a prominent figure in the industry) claims that four years later, in 1999, the DVD Forum stated that the format name was simply the three letters "DVD" and did not stand for anything.[4] The official DVD specification documents have never defined DVD. Usage in the present day varies, with "DVD", "Digital Video Disc", and "Digital Versatile Disc" all being common.

[edit] DVD disc capacity
Single layer capacity Dual/Double layer capacity
Physical size GB GiB GB GiB
12 cm, single sided 4.7 4.38 8.5 7.92
12 cm, double sided 9.4 8.75 17.1 15.93
8 cm, single sided 1.4 1.30 2.6 2.42
8 cm, double sided 2.8 2.61 5.2 4.84

Note: GB here means gigabyte, equal to 109 (or 1,000,000,000) bytes. Many programs will display gibibyte (GiB), equal to 230 (or 1,073,741,824) bytes.

Example: A disc with 8.5 GB capacity is equivalent to: (8.5 × 1,000,000,000) / 1,073,741,824 ≈ 7.92 GiB.

Size Note: There is a difference in size between + and - DL DVD formats. For example, the 12 cm single sided disk has capacities:

Disk Type Sectors bytes GB GiB
DVD-R SL 2,298,496 4,707,319,808 4.7 4.384
DVD+R SL 2,295,104 4,700,372,992 4.7 4.378
DVD-R DL 4,171,712 8,543,666,176 8.5 7.957
DVD+R DL 4,173,824 8,547,991,552 8.5 7.961

[edit] Capacity nomenclature
The four basic types of DVD are referred to by their capacity in gigabytes, rounded up to the nearest integer.

DVD type Name
Single sided, single layer DVD-5
Single sided, dual layer DVD-9
Double sided, single layer DVD-10
Double sided, dual layer DVD-18

Another format in limited use is a double sided DVD with one side comprising a single layer of data while the opposite side comprises two layers of data (effectively a DVD-5 on one side bonded to a DVD-9 on the other). This format holds approximately 13.2 GB of data and is known as DVD-14.[5]

[edit] DVD recordable and rewriteable
Main article: DVD recordable
HP initially developed recordable DVD media from the need to store data for back-up and transport.

DVD recordables are now also used for consumer audio and video recording. Three formats were developed: -R/RW (minus/dash), +R/RW (plus), -RAM (which is strictly speaking not random access memory).

[edit] Dual layer recording
Dual Layer recording allows DVD-R and DVD+R discs to store significantly more data, up to 8.5 Gigabytes per disc, compared with 4.7 Gigabytes for single-layer discs. DVD-R DL was developed for the DVD Forum by Pioneer Corporation, DVD+R DL was developed for the DVD+RW Alliance by Philips and Mitsubishi Kagaku Media (MKM). [6]

A Dual Layer disc differs from its usual DVD counterpart by employing a second physical layer within the disc itself. The drive with Dual Layer capability accesses the second layer by shining the laser through the first semi-transparent layer. The layer change mechanism in some DVD players can show a noticeable pause, as long as two seconds by some accounts. This caused more than a few viewers to worry that their dual layer discs were damaged or defective, with the end result that studios began listing a standard message explaining the dual layer pausing effect on all dual layer disc packaging.

DVD recordable discs supporting this technology are backward compatible with some existing DVD players and DVD-ROM drives.[7] Many current DVD recorders support dual-layer technology, and the price point is comparable to that of single-layer drives, though the blank media remain significantly more expensive.

[edit] DVD-Video
Main article: DVD-Video
DVD-Video is a standard for storing video content on DVD media. In the US, weekly DVD-Video rentals first out-numbered weekly VHS cassette rentals in June 2003, illustrating the rapid adoption rate of the technology in the marketplace.[8]

Though many resolutions and formats are supported, most consumer DVD-Video disks use either 4:3 or 16:9 aspect ratio MPEG-2 video, stored at a resolution of 720×480 (NTSC) or 720×576 (PAL). Audio is commonly stored using the Dolby Digital (AC-3) and/or Digital Theater System (DTS) formats, ranging from monaural to 5.1 channel "Surround Sound" presentations. DVD-Video also supports features like selectable subtitles, multiple camera angles and multiple audio tracks.

[edit] DVD-Audio
Main article: DVD-Audio
DVD-Audio is a format for delivering high-fidelity audio content on a DVD. It offers many channel configuration options (from mono to 5.1 surround sound) at various sampling frequencies and sample rates. Compared with the CD format, the much higher capacity DVD format enables the inclusion of either considerably more music (with respect to total running time and quantity of songs) or far higher audio quality (reflected by higher linear sampling rates and higher vertical bit-rates, and/or additional channels for spatial sound reproduction).

Despite DVD-Audio's superior technical specifications, there is debate as to whether the resulting audio enhancements are distinguishable to typical human ears. DVD-Audio currently forms a niche market, probably due to its dependency upon new and relatively expensive equipment.

[edit] Security
Main article: CPRM
DVD-Audio discs employ a robust copy prevention mechanism, called Content Protection for Prerecorded Media (CPPM) developed by the 4C group (IBM, Intel, Matsushita, and Toshiba).

To date, CPPM has not been "broken" in the sense that DVD-Video's CSS has been broken, but ways to circumvent it have been developed.[9] By modifying commercial DVD(-Audio) playback software to write the decrypted and decoded audio streams to the hard disk, users can, essentially, extract content from DVD-Audio discs much in the same way they can from DVD-Video discs.

[edit] Competitors and successors
There are several possible successors to DVD being developed by different consortiums: Sony/Panasonic's Blu-ray Disc (BD), Toshiba's HD DVD and Maxell's Holographic Versatile Disc (HVD).

In April 2000, Sonic Solutions and Ravisent announced hDVD, an HDTV extension to DVD that presaged the HD formats that debuted 6 years later.[10]

On November 19, 2003, the DVD Forum decided by a vote of eight to six that HD DVD will be its official HDTV successor to DVD. This had no effect on the competing Blu-ray Disc Association's (BDA) determination that its format would succeed DVD, especially since most of the voters belonged to both groups.[citation needed]

On April 15, 2004, in a co-op project with TOPPAN Printing Co., the electronics giant Sony Corp. successfully developed the paper disc, a storage medium that is made out of 51% paper and offers up to 25 GB of storage, about five times more than the standard 4.7 GB DVD. The disc can be easily cut with scissors and recycled offering an environmentally friendly storage media.

As reported in a mid 2005 issue of Popular Mechanics, it is not yet clear which technology will win the format war over DVD. HD DVD discs have a lower capacity than Blu-ray Discs (15 GB vs. 25 GB for single layer, 30 GB vs. 50 GB for dual layer). Other speculations as to which format will win include Blu-ray Disc's larger hardware vendor and movie studio support, and HD DVD's faster read times.

This situation—multiple new formats fighting as the successor to a format approaching purported obsolescence—previously appeared as the "war of the speeds" in the record industry of the 1950s. It is also similar to the VHS/Betamax war in consumer video recorders in the late 1980s.

The new generations of optical formats have restricted access through many various digital rights management schemes such as AACS and HDCP; it remains to be seen what impact the limitation of fair use rights has on their adoption in the marketplace.

Compact Disc

2007-06-30T04:26:00.000-07:00

The closely spaced tracks on the readable surface of a Compact Disc cause light to diffract into a full visible colour spectrum
Media type: Optical disc
Encoding: various
Capacity: up to 700 MB
Read mechanism: 780 nm wavelength semiconductor laser
Developed by: Sony & Philips
Usage: Audio and data storage

Optical disc authoring
Optical disc
Optical disc image
Recorder hardware
Authoring software
Recording technologies
Recording modes
Packet writing

Optical media types
Laserdisc
Compact Disc/CD-ROM: CD-R, CD-RW
MiniDisc
DVD: DVD-R, DVD-R DL, DVD+R,
DVD+R DL, DVD-RW, DVD+RW,
DVD-RW DL, DVD+RW DL, DVD-RAM
Blu-ray Disc: BD-R, BD-RE
HD DVD: HD DVD-R: HD DVD-RAM
UDO
UMD
Holographic data storage
3D optical data storage
History of optical storage media

Standards
Rainbow Books
File systems
ISO 9660
Joliet
Rock Ridge
Amiga Rock Ridge extensions
El Torito
Apple ISO9660 Extensions
Universal Disk Format
Mount Rainier

A Compact Disc or CD is an optical disc used to store digital data, originally developed for storing digital audio. The CD, available on the market in late 1982, remains the standard physical medium for commercial audio recordings as of 2007. An audio CD consists of one or more stereo tracks stored using 16-bit PCM coding at a sampling rate of 44.1 kHz. Standard CDs have a diameter of 120 mm and can hold approximately 80 minutes of audio. There are also 80 mm discs, sometimes used for CD singles, which hold approximately 20 minutes of audio. Compact Disc technology was later adapted for use as a data storage device, known as a CD-ROM, and to include record-once and re-writable media (CD-R and CD-RW respectively). CD-ROMs and CD-Rs remain widely used technologies in the Computer industry as of 2007. The CD and its extensions have been extremely successful: in 2004, the annual worldwide sales of CD-Audio, CD-ROM, and CD-R reached about 30 billion discs.

Contents [hide]
1 History
2 Physical details
2.1 Disc shapes and diameters
2.2 "Shape CD"
3 Audio format
3.1 Storage capacity and playing time
3.2 Main physical parameters
3.3 Data structure
4 CD-ROM
5 Manufacture
6 Recordable CD
7 Copy protection
8 References
9 See also
10 External links

[edit] History
In 1979 Philips and Sony set up a joint task force of engineers to design the new digital audio disc. Prominent members of the task force were Kees Immink and Toshitada Doi. After a year of experimentation and discussion, the taskforce produced the "Red Book", the Compact Disc standard. Philips contributed the general manufacturing process, based on video LaserDisc technology. Philips also contributed the Eight-to-Fourteen Modulation (EFM), which offers both a long playing time and a high resilience against disc handling damage such as scratches and fingerprints, while Sony contributed the error-correction method, CIRC. The Compact Disc Story[1], told by a former member of the taskforce, gives background information on the many technical decisions made, including the choice of the sampling frequency, playing time, and disc diameter. According to Philips, the Compact Disc was thus "invented collectively by a large group of people working as a team[2]."

The Compact Disc reached the market in late 1982 in Asia, and early the following year in the United States and other markets. The first CDs available were 16 Japanese-made titles from CBS/Sony. This event is often seen as the "Big Bang" of the digital audio revolution. The new audio disc was enthusiastically received, especially in the early-adopting classical music and audiophile communities and its handling quality received particular praise. As the price of players sank rapidly, the CD began to gain popularity in the larger popular and rock music markets.

The CD was originally thought of as an evolution of the gramophone record, rather than primarily as a data storage medium. Only later did the concept of an 'audio file' arise, and the generalising of this to any data file. From its origins as a music format, Compact Disc has grown to encompass other applications. In June 1985, the CD-ROM (read-only memory) and, in 1990, CD-Recordable were introduced, also developed by Sony and Philips.

While CDs are significantly more durable than earlier audio formats, they are susceptible to damage from daily usage and environmental factors. Libraries and archives should enact optical media preservation procedures to ensure continued usability.

[edit] Physical details

The optical lens of a CD drive.A Compact Disc is made from a 1.2 mm thick disc of almost pure polycarbonate plastic and weighs approximately 16 grams. A thin layer of Aluminium (or rarely gold, used for its data longevity, such as in some limited-edition audiophile CDs) is applied to the surface to make it reflective, and is protected by a film of lacquer. The lacquer is normally printed directly and not with an adhesive label. Common printing methods for compact discs are screen-printing and offset printing.

CD data is stored as a series of tiny indentations (pits), encoded in a tightly packed spiral track moulded into the top of the polycarbonate layer. The areas between pits are known as 'lands'. Each pit is approximately 100 nm deep by 500 nm wide, and varies from 850 nm to 3.5 μm in length.

The spacing between the tracks, the pitch, is 1.6 μm. A CD is read by focusing a 780 nm wavelength semiconductor laser through the bottom of the polycarbonate layer. The change in height between pits and lands leads to a difference in intensity in the light reflected. By measuring the intensity change with a photodiode, it is possible to read the data from the disc.

The pits and lands themselves do not directly represent the zeros and ones of binary data. Instead, Non-return-to-zero, inverted encoding is used: a change from pit to land or land to pit indicates a one, while no change indicates a zero. This in turn is decoded by reversing the Eight-to-Fourteen Modulation used in mastering the disc, and then reversing the Cross-Interleaved Reed-Solomon Coding, finally revealing the raw data stored on the disc.

Pits are much closer to the label side of a disc so that defects and dirt on the clear side can be out of focus during playback. Discs consequently suffer more damage because of defects such as scratches on the label side, whereas clear-side scratches can be repaired by refilling them with plastic of similar index of refraction, or by polishing.

A Mini-CD is 8 centimeters in diameter
[edit] Disc shapes and diameters
The digital data on a CD begins at the center of the disc and proceeds outwards to the edge, which allows adaptation to the different size formats available. Standard CDs are available in two sizes. By far the most common is 120 mm in diameter, with a 74 or 80-minute audio capacity and a 650 or 700 MB data capacity. 80 mm discs ("Mini CDs") were originally designed for CD singles and can hold up to 21 minutes of music or 184 MB of data but never really became popular. Today nearly all singles are released on 120mm CDs, which is called a Maxi single.

[edit] "Shape CD"
Novelty shaped CDs are also available in a number of shapes and sizes, and are mostly used for marketing. The most common variant is a "business card" CD, a CD-single with portions removed at the top and bottom to more closely resemble the form-factor of a business card.

Physical size Audio Capacity CD-ROM Data Capacity
12 cm (standard) 74-80 min 650-703 MB
8 cm (mini-CD) 21-24 min 185-210 MB
"Business card" ~6 min ~55 MB

[edit] Audio format
The technical format of an audio compact disc (Compact Disc Digital Audio—CDDA) is laid down in a document produced in 1980 by the format's joint creators, Sony and Philips. The document is known colloquially as the "Red Book" after the colour of its cover. The format is a two-channel 16-bit PCM encoding at a 44.1 kHz sampling rate. Four-channel sound is an allowed option within the Red Book format, but has never been implemented.

The sampling rate of 44.1 kHz is inherited from a method of converting digital audio into an analog video signal for storage on video tape, which was the most affordable way to get the data from the recording studio to the CD manufacturer at the time the CD specification was being developed. A device that turns an analog audio signal into PCM audio, which in turn is changed into an analog video signal is called a PCM adaptor. This technology could store six samples (three samples per each stereo channel) in a single horizontal line. A standard NTSC video signal has 245 usable lines per field, and 59.94 fields/s, which works out at 44,056 samples/s/stereo channel. Similarly, PAL has 294 lines and 50 fields, which gives 44,100 samples/s/stereo channel. This system could either store 14-bit samples with some error correction, or 16-bit samples with almost no error correction.

There was a long debate over whether to use 14- or 16-bit samples, and 44,056 or 44,100 samples/s, when the Sony/Philips task force designed the Compact Disc; Philips had already developed a 14 bit D/A converter, but Sony insisted on 16 bit. In the end, 16 bits and 44.1 kilosamples per second prevailed. Philips found a way to produce 16-bit quality using their 14-bit DAC by using four times oversampling.

[edit] Storage capacity and playing time
The original target storage capacity for a CD was an hour of audio content, and a disc diameter of 115 mm was sufficient for this. According to Philips, Sony vice-president Norio Ohga suggested extending the capacity to 74 minutes to accommodate a complete performance of Beethoven’s 9th Symphony;[3] however, Kees Immink of Philips denies this.[1] The extra playing time subsequently required the change to a 120 mm disc.

According to a Sunday Tribune interview [1] the story is slightly more involved. At that time (1979) Philips owned Polygram, one of the world’s largest distributors of music. Polygram had set up a large experimental CD plant in Hanover, Germany, which could produce huge amounts of CDs having, of course, a diameter of 115 mm. Sony did not yet have such a facility. If Sony had agreed on the 115 mm disc, Philips would have had a significant competitive edge in the market. Sony was aware of that, did not like it, and something had to be done. The long-playing time of Beethoven's Ninth imposed by Ohga was used to push Philips to accept 120 mm, so that Philips’ Polygram lost its edge on disc fabrication.

The 74-minute playing time of a CD, being more than that of most long-playing vinyl albums, was often used to the CD’s advantage during the early years when CDs and LPs vied for commercial sales. CDs would often be released with one or more bonus tracks, enticing consumers to buy the CD for the extra material. However, attempts to combine double LPs onto one CD occasionally resulted in an opposing situation in which the CD would actually offer fewer tracks than the LP equivalent. An example is the 1987 album Kiss Me, Kiss Me, Kiss Me by The Cure, which states in the CD liner notes: "The track Hey You!!! which appears on the double album and cassette has been omitted so as to facilitate a single compact disc." The 2006 re-release of this album saw the re-inclusion of the missing track. Another example is the original late-1980s Warner Bros. Records reissue of Fleetwood Mac's Tusk album, which substituted the long album version of "Sara" with the shorter single version. Enough complaints were lodged to eventually convince Warner Bros. to remaster the album in the mid-1990s with the original contents intact.

[edit] Main physical parameters
The main parameters of the CD (taken from the September 1983 issue of the compact disc specification) are as follows:

Scanning velocity: 1.2–1.4 m/s (constant linear velocity) - equivalent to approximately 500 rpm at the inside of the disc, and approximately 200 rpm at the outside edge. (A disc played from beginning to end slows down during playback.)
Track pitch: 1.6 μm.
Disc diameter 120 mm.
Disc thickness: 1.2 mm.
Inner radius program area: 25 mm.
Outer radius program area: 58 mm.
Center spindle hole diameter: 15 mm
The program area is 86.05 cm² and the length of the recordable spiral is 86.05 cm² / 1.6 μm = 5.38 km. With a scanning speed of 1.2 m/s, the playing time is 74 minutes, or around 650 MB of data on a CD-ROM. If the disc diameter were only 115 mm, the maximum playing time would have been 68 minutes, i.e. six minutes less. A disc with data packed slightly more densely is tolerated by most players (though some old ones fail). Using a linear velocity of 1.2 m/s and a track pitch of 1.5 μm leads to a playing time of 80 minutes, or a capacity of 700 MB. Even higher capacities on non-standard discs (up to 99 minutes) are available at least as recordables, but generally the tighter the tracks are squeezed the worse the compatibility.

[edit] Data structure
The smallest entity in the CD audio format is called a frame. A frame can accommodate six complete 16-bit stereo samples, i.e. 2×2×6 = 24 bytes. A frame comprises 33 bytes, of which 24 are audio bytes (six full stereo samples), eight CIRC-generated error correction bytes and one subcode byte. The eight bits of a subcode byte are available for control and display. Under Eight-to-Fourteen Modulation (EFM) rules, each data/audio byte is translated into 14-bit EFM words, which alternate with 3-bit merging words. In total we have 33*(14+3) = 561 bits. A 27-bit unique synchronization word is added, so that the number of bits in a frame totals 588. The synchronization word cannot occur in the normal bit stream, and can thus be used to identify the beginning of a frame. Data on a CD-ROM are organized in both frames and sectors, where a CD-ROM sector contains 98 frames, and holds 98×24 = 2352 (user) bytes, of which 304 bytes are normally used for sector IDs and an additional layer of error correction, leaving 2048 bytes for payload data.

The largest entity on a CD is called a track. A CD can contain 99 tracks.

Current manufacturing processes allow an audio CD to contain up to 80 minutes (variable from one replication plant to another) without requiring the content creator to sign a waiver. Thus, in current practice, maximum CD playing time has crept higher while maintaining acceptable standards of reliability.

[edit] CD-ROM
Main article: CD-ROM
For its first few years of existence, the compact disc was purely an audio format. However, in 1985 the Yellow Book CD-ROM standard was established by Sony and Philips, which defined a non-volatile optical data computer data storage medium using the same physical format as audio compact discs, readable by a computer with a CD-ROM drive.

[edit] Manufacture
Main article: CD manufacturing
Replicated CDs are mass-produced initially using a hydraulic press. Small granules of raw polycarbonate plastic are fed into the barrel while under heat. The screw inside the barrel forces the liquified plastic into the mold cavity. Equipped with a metal stamper the mold closes, allowing the plastic to cool and harden. Once opened, the disc substrate is removed from the mold by a robotic arm, and a 15 mm diameter center hole (called a stacking ring) is removed. The cycle time, the time it takes to "stamp" one CD, is usually 2-3 seconds.

This method produces the clear plastic blank part of the disc. After the metallic layer is applied to the clear blank substrate the disc goes under a UV light for drying and it is ready to go to press. To press the CD first a glass master is cut using a high-power laser on a device similar to a CD writer. This glass master is a positive master. After testing it is used to make a die by pressing it against a metal disc.

The die then becomes a negative image: a number of them can be made depending on the number of pressing mills that are to be running off copies of the final CD. The die then goes into the press and the image is pressed onto the blank CD leaving a final positive image on the disc. A small circle of varnish is then applied as a ring around the centre of the disc and a fast spin spreads it evenly over the surface. The disc can then be printed and packed.

Manufactured CDs that are sold in stores are wrapped/sealed via a process called "polywrapping" or sometimes are shrink wrapped.

[edit] Recordable CD
Main article: CD-R

A typical 700-megabyte CD-RRecordable compact discs, CD-Rs, are injection moulded with a "blank" data spiral. A photosensitive dye is then applied, after which the discs are metalized and lacquer coated. The write laser of the CD recorder changes the color of the dye to allow the read laser of a standard CD player to see the data as it would an injection moulded compact disc. The resulting discs can be read by most CD-ROM drives and played in most audio CD players. CD-R recordings are designed to be permanent. Over time the dye's physical characteristics may change, however, causing read errors and data loss until the reading device cannot recover with error correction methods. The design life is from 20 to 100 years depending on the quality of the discs, the quality of the writing drive, and storage conditions. However, testing has demonstrated such degradation in as little as 18 months under ideal storage conditions[4].

Four Memorex CD-Rs.CD-RW is a re-recordable medium that uses a metallic alloy instead of a dye. The write laser in this case is used to heat and alter the properties (amorphous vs. crystalline) of the alloy, and hence change its reflectivity. A CD-RW does not have as great a difference in reflectivity as a pressed CD or a CD-R, and so many CD audio players cannot read CD-RW discs, although most stand-alone DVD players can.

CD-Rs follow the Orange Book standard.

[edit] Copy protection
Main article: CD/DVD copy protection
The Red Book audio specification, except for a simple 'anti-copy' bit in the subcode, does not include any serious copy protection mechanism. Starting in early 2002, attempts were made by record companies to market "copy-protected" non-standard compact discs, which cannot be ripped (copied) to hard drives or easily converted to MP3s. One major drawback to these copy-protected discs is that most will not play on computer CD-ROM drives, as well as some standalone CD players that use CD-ROM mechanisms. Philips has stated that such discs are not permitted to bear the trademarked Compact Disc Digital Audio logo because they violate the Red Book specification. Moreover, there has been great public outcry over copy-protected discs because many see it as a threat to fair use. Numerous copy-protection systems have been countered by readily-available, often free, software. Also, any CD that can play on a standard audio CD player can be extracted via the standard S/PDIF digital output, rendering any copy protection ineffective.

Audio System Measurements

2007-06-30T04:23:00.000-07:00

Audio system measurements are made for several purposes. Designers take measurements so that they can specify the performance of a piece of equipment. Maintenance engineers make them to ensure equipment is still working to specification, or to ensure that the cumulative defects of an audio path are within limits considered acceptable. Some aspects of measurement and specification relate only to intended usage. For example, magnetic tape speeds and types, interface specifications, or power output.

Others are intended as an index of the quality, or 'fidelity', of reproduction perceivable by a human. It is important that such measurements accommodate psychoacoustic principles, so that they truly measure the system in a way that is 'subjectively valid'. Humans don't hear very low levels of sound, so there is reason to be concerned about the precise nature of noise at very low levels, than at higher levels.

Contents [hide]
1 Subjectivity and frequency weighting
2 Measurable performance
2.1 Analog electrical
2.2 Mechanical
2.3 Digital
3 Unquantifiable?
4 See also
5 References
6 External links

[edit] Subjectivity and frequency weighting
Measurements based on psychoachoustics, such as the measurement of noise, often use a weighting filter. It is well-established that human hearing is more sensitive to some frequencies than others, as demonstrated by equal-loudness contours, but it is not well appreciated that these contours vary depending on the type of sound. The measured curves for pure tones, for instance, are different from those for random noise. The ear also responds less to short bursts, below 100 to 200 ms, than to continuous sounds [1] such that a quasi-peak detector has been found to give the most representative results when noise contains click or bursts, as is often the case for noise in digital systems. [2] For these reasons a set of subjectively valid measurement techniques have been devised and incorporated into BS, IEC, EBU and ITU standards. These newer methods of audio quality measurement are used by broadcast engineers throughout most of the world, as well as by some audio professionals, though the older A-weighting standard for continuous tones is still commonly used by others. [1] Subjectively valid methods came to prominence in consumer audio in the UK and Europe in the 1970s, when the introduction of compact cassette tape and DBX and Dolby noise reduction techniques revealed the unsatisfactory nature of many basic engineering measurements. The specification of weighted CCIR-468 quasi-peak noise, and weighted quasi-peak wow and flutter became particularly widely used and attempts were made to find more valid methods for distortion measurement.

No single measurement can assess audio quality. Instead, it is usual to take a series of measurements to test for the various types of degradation that can reduce fidelity. Thus, when testing an analogue tape machine it is necessary to test for wow and flutter and tape speed variations over longer periods, as well as for distortion and noise. When testing a digital system, testing for speed variations is normally considered unnecessary given the nearly ubiquitous accurate clocks in digital circuitry, but testing for aliasing and timing jitter is often desirable, as these have caused audible degradation in many systems. The claim is often made that different methods of measuring noise, or distortion, are better suited to different items of equipment is not widely believed among professional audio engineers.

Once subjectively valid methods have been shown to correlate well with listening tests over a wide range of conditions, then such methods are generally adopted as preferred. But it's important to realise that engineering methods are not always sufficient to when comparing like with like. One CD player, for example, might have higher measured noise than another CD player when measured RMS, or even A-weighted RMS, yet sound quieter and measure lower when 468-weighting is used. This could be because it has more noise at high frequencies, or even at frequencies beyond 20 kHz, both of which are less important since human ears are less sensitive to them. See noise shaping.) This effect is how Dolby B works and why it was introduced. Cassette noise, which was predominately high frequency and unavoidable given the small size and speed of the recorded track could be made subjectively much less important. The noise sounded 10 dB quieter, but failed to measure much better unless 468-weighting was used rather than A-weighting.

[edit] Measurable performance

[edit] Analog electrical
Frequency response
The signal should be passed at least over the audible range (usually quoted as 20 Hz to 20 kHz) with no significant peaks or dips. The human ear can discern differences in level of about 3 dB in some frequency ranges, so peaks and troughs must be less than this. Much modern equipment is capable of less than ±1 dB variation over the entire audible frequency range. Rapid variations over a small frequency range (ripple), or very steep rolloffs are considered undesirable as they can correspond to resonances associated with energy storage which produce delayed echoes and hence colouration, or decreased quality, of the sound.
Total harmonic distortion (THD)
In music material, there are distinct tones, and some kinds of distortion involve spurious double or triple the frequencies of those tones. Such harmonically related distortion is called harmonic distortion. For high fidelity, this is usually expected to be < 1% for electronic devices; mechanical elements such as loudspeakers usually have inescapable higher levels. Low distortion is relatively easy to achieve in electronics with use of negative feedback, but the use of high levels of feedback in this manner has been the topic of much controversy among audiophiles — see electronic amplifier. Essentially all loudspeakers produce more distortion than electronics, and 1–5% distortion is not unheard of at moderately loud listening levels. Human ears are less sensitive to distortion in the bass frequencies, and levels are usually expected to be under 10% at loud playback. Distortion which creates only even-order harmonics for a sine wave input is sometimes considered less bothersome than odd-order distortion.
Output power
Output power for amplifiers is ideally measured and quoted as maximum sinewave (ie, RMS) power output per channel, at a specified distortion level at a particular load, which by convention and government regulation, is considered the most meaningful measure of power available on music signals, though real, non-clipping music has a high peak-to-average ratio, and usually averages well below the maximum possible. The commonly given measurement of PMPO (peak music power out) is largely meaningless and often used in marketing literature; in the late 1960s there was much controversy over this point and the US Government (FTA) required that RMS figures be quoted for all high fidelity equipment. Music power has been making a comeback in recent years. See also Audio power.
Power specifications require the load impedance to be specified, and in some cases two figures will be given (for instance, a power amplifier for loudspeakers will be typically measured at 4 and 8 ohms). Any amplifier will drive more current to a lower impedance load. For example, it will deliver more power into a 4-ohm load, as compared to 8-ohm, but it must not be assumed that it is capable of sustaining the extra current unless it is specified so. Power supply limitations may limit high current performance.
Intermodulation distortion (IMD)
Distortion which is not harmonically related to the signal being amplified is intermodulation distortion. It is a measure of the level of spurious signals resulting from unwanted combination of different frequency input signals. This effect results form non-linearities in the system. Again, sufficiently high levels of negative feedback can reduce this effect, as for instance in an amplifier. Many believe it is better to design electronics to minimize feedback levels. Low intermodulation equipment is difficult to design while meeting other high accuracy requirements. Intermodulation in speaker drivers is, as with harmonic distortion, almost always larger than in most electronics. Reducing cone excursion is one way to reduce intermodulation distortion as is designing and building crossovers so that out of band signals are reduced quickly. This raises other problems related to crossover designs and is an example of the tradeoffs which must be made in high quality audio design.
Noise
The level of unwanted noise generated by the system itself, or by interference from external sources added to the signal. Hum usually refers to noise only at power line frequencies (as opposed to broadband white noise), which is introduced through induction of power line signals into the inputs of gain stages. Or from inadequately regulated power supplies.
Crosstalk
The introduction of noise (from another signal channel) caused by stray inductance or capacitance between components or lines. Crosstalk reduces, sometimes noticeablly, separation between channels (eg, in a stereo system). It is given in dB relative to a nominal level of signal in the path receiving interference. Crosstalk is normally only a problem in equipment in which several channels are handled in the same chassis.
Common-mode rejection ratio (CMRR)
All electronic equipment with inputs is susceptible to this problem. In balanced audio systems, there are equal and opposite signals (difference-mode) in inputs, and any interference imposed on both leads will be subtracted, canceling out that interference (ie, the common-mode). CMRR is a measure of a system's ability to ignore any such interference and especially hum which arises at its input. It is generally only significant with long lines on an input, or when some kinds of ground loop problems exist. Unbalanced inputs do not have common mode resistance; induced noise on their inputs appears directly as noise or hum.
Dynamic range and Signal-to-noise ratio (SNR)
The difference between the maximum level a component can accommodate and the noise level it produces. Input noise is not counted in this measurement. It is measured in dB.
Dynamic range refers to the ratio of maximum to mimimum loudness in a given signal source (eg, music or programme material), and this measurement also quantifies the maximum dynamic range an audio system can carry. This is the ratio (usually expressed in dB) between the noise floor of the device with no signal and the maximum signal (usually a sine wave) that can be output at a specified (low) distortion level.
Since the early 1990s it has been recommened by several authorities including the Audio Engineering Society that measurements of dynamic range be made with an audio signal present. This avoids questionable measurements based on the use of blank media, or muting circuits.
Signal-to-noise ratio (SNR), however, is the ratio between the noise floor and an arbitrary reference level or alignment level. In "professional" recording equipment, this reference level is usually +4 dBu (IEC 60268-17), though sometimes 0 dBu (UK and Europe - EBU standard Alignment level). 'Test level', 'measurement level' and 'line-up level' mean different things, often leading to confusion. In "consumer" equipment, no standard exists, though −10 dBV and −6 dBu are common.
Different media characteristically exhibit different amounts of noise and headroom. Though the values vary widely between units, a typical analogue cassette might give 60 dB, a CD almost 100 dB. Most modern quality amplifiers have >110 dB dynamic range, which approaches that of the human ear, usually taken as around 160 dB. See Programme levels.
Phase distortion, Group delay, and Phase delay
A perfect audio component will maintain the phase coherency of a signal over the full range of frequencies. Phase distortion can be extremely difficult to reduce or eliminate. The human ear is largely insensitive to phase distortion, though it is exquisitely sensitive to relative phase relationships within heard sounds. For many this figure lacks importance; however, there are many who argue its significance. Multi-driver loudspeaker systems have complex phase distortions, caused by crossovers, by driver placment relative to other drivers, and by internal driver characteristics.
Transient distortion
A system may have low distortion for a steady-state signal, but not on sudden transients. This problem can be traced to amplifier power supplies in some instances, to insufficient high frequency performance in amplifiers, to negative feedback in amplifiers, or in loudspeakers to the mass and resonances of drivers and enclosures. Related measurements are slew rate and rise time. Transient distortion can be hard to measure. Many otherwise good power amplifier designs have foudn to have inadequate slew rates, by modern standards. Most loudspeakers generate significant amounts of transient distortion, though some designs are less prone to this (e.g. electrostatic loudspeakers, plasma arc tweeters, ribbon tweeters).
Damping factor
A higher number is generally thought better. This is a measure of how well a power amplifier can control the undesired motion of a loudspeaker driver due largely to mechanical reactance. The amplifier must be able to damp out resonances caused by the mechanical motions (eg, inertia) of the moving parts of the speaker. For the common voice coil drivers, this essentially involves ensuring that the output impedance of the amplifier is close to zero. Damping factor is actually a relative way of specifying the output impedance of an amplifier with a particular load. It is affected by the cables used to connect the speakers to the amplifier, and by the amount of negative feedback especially in solid state amplifiers.

[edit] Mechanical
Wow and flutter
These measurements are related to physical motion in a component, largely the drive mechanism of analogue media, such as vinyl records and magnetic tape. "Wow" is slow speed (a few Hz) variation, caused by longer term drift of the drive motor speed, whereas "flutter" is faster speed (a few tens of Hz) variations, usually caused by mechanical defects such as out-of-roundness of the capstan of a tape transport mechanism. The measurement is given in % and a lower number is better.
Rumble
The measure of the low frequency (many tens of Hz) noise contributed by the turntable of an analogue playback system. It is caused by imperfect bearings, by uneven motor windings, by vibrations in driving bands in some turntables, by room vibrations (eg, from traffic) which is transmitted by the turntable mounting and so to the phono cartridge. A lower number is better.

[edit] Digital
Note that digital systems do not suffer from many of these effects at a signal level, though the same processes occur in the circuitry, since the data being handled is symbolic. As long as the symbol survives the transfer between components, and can be perfectly regenerated (eg, by pulse shaping techniques) the data itself is perfectly maintained. The data is typically buffered in a memory, and is clocked out by a very precise crystal oscillator. The data usually does not degenerate as it passes through many stages, because each stage regenerates new symbols for transmission.

But digital systems have their own problems. Digitizing adds noise which is measurable, and which depends on the resolution ('number of bits") of the system, regardless of other quality issues. Clock timing errors (jitter) result in non-linear distortion of the signal. The quality measurement for a digital system centers on the probability of an error in transmission or reception. Otherwise the quality of the system is defined more by design intent (ie, specifications) than measurements, such as the sample rate and bit depth. In general, digital systems are much less prone to error than analog systems. However, nearly all digital systems contain analog inputs and/or outputs, and certainly all of those which interact with the analog world do so. These analog components of the digital system can suffer analog effects and potentially compromise the integrity of a well designed digital system.

Jitter
A measurement of the variation in period between clock cycles, which should theoretically be exactly the same period. Less jitter is better.
Sample rate
A specification of the rate at which measurements are taken of the analog signal. This is measured in samples per second, or hertz. A higher sampling rate allows a greater total bandwidth or flatband frequency response. It can also reduce the effects of jitter.
Bit depth
A specification of the accuracy of each measurement. For example, a 3-bit system would be able to measure 23 = 8 different levels, so it would round the actual level at each point to the nearest representable. Typical values for audio are 8-bit, 16-bit, 24-bit, and 32-bit. The bit depth determines the theoretical maximum signal-to-noise ratio or dynamic range for the system. It is common for devices to create more noise than the minimum possible noise floor, however. Sometimes this is done intentionally; dither noise is added to decrease the negative effects of quantization noise by converting it into a higher level of uncorrelated noise.
To calculate the maximum theoretical dynamic range of a digital system, find the total number of levels in the system. Dynamic Range = 20·log(# of different levels). Note: the log function has a base of 10. Example: An 8-bit system has 256 different possibilities, from 0 – 255. The smallest signal is 1 and the largest is 255. Dynamic Range = 20·log(255) = 48 dB.
Sample accuracy/synchronization
Not as much a specification as an ability. Since independent digital audio devices are each run by their own crystal oscillator, and no two crystals are exactly the same, the sample rate will be slightly different. This will cause the devices to drift apart over time. The effects of this can vary. If one digital device is used to monitor another digital device, this will cause dropouts in the audio, as one device will be producing more or less data than the other per unit time. If two independent devices record at the same time, one will lag the other more and more over time. This effect can be circumvented with a wordclock synchronization.
Linearity
Differential non-linearity and integral non-linearity are two measurements of the accuracy of an analog-to-digital converter. Basically, they measure how close the threshold levels for each bit are to the theoretical equally-spaced levels.

[edit] Unquantifiable?
Some audiophiles have postulated that the present set of audio measurements, as exemplified by the above list, does not fully represent all that is significant in accurate music reproduction, and instead represents only those aspects which are relatively easy and cost-effective to measure with our current technology. Given the complexity and sophistication of human hearing and perception, it is felt that some consideration should be given to the possibility that there may be aspects of music reproduction that have yet to be identified.

All of the above measurements are quantitative, not qualitative. Subjectivists claim that listening tests are more appropriate for appraising the quality of an audio system than measuring the accuracy with which it can reproduce a waveform.

Audio Editing

2007-06-30T04:21:00.000-07:00

Audio editing was a new technology that developed in the middle part of the 20th century with the advent of magnetic tape recording. Originally, editing was done on reel-to-reel tape machines and edits were made with straight razors and special tape to connect pieces of tape that had been cut. Audio editors would listen to recorded tapes at low volumes, and then located specific sounds using a process called scrubbing, which is the slow rocking back and forth of the tape reels across the playback heads of the tape deck.

With the development of microcomputer technology, Sound Recordists were able to digitize their recordings and edit them as files on a computer's hard disk. The computer programs responsible for this task are known as digital audio editors. The earliest program to become widely used in this application was a wave editor called Sound Designer in the late 1980s and early 1990s. Sound Designer was created by a company called Digidesign who achieved early industry dominance. In recent years, however, that dominance has been challenged by a number of companies attempting to grab a portion of Digidesign's market share.

In recent years, with the growing popularity of GNU/Linux, a number of Open Source software projects have sprung up in order to develop an open source audio editing program. This movement has been bolstered recently by the development of ALSA, and the Linux low latency kernel patch, which allow the GNU/Linux Operating System to achieve audio processing performance equal to that of commercial operating systems. The multi-platform package Audacity is currently the most fully-featured free software audio editor.

Amplifier

2007-06-30T04:19:00.001-07:00

Contents [hide]
1 General characteristics of amplifiers
1.1 Gain
1.2 Output dynamic range
1.3 Bandwidth and rise time
1.4 Settling time and aberrations
1.5 Slew rate
1.6 Noise
1.7 Efficiency
1.8 Linearity
2 Electronic amplifiers
2.1 Amplifier classes
2.2 Vacuum tube (valve) amplifiers
2.3 Transistor amplifiers
2.4 Operational amplifiers (op-amps)
2.5 Fully differential amplifiers (FDA)
2.6 Video amplifiers
2.6.1 Oscilloscope vertical amplifiers
2.6.2 Distributed amplifiers
2.7 Microwave amplifiers
2.7.1 Travelling wave tube (TWT) amplifiers
2.7.2 Klystrons
3 Musical instrument (audio) amplifiers
4 Other amplifier types
4.1 Carbon microphone
4.2 Magnetic amplifier
4.3 Optical amplifiers
4.4 Miscellaneous types
5 References
6 See also

[edit] General characteristics of amplifiers
Most amplifiers can be characterized by a number of parameters.

[edit] Gain
The gain is the ratio of output power to input power, and is usually measured in decibels (dB). (When measured in decibels it is logarithmically related to the power ratio: G(dB)=10log(Pout/Pin)).

[edit] Output dynamic range
Output dynamic range is the range, usually given in dB, between the smallest and largest useful output levels. Since the lowest useful level is limited by output noise, this is quoted as the amplifier dynamic range.

[edit] Bandwidth and rise time
The bandwidth (BW) of an amplifier is usually defined as the difference between the lower and upper half power points. This is therefore also known as the −3 dB BW. Bandwidths for other response tolerances are sometimes quoted (−1 dB, −6 dB etc.).

As an example, a good audio amplifier will be essentially flat between twenty hertz to about twenty kilohertz (the range of normal human hearing), so the amplifier's usable frequency response needs to extend considerably beyond this (one or more octaves either side) and typically a good audio amplifier will have -3 dB points < 10 and > 65 kHz.

The rise time of an amplifier is the time taken for the output to change from 10% to 90% of its final level when driven by a step input.

Many amplifiers are ultimately slew rate limited (typically by the impedance of a drive current having to overcome capacitive effects at some point in the circuit), which may limit the full power bandwidth to frequencies well below the amplifiers frequency response when dealing with small signals.

For a Gaussian response system (or a simple RC roll off), the rise time is approximated by:

Tr * BW = 0.35, where Tr is in seconds and BW is in Hz.

[edit] Settling time and aberrations
Time taken for output to settle to within a certain percentage of the final value (say 0.1%). This is usually specified for oscilloscope vertical amplifiers and high accuracy measurement systems.

[edit] Slew rate
Slew rate is the maximum rate of change of output variable, usually quoted in volts per second (or microsecond).

[edit] Noise
This is a measure of how much noise is introduced in the amplification process. Noise is an undesirable but inevitable product of the electronic devices and components. It is measured in either decibels or the peak output voltage produced by the amplifier when no signal is applied.

[edit] Efficiency
Efficiency is a measure of how much of the input power is usefully applied to the amplifier's output. Class A amplifiers are very inefficient, in the range of 10–20% with a max efficiency of 25%. Modern Class AB amps are commonly between 35–55% efficient with a theoretical maximum of 78.5%. Commercially available Class D amplifiers have reported efficiencies as high as 97%. The efficiency of the amplifier limits the amount of total power output that is usefully available. Note that more efficient amplifiers run much cooler, and often do not need any fans even in multi-kilowatt designs.

[edit] Linearity
An ideal amplifier would be a totally linear device, but real amplifiers are only linear within certain practical limits. When the signal drive to the amplifier is increased, the output also increases until a point is reached where some part of the amplifier becomes saturated and cannot produce any more output; this is called clipping, and results in distortion.

Some amplifiers are designed to handle this in a controlled way which causes a reduction in gain to take place instead of excessive distortion; the result is a compression effect, which (if the amplifier is an audio amplifier) will sound much less unpleasant to the ear. For these amplifiers, the 1dB compression point is defined as the input power (or output power) where the gain is 1dB less than the small signal gain.

Linearization is an emergent field, and there are many techniques, such us feedforward, predistortion, postdistortion, EER, LINC, CALLUM, cartesian feedback, etc., in order to avoid the undesired effects of the non-linearities.

[edit] Electronic amplifiers
Main article: electronic amplifier
There are many types of electronic amplifiers for different applications.

One common type of amplifier is the electronic amplifier, commonly used in radio and television transmitters and receivers, high-fidelity ("hi-fi") stereo equipment, microcomputers and other electronic digital equipment, and guitar and other instrument amplifiers. Its critical components are active devices, such as vacuum tubes or transistors.

[edit] Amplifier classes
Amplifiers are commonly classified by the conduction angle (sometimes known as 'angle of flow') of the input signal through the amplifying device; see electronic amplifier.

Class A
Where efficiency is not a consideration, most small signal linear amplifiers are designed as Class A, which means that the output devices are always in the conduction region. Class A amplifiers are typically more linear and less complex than other types, but are very inefficient. This type of amplifier is most commonly used in small-signal stages or for low-power applications (such as driving headphones).
Class B
In Class B, there are two output devices (or sets of output devices), each of which conducts alternately for exactly 180 deg (or half cycle) of the input signal.
Class AB
Class AB amplifiers are a compromise between Class A and B, which improves small signal output linearity; conduction angles vary from 180 degrees upwards, selected by the amplifier designer. Usually found in low frequency amplifiers (such as audio and hi-fi) owing to their relatively high efficiency, or other designs where both linearity and efficiency are important (cell phones, cell towers, TV transmitters).
Class C
Popular for high power RF amplifiers, Class C is defined by conduction for less than 180° of the input signal. Linearity is not good, but this is of no significance for single frequency power amplifiers. The signal is restored to near sinusoidal shape by a tuned circuit, and efficiency is much higher than A, AB, or B classes of amplification.
Class D
Class D amplifiers use switching to achieve a very high power efficiency (more than 90% in modern designs). By allowing each output device to be either fully on or off, losses are minimized. A simple approach such as pulse-width modulation is sometimes still used; however, high-performance switching amplifiers use digital techniques, such as sigma-delta modulation, to achieve superior performance. Formerly used only for subwoofers due to their limited bandwidth and relatively high distortion, the evolution of semiconductor devices has made possible the development of high fidelity, full audio range Class D amplifiers, with S/N and distortion levels similar to their linear counterparts.
Other classes
There are several other amplifier classes, although they are mainly variations of the previous classes. For example, Class H and Class G amplifiers are marked by variation of the supply rails (in discrete steps or in a continuous fashion, respectively) following the input signal. Wasted heat on the output devices can be reduced as excess voltage is kept to a minimum. The amplifier that is fed with these rails itself can be of any class. These kinds of amplifiers are more complex, and are mainly used for specialized applications, such as very high-power units. Also, Class E and Class F amplifiers are commonly described in literature for radio frequencies applications where efficiency of the traditional classes deviate substantially from their ideal values. These classes use harmonic tuning of their output networks to achieve higher efficiency and can be considered a subset of Class C due to their conduction angle characteristics.
Power Amplifier
The term "power amplifier" is a relative term with respect to the amount of power delivered to the load and/or sourced by the supply circuit. In general a power amplifier is designated as the last amplifier in a transmission chain and is the amplifier stage that typically requires most attention to power efficiency. For these reasons, a power amplifier is typically any of the above-mentioned classes except Class A.

[edit] Vacuum tube (valve) amplifiers
Main article: valve amplifier
According to Symons, while semiconductor amplifiers have largely displaced valve amplifiers for low power applications, valve amplifiers are much more cost effective in high power applications such as "radar, countermeasures equipment, or communications equipment" (p. 56). Many microwave amplifiers are specially designed valves, such as the klystron, gyrotron, traveling wave tube, and crossed-field amplifier, and these microwave valves provide much greater single-device power output at microwave frequencies than solid-state devices (p. 59).[1]

[edit] Transistor amplifiers
Main articles: transistor, bipolar junction transistor, and MOSFET
The essential role of this active element is to magnify an input signal to yield a significantly larger output signal. The amount of magnification (the "forward gain") is determined by the external circuit design as well as the active device.

Many common active devices in transistor amplifiers are bipolar junction transistors (BJTs) and metal oxide semiconductor field-effect transistors (MOSFETs).

Applications are numerous, some common examples are audio amplifiers in a home stereo or PA system, RF high power generation for semiconductor equipment, to RF and Microwave applications such as radio transmitters.

[edit] Operational amplifiers (op-amps)
Main articles: operational amplifier and instrumentation amplifier
An operational amplifier is a solid state integrated circuit amplifier which employs external feedback for control of its transfer function or gain.

[edit] Fully differential amplifiers (FDA)
Main article: fully differential amplifier
A fully differential amplifier is a solid state integrated circuit amplifier which employs external feedback for control of its transfer function or gain. It is similar to the operational amplifier but it also has differential output pins.

[edit] Video amplifiers
These deal with video signals and have bandwidths of about 5 MHz. Certain requirements for step response and overshoot are necessary in order for acceptable TV images to be presented.

[edit] Oscilloscope vertical amplifiers
These are used to deal with video signals to drive an oscilloscope display tube and can have bandwidths of about 500 MHz. The specifications on step response, rise time, overshoot and aberrations can make the design of these amplifiers extremely difficult. One of the pioneers in high bandwidth vertical amplifiers was the Tektronix company.

[edit] Distributed amplifiers
Main article: Distributed Amplifier
These use transmission lines to temporally split the signal and amplify each portion separately in order to achieve higher bandwidth than can be obtained from a single amplifying device. The outputs of each stage are combined in the output transmission line. This type of amplifier was commonly used on oscilloscopes as the final vertical amplifier. The transmission lines were often housed inside the display tube glass envelope.

[edit] Microwave amplifiers

[edit] Travelling wave tube (TWT) amplifiers
Main article: Traveling wave tube
Used for high power amplification at low microwave frequencies. They typically can amplify across a broad spectrum of frequencies; however, they are usually not as tunable as klystrons.

[edit] Klystrons
Main article: Klystron
Very similar to TWT amplifiers, but more powerful and with a specific frequency "sweet spot". They generally are also much heavier than TWT amplifiers, and are therefore ill-suited for light-weight mobile applications. Klystrons are tunable, offering selective output within their specified frequency range.

[edit] Musical instrument (audio) amplifiers
Main articles: instrument amplifier and audio amplifier
An audio amplifier is usually used to amplify signals such as music or speech.

[edit] Other amplifier types

[edit] Carbon microphone
One of the first devices used to amplify signals was the carbon microphone (effectively a sound-controlled variable resistor). By channeling a large electric current through the compressed carbon granules in the microphone, a small sound signal could produce a much larger electric signal. The carbon microphone was extremely important in early telecommunications; analog telephones in fact work without the use of any other amplifier. Before the invention of electronic amplifiers, mechanically coupled carbon microphones were also used as amplifiers in telephone repeaters for long distance service.

[edit] Magnetic amplifier
Main article: magnetic amplifier
A magnetic amplifier is a transformer-like device that makes use of the saturation of magnetic materials to produce amplification. It is a non-electronic electrical amplifier with no moving parts. The bandwidth of magnetic amplifiers extends to the hundreds of kilohertz.

An Amplidyne or Rototrol is a rotating machine like an electrical generator that provides amplification of electrical signals by the conversion of mechanical energy to electrical energy.

[edit] Optical amplifiers
Main article: Optical amplifier
Optical amplifiers amplify light through the process of stimulated emission.

[edit] Miscellaneous types
There are also mechanical amplifiers, such as the automotive servo used in braking.
Relays can be included under the above definition of amplifiers, although their transfer function is not linear (that is, they are either open or closed).
Also purely mechanical manifestations of such digital amplifiers can be built (for theoretical, didactical purposes, or for entertainment), see e.g. domino computer.
Another type of amplifier is the fluidic amplifier, based on the fluidic triode.

Sound Recording And Reproduction

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Contents [hide]
1 The cylinder phonograph
2 The disc
3 Electrical recording
4 Other recording formats
5 Magnetic tape
6 Stereo and Hi-fi
7 The Fifties and beyond
8 Digital recording
9 Voice to note
10 See also
11 Notes
12 External links

[edit] The cylinder phonograph
The first practical sound recording and reproduction device was the mechanical cylinder phonograph, invented by Thomas Edison in 1877 and patented in 1878, and in some ways resembled the phonoautograph patented by Edouard-Leon Scott de Martinville in 1857. The invention soon spread across the globe and over the next two decades the commercial recording, distribution and sale of sound recordings became a growing new international industry, with the most popular titles selling millions of units by the early 1900s. The development of mass-production techniques enabled cylinder recordings to become a major new consumer item in industrial countries and the cylinder was the main consumer format from the late 1880s until around 1910.

[edit] The disc
The next major technical development was the invention of the gramophone disc, generally credited to Emile Berliner and commercially introduced in the United States in 1889.

Discs were easier to manufacture, transport and store, and they had the additional benefit of being louder (marginally) than cylinders, which by necessity, were single-sided. Sales of the Gramophone record overtook the cylinder ca. 1910, and by the end of World War I the disc had become the dominant commercial recording format. In various permutations, the audio disc format became the primary medium for consumer sound recordings until the end of the 20th century, and the double-sided 78rpm shellac disc was the standard consumer music format from the early 1910s to the late 1950s.

Although there was no universally accepted speed, and various companies offered discs that played at several different speeds, the major recording companies eventually settled on a de facto industry standard of 78 revolutions per minute, which gave the disc format its common nickname, the "seventy-eight". Discs were made of shellac or similar brittle plastic like materials, played with metal needles, and had a distinctly limited life.

[edit] Electrical recording
Sound recording began as a mechanical process and remained so until the 1920s (with the exception of the 1898 Telegraphone) when a string of groundbreaking inventions in the field of electronics revolutionised sound recording and the young recording industry. These included sound transducers such as microphones and loudspeakers, and various electronic devices such as the mixing desk, designed for the amplification and modification of electrical sound signals.

After the Edison phonograph itself, arguably the most significant advances in sound recording were the electronic systems invented by two American scientists between 1900 and 1924.

In 1906 Lee De Forest invented the "Audion" triode vacuum-tube, electronic valve, which could greatly amplify weak electrical signals, (one early use was to amplify long distance telephone in 1915) which became the basis of all subsequent electrical sound systems until the invention of the transistor. The valve was quickly followed by the invention of the Regenerative circuit, Super-Regenerative circuit and the Superheterodyne receiver circuit, all of which were invented and patented by the young electronics genius Edwin Armstrong between 1914 and 1922. Armstrong's inventions made higher fidelity electrical sound recording and reproduction a practical reality, facilitating the development of the electronic amplifier and many other devices; after 1925 these systems had become standard in the recording and radio industry. Armstrong's groundbreaking inventions (including FM radio) also made possible the broadcasting of long-range, high-quality radio transmissions of voice and music. The importance of Armstong's Superheterodyne circuit cannot be under-estimated -- it was the central component of almost all analog amplification and radio-frequency transmitter and receiver devices of the 20th century.

Beginning during World War One, experiments were undertaken in the United States and Great Britain to reproduce among other things, the sound of a Submarine (u-boat) for training purposes. The acoustical recordings of that time proved entirely unable to reproduce the sounds, and other methods were actively sought. Radio had developed independently to this point, and now Bell Laboritories sought a marriage of the two disparate technologies, greater than the two separately. The first experiments were not very promising, but by 1920 greater sound fidelity was achieved using the electrical system than had ever been realized acoustically. One early recording made without fanfare or announcement was the dedication of the Tomb of the Unknown Soldier at Arlington Cemetery.

By early 1924 such dramatic progress had been made, that Bell Labs arranged a demonstration for the leading recording companies, Victor Talking Machine, and Columbia Phonograph Co's.

Columbia, always in financial straits, could not afford it, and Victor, essentially leaderless since the Mental collapse of Founder E. Johnson, left the demonstration without comment. English Columbia, by then a separate Company, got a hold of a test pressing from these sessions, and realized the immediate and urgent need to have the new system. Bell was only offering its method to United States Companies, and to circumvent this, Managing Director Louis Sterling of British Columbia, bought his once parent company, and signed up for electrical recording. When Victor Talking Machine was apprised of the Columbia deal, they too quickly signed. Columbia made its first electrical recordings on February 25, 1925 with Victor following a few weeks later. The two then agreed privately to "be quiet" until November 1925, by which time enough electrical repretory would be available.

[edit] Other recording formats
This period also saw several other historic developments including the introduction of the first practical magnetic sound recording system, the magnetic wire recorder, which was based on the work of Danish inventor Valdemar Poulsen. Magnetic wire recorders were effective, but the sound quality was poor, so between the wars they were primarily used for voice recording and marketed as business dictating machines.

In the 1930s radio pioneer Guglielmo Marconi developed a system of magnetic sound recording using steel tape. This was the same material used to make razor blades, and not surprisingly the fearsome Marconi-Stille recorders were considered so dangerous that technicians had to operate them from another room for safety. Because of the high recording speeds required, they used enormous reels about one metre in diameter, and the thin tape frequently broke, sending jagged lengths of razor steel flying around the studio.

The other major invention in sound recording in this period was the optical sound-on-film system, also generally credited to Lee De Forest. Although famous early "Talkies" like The Jazz Singer used a sound-on-disc system, the film industry eventually adopted the optical sound-on-film system and it revolutionised the movie industry in the 1930s, ushering in the era of 'talking pictures'. Optical sound-on-film, based on the photoelectric cell, became the standard film audio system throughout the world until it was superseded in the 1960s.

[edit] Magnetic tape
The other major inventions of this period were magnetic tape and the tape recorder(Telegraphone). Paper-based tape was first used but was soon superseded by polyester and acetate backing due to dust drop and hiss. Acetate was more brittle than polyester and snapped easily. This technology, the basis for almost all commercial recording from the 1950s to the 1980s, was invented by German audio engineers in the 1930s, who also discovered the technique of AC biasing, which dramatically improved the frequency response of tape recordings. Tape recording was perfected just after the war by American audio engineer John T. Mullin, whose pioneering recorders were based on captured German recorders, and the Ampex company produced the first commercially available tape recorders in the late 1940s.

Magnetic tape brought about sweeping changes in both radio and the recording industry. Sound could be recorded, erased and re-recorded on the same tape many times, sounds could be duplicated from tape to tape with only minor loss of quality, and recordings could now be very precisely edited by physically cutting the tape and rejoining it.

Within a few years of the introduction of the first commercial tape recorder, the Ampex 200 model, launched in 1948, American musician-inventor Les Paul had invented the first multitrack tape recorder, bringing about another technical revolution in the recording industry. Tape made possible the first sound recordings totally created by electronic means, opening the way for the bold sonic experiments of the Musique Concrète school and avant garde composers like Karlheinz Stockhausen, which in turn led to the innovative pop music recordings of artists such as Frank Zappa, The Beatles and The Beach Boys.

Tape enabled the radio industry for the first time to pre-record many sections of program content such as advertising, which formerly had to be presented live, and it also enabled the creation and duplication of complex, high-fidelity, long-duration recordings of entire programs. It also, for the first time, allowed broadcasters, regulators and other interested parties to undertake comprehensive logging of radio broadcasts. Innovations like multitracking and tape echo enabled radio programs and advertisements to be pre-produced to a level of complexity and sophistication that was previously unattainable and tape also led to significant changes to the pacing of program content, thanks to the introduction of the endless-loop tape cartridge.

The vinyl microgroove record was introduced in the late 1940s, and the two main vinyl formats -- the 7-inch single turning at 45 rpm and the 12-inch LP (long-playing) record turning at 33⅓ rpm -- had totally replaced the 78 rpm shellac disc by the end of the 1950s. Vinyl offered improved performance, both in stamping and in playback, and came to be generally played with polished diamond styli, and when played properly (precise tracking weight, etc.) offered longer life. Vinyl records were, over-optimistically, advertised as "unbreakable". They were not, but were much less brittle and breakable than shellac. Nearly all were tinted black, but some were colored, as red, swirled, translucent, etc.

[edit] Stereo and Hi-fi
Magnetic tape also enabled the development of the first practical commercial sound systems that could record and reproduce high-fidelity stereophonic sound. Experiments with stereo dated back to the 1880s and during the 1930s and 1940s there were many attempts to record in stereo using discs, but these were hampered by problems with synchronization.

The first major breakthrough in practical stereo sound was made by Bell Laboratories, who in 1937 demonstrated a practical system of two-channel stereo, using dual optical sound tracks on film. Major movie studios quickly developed three-track and four-track sound systems, and the first stereo sound recording in a commercial film was made by Judy Garland for the MGM movie Listen, Darling in 1938. The first commercially-released movie with a full stereo soundtrack was Walt Disney's Fantasia, released in 1940.

German audio engineers working on magnetic tape are reported to have developed stereo recording by 1943, but it was not until the introduction of the first commercial two-track tape recorders by Ampex in the late 1940s that stereo tape recording became commercially feasible. However, despite the availability of multitrack tape, stereo did not become the standard system for commercial music recording for some years and it remained a specialist market during the 1950s. This changed after the late 1957 introduction of the "Westrex stereo phonograph disc".

Most pop singles were mixed into monophonic sound until the mid 1960s, it was common for major pop releases to be issued in both mono and stereo until the early 1970s. Many Sixties pop albums now available only in stereo were originally intended to be released only in mono, and the so-called "stereo" version of these albums were created by simply separating the two tracks of the master tape. In the mid Sixties, as stereo became more popular, many mono recordings (such as The Beach Boys' Pet Sounds) were remastered using the so-called "fake stereo" method, which spread the sound across the stereo field by directing higher-frequency sound into one channel and lower-frequency sounds into the other.

[edit] The Fifties and beyond
Magnetic tape transformed the recording industry, and by the late-1950s the vast majority of commercial recordings were being mastered on tape. The electronics revolution that followed the invention of the transistor brought other radical changes, the most important of which was the introduction of the world first "personal music device", the miniaturized transistor radio, which became a major consumer luxury item in the 1960s, transforming radio broadcasting from a static group experience into a mobile, personal listening activity.

The first multitrack recording made using magnetic tape was "How High the Moon" by Les Paul, on which Paul played eight overdubbed guitar tracks. In the 1960s Brian Wilson of The Beach Boys, Frank Zappa and The Beatles (with producer George Martin) were among the first popular artists to explore the possibilities of multitrack techniques and effects on their landmark albums Pet Sounds, Freak Out! and Sgt. Pepper's Lonely Hearts Club Band.

The next important innovation was the compact cassette, introduced by the Philips electronics company in 1964. The cassette became a major consumer audio format and advances in microelectronics eventually led to the development of the Sony Walkman, introduced in the 1970s, which gave a major boost to the mass distribution of music recordings. Cassettes became the first successful consumer recording/re-recording medium as opposed to the gramophone record, which was a pre-recorded playback medium.

A key advance in audio fidelity came with the Dolby A noise reduction system, invented by Ray Dolby and introduced in 1966. Dolby's noise reduction system, which greatly improved the sound of cassette tape recordings, also found wide application in the recording and film industries. Dolby A was crucial to the popularisation and commercial success of the compact cassette as a domestic recording and playback medium, and became a part of the booming "hi-fi" market of the 1970s and beyond.

The multitrack audio cartridge was in wide use in the radio industry, from the late 1950s to the 1980s, but in the 1960s the pre-recorded 8-track cartridge was launched as a consumer audio format. Aimed particularly at the automotive market, they were the first practical, affordable car hi-fi systems, and they offered superior sound quality to the compact cassette. However the smaller size and greater durability -- augmented by the ability to create home-recorded music "compilations" -- saw the cassette become the dominant consumer format for portable audio devices in the 1970s and 1980s.

There had been experiments with multi-channel sound for many years -- usually for special musical or cultural events -- but the first commercial application of the concept came in the early 1970s with the introduction of Quadraphonic sound. This spin-off development from multitrack recording used four tracks (instead of the two used in stereo) and four speakers to create a 360-degree audio field around the listener. Following the release of the first consumer 4-channel hi-fi systems, a number of popular albums were released in the Quadraphonic format; among the best known are Mike Oldfield's Tubular Bells and Pink Floyd's The Dark Side of the Moon. Quadraphonic sound was not a commercial success, and it eventually faded out in the late 1970s, although this early venture paved the way for the eventual introduction of domestic Surround Sound systems, which have gained enormous popularity since the introduction of the DVD.

The replacement of the thermionic valve(vacuum tube) by the smaller, cooler and less power-hungry transistor also accelerated the sale of consumer high-fidelity "hi-fi" sound systems from the 1960s onward. In the 1950s most record players were monophonic and relatively low fidelity in sound quality, and few consumers could afford high-quality stereophonic sound systems. In the 1960s American manufacturers introduced a new generation of "modular" hi-fi components -- turntables, integrated amplifiers, tape recorders and other ancillary equipment (like the graphic equaliser), which could be connected together to create a complete home sound system. These developments were rapidly taken up by the Japanese electronics companies, which soon flooded the world market with relatively cheap, high-quality components. By the 1980s, corporations like Sony had become world leaders in the music recording and playback industry.

[edit] Digital recording
The invention of digital sound recording and the compact disc in 1983 brought significant improvements in the durability of consumer recordings. The CD initiated another massive wave of change in the consumer music industry, with vinyl records effectively relegated to a small niche market by the mid-1990s.

The most recent and revolutionary developments have been in digital recording, with the invention of purely electronic consumer recording formats such as the WAV digital music file and the compressed file type, the MP3. This generated a new type of portable solid-state computerised digital audio player, the MP3 player. Another invention, by Sony, was the minidisc player, using ATRAC compression on small, cheap, re-writeable discs. This was in vogue in the 1990s, and is still popular, especially in a newer, longer playing and higher fidelity version. New technologies such as Super Audio CD, DVD-A, Blu ray Disc and HD DVD continue to set very high standard in evolution of digital audio storage.

This technology spreads across various associated fields, from hi-fi to professional audio, internet radio and podcasting.

Technological developments in recording and editing have transformed the record, movie and television industries in recent decades. Audio editing became practicable with the invention of magnetic tape recording, but the use of computers has made editing operations faster and easier to execute, and the use of hard-drives for storage has made recording cheaper. We now divide the process of making a recording into tracking, mixing and mastering. Multitrack recording makes it possible to capture sound from several microphones, or from different 'takes' to tape or disc with maximum headroom and quality, allowing maximum flexibility in the mixing and mastering stages for editing, level balancing, compressing and limiting, and the addition of effects such as reverberation, equalisation, flanging and many more.

In the 1920s, the first talkies came out, featuring the new sound-on-film technology which used photoelectric cells to record and reproduce sound signals that were optically recorded directly onto the movie film. The advent of talkies, spearheaded by The Jazz Singer in 1927, saw the rapid demise of live cinema musicians and orchestras, which were replaced with pre-recorded soundtracks, causing 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]

Recording Studio

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A recording studio is a facility for sound recording. Ideally, the space is specially designed by an acoustician to control audio reflections. Different types of studios record bands and artists, voiceovers and music for television shows, movies, cartoons, and commercials, and/or even record a full orchestra. The typical recording studio consists of a room called the "studio", where instrumentalists and vocalists perform; and the "control room", which houses the equipment for recording, routing and manipulating the sound. Often, there will be smaller rooms called "isolation booths" present to accomodate loud instruments such as drums or electric guitar, to keep these sounds from being audible to the microphones that are capturing the sounds form other instruments or vocalists.

Photo of a control room showing the "desk" area. Visible is a computer-based "DAW". To the right of the computer and displays is a digital mixing console. Portions of reference monitors can also be seen.
A vocalist inside the "studio", or "live room" area of a recording studio.

Siemens Sound Studio ca. 1956.Contents [hide]
1 History
1.1 1890s to 1930s
1.2 1940s to 1970s
2 Project studios
3 Isolation booth
4 See also
5 External links

[edit] History

[edit] 1890s to 1930s
In the era of acoustic recordings (prior to the introduction of microphones, electrical recording and amplification) the earliest recording studios were very basic facilities, being essentially soundproof rooms that isolated the performers from outside noise. During this era it was not uncommon for recordings to be made in any available location, such as a local ballroom, using portable acoustic recording equipment.

In this period, master recordings were made by a direct-to-disc cutting process -- performers were typically grouped around a large acoustic horn (an enlarged verion of the familiar phonograph horn) and the acoustic energy from the voices and/or instruments was channeled through the horn's diaphragm to a mechanical cutting lathe located in the next room, which inscribed the signal as a modulated groove directly onto the surface of the master cylinder or disc.

Following the invention and commercial introduction of the microphone, the electronic amplifier, the mixing desk and the loudspeaker, the recording industry gradually converted to electric recording and this technology had almost totally replaced mechanical acoustic recording methods by 1933.

[edit] 1940s to 1970s
Electrical recording was common by the early 1930s, and mastering lathes were now electrically powered, but master recordings still had to be cut direct-to-disc. In line with the prevailing musical trends, studios in this period were primarily designed for the live recording of symphony orchestras and other large instrumental ensembles. Engineers soon found that large, reverberant spaces like concert halls created a vibrant acoustic signature that greatly enhanced the sound of the recording, and in this period large, acoustically "live" halls were favored, than the acoustically "dead" booths and studio rooms that became common after the 1960s.

Because of the limits of the recording technology, studios of the mid-20th century were designed around the concept of grouping musicians and singers, rather than separating them, and placing the performers and the microphones strategically to capture the complex acoustic and harmonic interplay that emerged during the performance. Modern sound stages still sometimes use this approach for large film scoring projects today

Because of their superb acoustics, many of the larger studios were converted churches. Examples include George Martin's AIR Studios in London, the famed Columbia Records 30th Street Studio in New York City (a converted Armenian church, with a ceiling over 100 feet high), and the equally famous Decca Records Pythian Temple studio in New York (where artists like Louis Jordan, Bill Haley and Buddy Holly were recorded) which was also a large converted church that featured a high, domed ceiling in the center of the hall.

Electric recording studios in the mid-20th century often lacked isolation booths, baffles, and sometimes even speakers, and it was not until the 1960s, with the introduction of the high-fidelity headphones that it became common practice for performers to use headsets to monitor their performance during recording and listen to playbacks.

It was difficult to isolate all the performers -- a major reason that this practice was not used was simply because recordings were usually made as live ensemble 'takes' and all the performers needed to be able to see each other and the ensemble leader while playing. The recording engineers who trained in this period learned to take advantage of the complex acoustic effects that could be created though "leakage" between different microphones and groups of instruments, and these technicians became extremely skilled at capturing the unique acoustic properties of their studios and the musicians in performance.

Facilities like the Columbia Records 30th Street Studio in New York and EMI's Abbey Road Studio in London were renowned for their 'trademark' sound -- which was (and still is) easily identifiable by audio professionals -- and for the skill of their staff engineers.

The use of different kinds of microphones and their placement around the studio was a crucial part of the recording process, and particular brands of microphone were used by engineers for their specific audio characteristics. The smooth-toned ribbon microphones developed by the RCA company in the 1930s were crucial to the 'crooning' style perfected by Bing Crosby, and the famous Neumann U47 condenser microphone was one of the most widely used from the 1950s. This model is still widely regarded by audio professionals as one of the best microphones of its type ever made.

Learning the correct placement of microphones was a major part of the training of young engineers, and many became extremely skilled in this craft. Well into the 1960s, in the classical field it was not uncommon for engineers to make high-quality orchestral recordings using only one or two microphones suspended above the orchestra.

In the 1960s, engineers began experimenting with placing microphones much closer to instruments than had previously been the norm. The distinctive rasping tone of the horn sections on the Beatles recordings "Good Morning Good Morning" and "Lady Madonna" were achieved by having the saxophone players position their instruments so that microphones were virtually inside the mouth of the horn.

The unique sonic characteristics of the major studios imparted a special character to many of the most famous popular recordings of the 1950s and 1960s, and the recording companies jealously guarded these facilities. According to sound historian David Simons, after Columbia took over the 30th Street Studios in the late 1940s, A&R manager Mitch Miller issued a standing order that the drapes and other fittings left by the previous occupants were not to be touched, and the cleaners had specific orders never to mop the bare wooden floor for fear it might alter the acoustic properties of the hall.

There were several other features of studios in this period that contributed to their unique "sonic signatures". As well as the inherent sound of the large recording rooms, many of the best studios incorporated specially-designed echo chambers, purpose-built rooms which were often built beneath the main studio.

These were typically long, low rectangular spaces constructed from hard, sound-reflective materials like concrete, fitted with a loudspeaker at one end and one or more microphones at the other. During a recording session, a signal from one or more of the microphones in the studio could be routed to the loudspeaker in the echo chamber; the sound from the speaker reverberated through the chamber and the enhanced signal was picked up by the microphone at the other end. This echo-enhanced signal -- which was often used to 'sweeten' the sound of vocals -- could then be blended in with the primary signal from the microphone in the studio and mixed into the track as the master recording was being made.

Special equipment was another notable feature of the "classic" recording studio. The biggest studios were owned and operated by large media companies like RCA, Columbia and EMI, who typically had their own electronics research and development divisions that designed and built custom-made recording equipment and mixing consoles for their studios.

Likewise, the smaller independent studios were often owned by skilled electronics engineers who designed and built their own desks and other equipment. A good example of this is the famous Gold Star Studios in Los Angeles, the site of many famous American pop recordings of the 1960s. Co-owner David S. Gold built the studio's main mixing desk and many additional pieces of equipment and he also designed the studio's unique trapezoidal echo chambers.

During the 1950s and 1960s the sound of pop recordings was further defined by the introduction of proprietary sound processing devices such as equalizers and compressors, which were manufactured by specialist electronics companies. One of the best known of these was the famous Pultec equalizer, which was used by almost all the major commercial studios of the time.

With the introduction of multi-track recording, it became possible to record instruments and singers separately and at different times on different tracks on tape, although it was not until the 1970s that the large recording companies began to adopt this practice widely, and throughout the Sixties many "pop" classics were still recorded live in a single take.

After the Sixties the emphasis shifted to isolation and sound-proofing, with treatments like echo and reverberation added separately during the mixing process, rather than being blended in during the recording. One regrettable outcome of this trend, which coincided with rising inner-city property values, was that many of the largest studios were either demolished or redeveloped for other uses.

In the 1960s, recordings were analog recordings made using ¼-inch or ½-inch two-track magnetic tape. By the early 1970s, the technology progressed to using various types of multi-track tape. The most common of which is the 2-inch analog tape, capable of containing up to 24 individual tracks. Generally after an audio mix is set up on a 24-track tape machine, the signal is played back and sent to a different machine which records the combined signals (called printing) to a ½-inch 2-track stereo tape, called a master.

Prior to digital recording, the total number of available tracks onto which one could record was measured in multiples of 24, based on the number of 24-track tape machines being used. Presently, most recording studios now use digital recording equipment which only limits the number of available tracks based on the capacity of the mixing console or computer hardware interface.

Analog tape machines are still well sought after as some purists label digitally recorded audio as sounding too harsh, and the scarcity and age of analog tape machines greatly increases their value, as does the fact that many audio engineers still insist on recording only to analog tape. This harshness is widely attributed by them to the fact that digital recording will sample a sound wave many times per second allowing an illusion of solid sound waves to be created, where in contrast, analog tape captures a sound wave in its entirety.

However, others simply argue that the lack of high frequency noise and the higher fidelity of the digital medium make the recorded higher frequencies more prominent, which results in such perceived harshness in contrast to analog recording. Still others point to problems of early digital recordings caused by the inexperience of sound engineers with the new medium as the cause for critics to the digital systems. Finally, another possibly relevant effect derives from the fact that, since CD-quality audio uses a sampling rate of 44.1 kHz, no frequencies above the Nyquist frequency of 22050 Hz are acceptable for recording, otherwise aliasing occurs. Because of that, very steep low-pass filters are used on frequencies above 20 kHz (the theoretical limit for human hearing) that introduce slight distortions into the audible-range signal. This is one of the several reasons for the push on high-end equipment towards higher sampling rates, such as 48 kHz (used in video production), 88.2 kHz, 96 kHz and even 192 kHz.

[edit] Project studios
A small, personal recording studio is sometimes called a project studio or home studio. Such studios often cater to specific needs of an individual artist, or are used as a non-commercial hobby. The first modern project studios came into being during the late 1980s, with the advent of affordable multitrack recorders, synthesizers and microphones. The phenomenon has flourished with falling prices of MIDI equipment and accessories, as well as inexpensive digital hard-disk recording products.

Recording drums and electric guitar in a home studio is challenging, because they are usually the loudest instruments. Conventional drums require soundproofing in this scenario, unlike electronic or sampled drums. Getting an authentic electric guitar amp sound including power-tube distortion requires a power attenuator (either power-soak or power-supply based) or an isolation box or booth. A convenient compromise is amp simulation, whether a modelling amp, preamp/processor, or software-based guitar amp simulator. Sometimes, musicians replace loud, inconvenient instruments such as drums, with keyboards, which today often provide highly realistic sampling.

[edit] Isolation booth
An isolation booth is a standard small room in a recording studio, which is soundproofed to keep out external sounds and keep in the internal sounds. A drummer, vocalist, or guitar speaker cabinet, along with microphones, is acoustically isolated in the room. A professional recording studio has a control room, a large live room, and one or more small isolation booths. All rooms are soundproofed such as with double-layer walls with dead space and insulation in-between the two walls, forming a room-within-a-room.

There are variations of the same concept, including a portable standalone isolation booth, a compact guitar speaker isolation cabinet, or a larger guitar speaker cabinet isolation box.

A gobo panel achieves the same idea to a much more moderate extent; for example, a drum kit that is too loud in the live room or on stage can have acrylic glass see-through gobo panels placed around it to deflect the sound and keep it from bleeding into the other microphones, allowing more independent control of each instrument channel at the mixing board.

All rooms in a recording studio may have a reconfigurable combination of reflective and non-reflective surfaces, to control the amount of reverberation.

Producers And Modern Recording Technology

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In modern digital music, the producer is often the only person involved in the creation of a musical recording, and is responsible for both writing, performing, recording and arranging the material. The term "producer" is nearly synonymous with "musician" in this field. This change has been partly due to the increase of inexpensive yet powerful music production software, which allows for entire tracks to be composed, arranged and recorded at home on a PC or laptop, allowing the traditional roles of a team of people to be performed by one individual. Popular PC music production software packages include Ableton Live, Cakewalk SONAR, Cubase, FL Studio, Garage Band, Logic Pro, Pro Tools, Reason, and Sony ACID Pro. With the increased portability of digital recording equipment live album production has become much more cost effective than in the past couple of decades. This has resulted in thousands of live music recordings flooding the internet and music stores.

Evolution Of The Role Of The Producer

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Prior to the 1950s, the various stages of the recording and marketing process had been carried out by different professionals within the industry -- A&R managers found potential new artists and signed them to their labels; professional songwriters created new material; publishing agents sold these songs to the A&R people; staff engineers carried out the task of making the recordings in company-owned studios.

Freed from this traditional system by the advent of independent commercial studios, the new generation of entrepreneurial producers -- many of whom were former record company employees themselves -- were able to create and occupy a new stratum in the industry, taking on a more direct and complex role in the musical process. This development in music was mirrored in the TV industry by the concurrent development of videotape recording and the consequent emergence of independent TV production companies like Desilu, established by '50s TV superstars Lucille Ball and her then husband Desi Arnaz.

These producers now typically carried out most or all of these various tasks themselves, including selecting and arranging songs, overseeing sessions (and often engineering the recordings) and even writing the material. Independent music production companies rapidly gained a significant foothold in popular music and soon became the main intermediary between artist and record label, signing new artists to production contracts, producing the recordings and then licensing the finished product to record labels for pressing, promotion and sale. (This was a novel innovation in the popular music field, although a broadly similar system had long been in place in many countries for the production of content for broadcast radio.) The classic example of this transition is renowned British producer George Martin, who worked as a staff producer and A&R manager at EMI for many years, before branching out on his own and becoming a highly successful independent producer.

As a result of these changes, record producers began to exert a strong influence, not only on individual careers, but on the course of popular music. Other notable past and present independent producers include Don Kirshner (The Monkees), Mickie Most (The Animals, Herman's Hermits, Donovan), Tony Visconti (David Bowie, T. Rex), Rick Rubin (Metallica, Beastie Boys, Red Hot Chili Peppers, Linkin Park), Nigel Godrich (Radiohead, Beck, Travis), RZA (Wu-Tang Clan, Method Man, Ghostface, Raekwon), Dr. Dre (N.W.A, Eazy-E, Snoop Dogg, Eminem, 50 Cent, The Game), Norman Whitfield (The Temptations), and Timbaland (Justin Timberlake, Nelly Furtado, Jay-Z, Aaliyah, Missy Elliott).

Realising the potential for creating recordings that could match their musical vision, many successful recording artists have become producers in their own right. Examples are Trent Reznor, Steven Wilson, Nile Rodgers, Devin Townsend, Ken Andrews, Jeff Lynne, Brian Wilson and Brian Eno.

Some producers also became de facto recording artists, often creating records with anonymous studio musicians and releasing them under a pseudonym. Examples of this phenomenon include the records by fictional groups The Archies and Josie & The Pussycats, produced by Don Kirshner and Danny Jansen respectively, who were contracted by TV production companies to produce these records to promote the animated children's TV series of the same name. Similarly, Jeff Barry and Andy Kim recorded as The Archies.

Early Record Producers

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During the 1890s, Fred Gaisberg ran the first recording studio and provided the closest approximation of production by guiding an opera singer closer or further away from a gramophone's horn to match the dynamics in the score. (Citation: Gronow and Saunio 1998, p.8; Moorefield 2005, p.1).

However, within the first half of the 20th century, the record producer's role was comparable to that of a film producer, in that the record producer organized and supervised recording sessions, paid technicians, musicians and arrangers, and sometimes chose material for the artist.In the mid-1950s a new category emerged, that of the independent record producer. Among the most famous early independent producers are the famed songwriting-production duo Leiber & Stoller, "Wall of Sound" creator Phil Spector and British studio pioneer Joe Meek.

Magnetic tape enabled the establishment of independent recording studios in major recording centres such as London, Los Angeles and New York. Unlike the old record company studios, which were effectively a "closed shop," these new studios could be hired by the hour by anyone who could afford to do so.

The biggest and best commercial studios were typically established and operated by leading recording engineers. They were carefully constructed to create optimum recording conditions, and were equipped with the latest and best recording equipment and top-quality microphones, as well as electronic amplification gear and musical instruments.

Top-line studios such as Olympic Studios in London, United Western Recorders, Fine Recording in New York City, and Musart in Los Angeles quickly became among the most sought-after recording facilities in the world, and both these studios became veritable "hit factories" that produced many of the most successful pop recordings of the latter 20th century.

Record Producer

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In the music industry, a record producer (or music producer) has many roles, among them controlling the recording sessions, coaching and guiding the musicians, organizing and scheduling production budget and resources, and supervising the recording, mixing and mastering processes. This has been a major function of producers since the inception of sound recording, but in the later half of the 20th century producers also took on a wider entrepreneurial role. These activities comprise record production.

The music producer could be compared to the film director in that the producer's job is to create, shape and mould a piece of music in accordance with their vision for the album.

Acoustical Engineering

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Acoustical engineering is the branch of engineering dealing with sound and vibration. It is closely related to acoustics, the science of sound and vibration. Acoustical engineers are typically concerned with:

how to reduce unwanted sounds
how to make useful sounds
using sound as an indication of some other physical property
The art of reducing unwanted sounds is called noise control. Noise control engineers work with engineers in most industries to ensure that their products and processes are quiet.

The art of producing useful sounds includes the use of ultrasound for medical diagnosis, sonar, and sound reproduction.

A separate and related discipline, audio engineering, is the art of recording and reproducing speech and music for human use.

Practitioners

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An engineer at one of the audio consoles of the Danish Broadcasting Corporation (Danmarks Radio). The console is an NP-elektroakustik specially made for Danmarks Radio in the eighties.An audio engineer is someone with experience and training in the production and manipulation of sound through mechanical (analog) or digital means. As a professional title, this person is sometimes designated as a sound engineer or recording engineer instead. A person with one of these titles is commonly listed in the credits of many commercial music recordings (also in other productions that include sound, such as movies).

Audio engineers are generally familiar with the design, installation, and/or operation of sound recording, sound reinforcement, or sound broadcasting equipment. In the recording studio environment, the audio engineer is a person recording, editing, manipulating, mixing, and/or mastering sound by technical means in order to realize an artist's or record producer's creative vision. While usually associated with music production, an audio engineer may be involved in dealing with sound for a wide range of applications, including post-production for video and film, live sound reinforcement, advertising, multimedia, and broadcasting.

Some well-known sound engineers include Mick Guzauski, Roger Nichols, Bill Porter, Al Schmitt, and Eberhard Sengpiel.

Audio engineers operate mixing consoles, microphones, signal processors, tape machines, digital audio workstations, sequencing software, and speaker systems. Commonly an audio engineer is responsible for the technical aspects of a sound recording or other audio production, and works together with a record producer or director, although the engineer's role may also be integrated with that of the producer.

In typical sound reinforcement applications, audio engineers often assume the role of producer, making artistic decisions along with technical ones.

[edit] See also

Audio Engineering

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Audio engineering is a part of audio science dealing with the recording and reproduction of sound through mechanical and electronic means. The field of audio engineering draws on many disciplines, including electrical engineering, acoustics, psychoacoustics, and music. Unlike acoustical engineering, audio engineering generally does not deal with noise control or acoustical design. However, an audio engineer is often more affiliated with the creative aspects of audio rather than formal engineering, most professional audio engineers lack a formal and accredited Engineering degree. Audio engineering is different from acoustical engineering, which heavily relies on the underlying physics and mathematics of sound waves and their propagation.