Tue 26 Sep 06 @ 2:31 pm

Here is my playlist. Genre: electro, tech-house


1. Outwork - Elektro (Krispino Thump remix)
2. Love Connection - The Bomb (Phats & Small dub mix)
3. Antonie Clamaran & Mario Ochoa - Can You Feel It (Crazee remix)
4. Tomas Andersson - Washing Up (Tiga remix)
5. Romand Flugel - Geht's Noch (club mix)
6. Paul Masterson ft. Subway - What You Got, What You Do (club mix)
7. Antonie Clamaran - Take Off (club mix)

Mon 25 Sep 06 @ 11:29 am

Yes, it's finished. And it was great!
Here are few photos from festival. It was amazing ;) So many VDJ users and so many talents :)

Tue 30 May 06 @ 10:22 pm

SUBWOOFERS

History

The first commercial subwoofer was developed during the 1970s by Ken Kreisel, current president of M&K Sound/Miller & Kreisel Corporation in Los Angeles. Kreisel's business partner, Jonas Miller, owned a high-end audio store in Los Angeles, and customers of some of the highest quality electrostatic speakers complained about a reduction of bass response in the electrostatics, compared to conventional loudspeakers; Kreisel's solution was to design a powered loudspeaker that would reproduce only those frequencies that were too low for the electrostatic speakers to convey and thereby fill in the missing sonic information[1]. The first known use of a subwoofer in a recording session was for the mixing of the Steely Dan album Pretzel Logic when recording engineer Roger Nichols arranged for Kreisel to bring a prototype of his subwoofer to Village Recorders. Further design modifications were made by Kreisel over the next ten years (and continuing to the present day), and in the 1970s and 1980s by engineer John P. D'Arcy; record producer Daniel Levitin served as a consultant and "golden ears" for the design of the crossover network (used to partition the frequency spectrum so that the subwoofer would not attempt to reproduce frequencies that were too high for its effective range, and so that the main speakers would not need to process frequencies that were too low for their effective range).


Overview

Subwoofers use drivers (woofer) with cones typically coming in 10" or 12" sizes, but can be as large as 34", and as small as 4".

Nearly all woofers are driven by a voice coil in a magnetic field, connected to an amplifier. The voice coil assembly is basically an electric motor. As voltage is applied to the coil, it generates an electromagnetic field. This field is either repulsed or attracted to the fixed field of the magnet which surrounds the voice coil, causing the coil to push or pull like a piston. The voice coil is cemented to the back of the speaker cone, which creates sound waves as it's pushed back and forth.

Larger diameters tend to be advantageous because low frequencies involve moving a great deal of air. A recent trend has been for high excursion. Excursion is defined as how far the cone can linearly travel from its resting position. Some newer models can move as much as +/-2.5" , yielding an overall controlled displacement of 5" with the voice coil of the driver staying inside the magnetic field.

Subwoofers are usually powered by a high power amplifier, and often an electronic crossover with a Low-pass filter is used to ensure that higher frequencies will not be directed to the subwoofer.

The need to reproduce these frequencies has increased since older formats, such as vinyl records, have been replaced by digital formats, such as CDs, and particularly 5.1 formats such as Dolby Digital and DTS, in which the ".1" channel is dedicated solely to the subwoofer. The .1 channel is usually dedicated to extended bass frequencies, for example, the low frequencies of a gunshot, double bass, or thunder. This track is often used aggressively by mixing artists.


Professional audio

Subwoofers are found in professional applications such as live concerts, movie theatres, various other sound reinforcement applications (ranging from nightclubs to theme restaurants) and studios. Some of these applications require subwoofers designed for very high sound levels, such as the JBL 4645 or the Electro-Voice TL440 which use a larger than average 18" driver and are certified for use in THX movie theatres. Most movie theatre speakers (situated behind a perforated screen) typically use 15" drivers, so the larger subwoofers are used only to reproduce the lowest frequencies at high sound pressure levels.
Large concert sound systems always use subwoofers (referred to as "subs" by the engineers and crew). The bulk of the sound system is usually "flown" (suspended from the ceiling by chain hoists) and the subs are usually stacked on the stage or the ground in front of the stage to the left and right of the performance space.

An unusual example of the use of sub-woofers came with the release of Earthquake in 1974 where they used a system called Sensurround to create a feeling of an earthquake. This was simply a set of large sub-woofers designed to put out infrasound(felt but not heard). Similar systems are used in theme park rides, such as "Days of Thunder," which uses sound to simulate a physical impact.
Many times the subs are not part of the entire sound mix but are specifically fed just kick drum, bass guitar and other low-frequency content from a separate output on the main mixing console. Popular sub systems in use currently are made by companies such as EAW but usually the subs will be made by the manufacturer of the rest of the PA system.

The 18-inch woofer driver is the primary majority device for pro audio applications. They are usually direct radiating in a ported enclosure built of 13-ply birch. For electronic music events with at least a thousand audience members there are often more than 20 double-18-inch cabinets on either side of the stage. 12-inch drivers in very large folded horns are also becoming popular now. One of the most powerful subs manufactured can play as low as 25 Hz and can cover thousands of feet and uses two 12-inch woofers on a 13-foot (4 m) long folded horn.

Pro Audio subs have to be capable of very high output levels – after all, concert venues may seat 10,000s of individuals outdoors. On average, music applications generally require less capability than movie soundtracks in the very lowest octave, but modern popular music is changing this preconception and this is reflected in the design attention given to the subwoofer section of the PA system nowadays compared to a few decades ago. People who are accustomed to bass in home audio systems and car audio many times think that the subs in a concert PA system aren't putting out that much. As sound pressure is measured in decibels which are a logarithmic scale PA subs can be 10 times more powerful yet only measure a 2 more decibels. Also sound intensity obeys the inverse-square law in relation to distance from the sub and at outdoor events the crowd are many meters away from the PA equipment.

There are many challenges in woofer manufacture such as stopping the cone cleanly at each end of the in/out cycle, loudness which requires the cone to move farther in and out, and ringing when the cone is underdamped. There are challenges with maintaining a stable impedance. Woofer design is about effectively converting an low frequency amplifier signal to mechanical air movement with high efficiency.
Resonant frequency is one of a woofer's parameters and is determined by the compliance (flexibility) of the cone suspension, the mass of the cone, the magnetic field strength and the air resistance behind it. The lower the resonant frequency, the lower the frequency of sound that may be produced without distortion. Resonant frequency is listed in the Thiele/Small parameters as Fs.

All woofers have electrical/mechanical properties that dictate the correct box size and crossover components for a given finished loudspeaker. A given woofer may work well in one application and not in another. It is important to know and understand the Thiele/Small parameters in order to build a satisfactory loudspeaker in an enclosure.


Woofer design

There are many challenges in woofer manufacture such as stopping the cone cleanly at each end of the in/out cycle, loudness which requires the cone to move farther in and out, and ringing when the cone is underdamped. There are challenges with maintaining a stable impedance. Woofer design is about effectively converting an low frequency amplifier signal to mechanical air movement with high efficiency.
Resonant frequency is one of a woofer's parameters and is determined by the compliance (flexibility) of the cone suspension, the mass of the cone, the magnetic field strength and the air resistance behind it. The lower the resonant frequency, the lower the frequency of sound that may be produced without distortion. Resonant frequency is listed in the Thiele/Small parameters as Fs.

All woofers have electrical/mechanical properties that dictate the correct box size and crossover components for a given finished loudspeaker. A given woofer may work well in one application and not in another. It is important to know and understand the Thiele/Small parameters in order to build a satisfactory loudspeaker in an enclosure.




Cone materials

All cone materials have advantages and disadvantages. The three properties designers look for in cones are light weight, stiffness and lack of coloration/ringing. Exotic materials like Kevlar and magnesium are light and stiff but can have ringing problems. Materials like paper, coated paper and polymer will ring less but can be heavier and not as stiff.

There are good and bad woofers made with all types of cone materials. However, there is a lot more to driver construction than just cone material.


Power handling

A popular woofer measurement is power handling, an average amount of power the woofer can take. This rating is not well regulated and many woofer manufacturers advertise exaggerated numbers. The only time wattage ratings become important are in very high volume (loudness) situations and very low amplifier power situations. In high volume situations, a woofer's voice coil may overheat and damage the woofer. In low power situations, the amplifier will clip and send a distorted signal to the woofer, damaging the voice coil. For normal listening level applications, this number can be ignored.

Woofers designed for public address (PA) and instrument applications are similar in makeup to home audio woofers. Key design variances are: Cabinets are built for regular shipping and handling, woofer cones are usually larger to allow for higher sound levels, voice coils are more robust to withstand higher voltages. Generally, a home woofer used in a PA/instrument application will fail in short order. A PA/instrument woofer used in a home application will not have as much low volume detail.


Frequency ranges

Humans can hear down to around 20 Hertz. A loudspeaker that can produce bass down to 45 Hertz will sound full range to most people. Many small loudspeakers are designed to produce bass down to around 80-100 Hertz because it is assumed the end user will be using a subwoofer to cover the bottom 2 octaves. But to accurately produce the bottom octaves, a woofer must be large enough to move an appropriate volume of air for a given room. The larger the room, the larger the woofer will have to be to fill the room.
The chart below defines the general operating ranges of different sized woofers. The green area represents the optimal woofer range while the yellow represents the extended range. The purple area represents the music range of almost all instruments. The lighter purple areas extend the instrument range to include rarely played notes, say the first and last 10 keys on the piano. Comparing the instrument versus driver ranges, one can get an idea of the speaker building problem: no woofer does everything well.


Frequency response of woofers

For woofers, the frequency is the number of times the cone of the woofer goes in and out per second and is measured in Hertz. So at 20 Hertz, the cone is going in and out 20 times every second. The faster the cone moves, the higher the pitch. The farther in and out the cone moves in each cycle, the louder it sounds.


Enclosures

A loudspeaker enclosure is a cabinet designed for mounting of loudspeaker drive units. The major role of the enclosure is to prevent the out-of-phase sound waves from the rear of the speaker combining with the positive phase sound waves from the front of the speaker. This would result in interference patterns and cancellation causing the efficiency of the speaker to be compromised, particularly in the low frequencies where the wavelengths are large enough that interference will affect the entire listening area.
Most loudspeaker cabinets currently are box-shaped. Although loudspeaker cabinets may at times appear bulky in an environment, they serve a number of necessary purposes.


History of enclosures

Before the 1950's manufacturers did not fully enclose their loudspeaker cabinets. The back of the cabinet was typically left open. Since the rear of the loudspeakers in a cabinet broadcast soundwaves in 180 degrees out of phase from the front where one listens, mixing the out of phase soundwaves into the listening environment causes loss of bass and lower program volumes.

Research from the 1930's by Dr. H.F. Olsen showed experiments that curved loudspeaker cabinets have major benefits by eliminating most soundwave diffraction.

In the 1990's, US Enclosure Company (Ultimate Loudspeaker Enclosure Company) pioneered a method of molding loudspeaker walls into any curved shape using wall materials that were equal to wood loudspeaker walls.


Explanation

The ideal mounting for a loudspeaker would be a flat board of infinite size with infinite space behind it because the rear soundwaves cannot cancel the soundwaves from the front. An 'open baffle' loudspeaker is an approximation to this as the transducer is mounted on a simple board of size comparable to the lowest wavelength to be reproduced. However, for many purposes this is impractical and the enclosures must use other techniques to maximize the output of the loudspeaker.

Loudspeaker cabinets sometimes are "sealed" and sometimes "ported". Ported cabinets allow some of the sound energy inside the cabinet to be released generally to increase bass response. Many other engineering designs exist including acoustic transmission lines.

Enclosures always play a significant role in the sound production, adding resonances, diffraction, and other unwanted effects. Problems with resonance are usually reduced by increasing enclosure rigidity, added internal damping and increasing the mass of the walls of the enclosure. The speaker manufacturer Wharfedale has addressed the problem of cabinet resonance by using two layers of wood with the space between filled with sand. Home experimenters have designed speakers built from concrete sewer pipes for similar reasons.

Diffraction problems are addressed in the shape of the enclosure; avoiding sharp corners on the front of the enclosure for instance. Sometimes the differences in phase response of the different size drivers is addressed by setting the smaller drivers further back, by leaning or stepping the front baffle, so that the resulting wavefront from all drivers is coherent when it reaches the listener. The Acoustic Center of the driver, or physical position of each driver's voice coil, dictates the amount of rearward offset to time-align the drivers.

Ideal loudspeaker cabinets use heavy walls to contain the out of phase sound energy, and the walls are usually made of wood.


'Woofer' and 'subwoofer' enclosures

Enclosures used for woofer and subwoofer are applications that can be adequately modelled in the low frequency range (approximately 100–200 Hz and below) using acoustics and the lumped component model. For the purposes of this type of analysis, each enclosure has a loudspeaker topology. The most common enclosure types are listed below.

Closed-box enclosures
Infinite baffle

Closed box enclosure

A variation on the 'open baffle' is to place the loudspeaker in a very large sealed box. The loudspeaker driver's mass and compliance, i.e. the stiffness of the suspension of the cone, determines the resonant frequency and damping properties of the system, which affect the low-frequency response of the speaker; the response falls off very sharply below the cabinet resonant frequency (Fs), which can be determined by finding the peak impedance. The designer trades off bass response for flatness; the larger the resonant peak in the bass, the lower the speaker will seem to reproduce, but the more over-emphasized the resonant frequency will be. The box must be large enough that the internal pressure caused when the driver cone moves backwards into the cabinet does not rise high enough to affect this. The box is usually filled loosely with foam, pillow stuffing, fiberglass, or other wadding, converting the speaker's thermodynamic properties from adiabatic to isothermal.

Acoustic suspension
The closed-box or 'acoustic suspension' enclosure, rather than using a large box to avoid the effect of the internal air pressure, uses a smaller, tightly sealed box. The box is typically designed with a very small rate of leakage so that internal and external pressures can slowly equalise over time, allowing the speaker to adjust to changes in barometric pressure or altitude. In this case, the true suspension of the driver's cone is the air trapped inside the box which acts as a spring with very close to ideal behavior rather than the mechanical suspension of the speaker driver, which for this application must be very weak, just strong enough to keep the cone centered in the absence of any signal. The drawback of these speakers is their low efficiency, due to the loss of the power absorbed inside the cabinet.

Reflex enclosures
Bass-reflex

Bass reflex enclosure

Other types of enclosures attempt to improve the low frequency response or overall efficiency of the loudspeaker by using various combinations of reflex ports or passive radiating elements to transmit the energy from the rear of the speaker to the listener; these enclosures may also be referred to as vented/ported enclosures, bass reflex, transmission lines (see below). The interiors of such enclosures are also often lined with fiberglass matting for absorption. Reflex ports are tuned by amount of mass within the vent, using appropriate diameter and length to reach this point. This enclosure is the most common as it lends itself to small size and reasonable bass.

Compound or Band-pass

Compound or 4th order band-pass enclosure.

A 4th order bandpass is really just the same as a vented box where the contribution from the driver is trapped in a sealed box which modifies the resonance of the driver. In its simplest form it has two chambers. The dividing wall between the chambers has the driver mounted on it and the panel opposite to it (or the chamber into which the driver faces) has a port. If the enclosure on each side of the woofer has a port in it then the enclosure yields a 6th order band-pass response. This enclosure is considerably harder to design and tends to be driver-specific.

Passive radiator

Passive radiator enclosure

Sometimes a passive radiator (PR) or drone, similar to a speaker driver but without an electrically activated voice coil, is used instead of a reflex port. Passive radiators are used primarily to tune small volumes to low frequencies, where a port would need to be very long. They are also used to eliminate port turbulence and reduce power compression caused by high velocity airflow in ports. Passive radiators are tuned by their mass (Mmp) and the way their compliance interacts with the compliance of the air in the box. Passive radiators add a complication to vented systems which causes a notch in frequency response at the PR's free air resonant frequency and this causes a steeper rolloff below the drone's tuning frequency Fb and poorer transient response than standard vented loudspeakers. Due to the lack of vent turbulence and vent pipe resonances, many prefer the sound of PR's to reflex ports. PR's do add considerable cost to the system, however.


Other enclosure types

Transmission line
The transmission line system is a waveguide system in which the guide shifts the phase of the driver's rear output by at least 90°, thereby reinforcing the frequencies near the driver's Fs. Transmission lines tend to be larger than the other systems, due to the size and length of the line required by the design (1/4th of a wavelength). In 2004 there was a breakthrough in this design category. The coupler transmission line enclosure was designed in part by veteran audio design engineers Brad Judah and Andy Bartha of Nucore Electromagnetics. The new coupler transmission line speaker enclosure design features a non-resonant characteristic and flat frequency response from 14 Hz to 25 kHz combined with high efficiency.

Dipole

Dipole speakers and their radiation pattern.

A dipole enclosure in its simplest form is a driver located on a flat baffle. The baffle may be folded in order to conserve space. A rectangular cross-section is more common than a circular one since it is much easier to fabricate in folded form than a circular cross-section. The baffle dimensions are chosen to get the desired response, with larger dimensions giving a lower frequency before the front and rear waves combine and cancel. A dipole enclosure has a "figure-of-eight" radiation pattern, which means that there is a reduction in sound pressure or loudness at the sides as compared to the front and rear. This is very useful when it is desired to prevent the sound from being heard at places other than the listening room/venue.

Sun 28 May 06 @ 4:53 pm

The Definition of Turntablism

The term "Turntablism" was first coined in 1995 by DJ Babu (Beat Junkies) to describe a form of advanced turntable music stemming from Hip-Hop DJ'ing. As Babu stated in a brief 1996 interview :
"My definition of a Turntablist is a person who uses the turntables not to play music, but to manipulate sound and create music." (Interview with Christo Macias, Palo Alto, CA, May 1996.)
The International Turntablism Federation (ITF), a collective of the World's best Turntablists offers a similar description:
"Turntablist: One who uses the phonograph turntable as a component to make music as well as an instrument to literally play music."

(Official ITF Newsletter, 1996.)


The Elements of Turntablism

SCRATCHING:
In turntablism, there are currently two main styles of musical expression. The first and foremost is scratching (also known as cutting). Scratching is a technique by which the performer uses vinyl and moves the record back and forth against the needle to produce sounds of varying degree. The performer can push the record forward or backward (denoted as forward stroke and backward strokes). In some cases, creating a stroke isn't necessary. Simply playing the record for a short duration is sufficient.

BEAT JUGGLING:
The second common style of turntablism is beat-juggling. This technique is performed by using two records and manipulating the arrangement of the elements (drum sounds, headnotes, etc.) from both to create a new rhythmical composition


Turntablists Are the True Visionaries

At the annual conference of the National Campus/Community Radio Association in Halifax this past June, a motion was passed commiting the organizaton to establishing turntablism as a unique musical genre recognized by the Canadian Radio-television and Telecommunications Commission, and a form of Canadian content. Reporting on a workshop about "urban" music programming at the conference, Meril Rasmussen wrote: "It is obvious that the way some artists work a turntable and other sound equipment that it is a tool -- an instrument, if you will."

The desire for recognition of this artform pre-dates its current relative popularity; Zorlon Vode, writing for turntablism.com, says, "[Reknown avant garde composer] John Cage presented his ideas to the Seattle Arts Society in 1937 with the hope of getting some type of recognition for the turntable."

DJs who manipulate pre-recorded sounds using turntables to a degree that the original material is obscured or unrecognizable are creating new works, according to practitioners and supporters. The term "turntablism" was first used in 1995 by DJ Babu of the Beat Junkies, who states, "My defnition of a Turntablist is a person who uses the turntables not to play music, but to manipulate sound and create music." (interview with Christo Macias, quoted on turntablism.com) Turntablists who perform on the radio are the producers, artists and composers of the newly-created music, qualifying the material as Canadian content in accordance with the CRTC's Radio Regulations MAPL System Fact Sheet, argues the NCRA. Gaining Can-con status for turntablists will facilitate "urban" music programming on campus radio stations, which need to meet a minimum percentage of Canadian material.

The definition of turntablism, as stated in the NCRA's submisison to the CRTC's review of campus radio policy, is: "manipulation of previously recorded track(s) to the extent that they are substantially altered from their original format, and that the continuous or consistent alteration of the previously existing tracks(s) continues for one minute or more." This can be done by, "scratching back and forth on the turntables or beat mixing in another selection within 30 seconds of the original piece," says hip hop promotor Alok Sharma (DJ Kola).

The CRTC-commissioned Music Availability Study describes what turntablism is NOT: "Whereas turntablism modifies existing vinyl recordings sufficiently to be able to speak of the creation of new works of music, DJ mixing is the presentation of existing recorded material in a creative and musically sensitive way. Contrary to turntablists, DJs who practice DJ mixing cannot be said to be composing music or performing material of their own composition".

France is now, reportedly, the first country in the world to have succeeded in having turntablism recognized officially by their communications and broadcasting governmental body. Doing the same here "would revolutionize the way the CRTC and the Canadian public view turntablism," says DJ Kola.
At the 1998 conference of the National Campus/Community Radio Association, in Victoria, The Turnstylez performed for representatives of the CRTC. They demonstrated their skills manipulating sounds with turntables to re-create "traditional" instruments.

In 1998, The Turnstylez won the ITF (International Turntable Federation) North American "Team" category in New York City and continued to the World Finals in Amsterdam. The ITF is one of two major competitions which help to legitimize and advance turntablism worldwide. The 1998 DMC (Disco Mix club) World Champion is A-Trak, a high school student from Montreal. Mark Miller, writing in the Globe and Mail, reviewed an A-Trak performance at the 1999 Montreal Jazz Festival. "If A-Trak and Kid Koala were pianists, there would be every reason to paise their soft hands and nimble fingers, their imagination and their rhythmic conception. The fact that they each manipulate two turntables, rather than a keyboard and three pedals, shouldn't really change the assessment, should it?"

CKDU programmer Jesus Murphy concludes, "Recognizing turntablism as Cancon will not only help tremendously in filling our requirements, it will also lend legitimacy to an artform that has struggled for recognition and respect for two decades now. This will be a victory for radio and for hip-hop as a whole."


Wheels of Steel

I suppose it was just a matter of time before someone added the dreaded "ism" suffix after the word "turntable". After all, members of the music community’s elite have been routinely dismissing the skills of turntable artists for years now, so maybe the respectable ending will lend it a bit of credence in those circles. "Are they actually creating music," they snort. ""If they are simply playing other people’s records?"

Good question. Were the Rolling Stones actually creating new music by ripping off old howling wolf songs? Is Puff Daddy making any headway by reworking virtually every top 10 hit from the 80’s? I guess it depends on your perspective.

"The turntable can adapt or mimic the violin, the drum, the guitar, the bass, and any other type of instrument. What you are using is records and records contain all these different instruments. The turntable can almost morph into any instrument" - DJ Ro Swift
Despite the fact that the turntable has been in common public use since the 1920’s, its acceptance as an actual instrument instead of merely a playback device has been fairly recent. as early as 1939, American avant garde composer John Cage was writing compositions that included turntables bearing test-tone records being manipulated, but it wasn’t until the late 70’s, as the influx of Jamaican culture began permeating the boroughs of New York that the concept of the disc jockey as artist began to even be considered.

All it really took was someone with the bright idea of plugging a set of speakers, a mixer, and a couple of turntables into a Bronx lamp pole to create the instant street party. The man was Kool Herc, a Kingston immigrant to the United States and although he was probably just trying to infuse a bit of the culture he grew up with in his new home, he set the standard for hip hop culture that continues and thrives some two decades later.

While most of the innovations in turntablism have occurred within the hip hop community, one cannot ignore the contributions of avant gardists like Otomo Yoshihido or Christian Marclay... the latter of which took the concept of "cutting and scratching" tthe (il)logical extreme by actually sawing several records to pieces and then gluing the bits together to form entirely new pieces of music.

In the mid to late nineties, innovations in turntablism are most commonly found in DJ crews like New York’s X-Ecutioners, the Bay Area’s invisible Skratch Piklz or Toronto’s Turnstylez. These groups (almost always trios or quartets) create entire scores o f music beats, rhythms, melodies, and often three or four part harmonies completely from the sounds of vinyl being manipulated, torn apart , and reorganized.
In 1999 the Canadian Radio-Television and Telecommunications Commission, as a part of its proposed new policy for campus radio, is looking at turntablism as a distinct musical genre. The implication is that any DJ on campus/community radio in Canada who creates brand new music by significantly altering existing vinyl sources may be considered Canadian Content. this is a welcome step and puts the CRTC at the forefront of recognizing new music.

Russell Gragg !earshot April 1999


The Turning Tables of Life in the Poor City

by Meril Rasmussen

I went to Mopa Dean's and Alok Sharma's workshop on Urban Music but I didn't make it to the following discussion of Turntablism, although I talked to people who did. The discussion of Urban Music was informative and demystifying. Urban music is a catch-all phrase used by the industry to include hip hop, R&B and - especially in Canada - soca, reggae, calypso...Alok suggests that the word has classist, racist undertones but Mopa pointed out that music pundit Chuck D has no problem with the term.
With the possible exception of Virgin, Urban music is under-acknowledged by the big record companies even though it accounts for a hefty percentage of sales. There were various ideas about the reasons for this. Much Music seems to be another ally to Canadian Urban.

Canadian history seems tied in with Maestro Fresh Wes who self-produced and self-videoed his way to stardom a decade ago but didn't succeed then in busting open the Canadian market. The industry at that point was confused about how to market Canadian Urban Music and when Maestro's third album flopped, the door closed behind him. Now the Underground has developed in Maestro's vein. Once artists have proved themselves in the DIY Underground they might get picked up. It looks like Cho Clair is now once again poised to open up doors fro Canadian artists with a much stronger foundation in place.

As for Turntabilism: It is obvious that the way some artists work a turntable and other sound equipment that it is a tool - an instrument, if you will. It is only a matter of time before this contribution is recognized and considereed CanCon and the copyright issue sounds like a nightmare. hopefully the CRTC is making open ended policy and the powers that be will not try to impeded or under appreciate these contributions. That's the best I can do but talk to Mopa or Alok. They know.



The History of Turntablism


Grandmaster Flash on the "Wheels of Steel."


The history of turntablism spans nearly 65 years. Like any other instrument, the turntable went through many stages and variations as an instrument.

Two musicians constitute the prehistory of turntable music. As early as 1937 American avante-garde composer John Cage had ideas for using records to create music, which he describes in his book, "The Future of Music: Credo." By 1948 a French avante-garde composer, Pierre Schaeffer, used multiple turntables to cue desired sounds from existing recordings and altered the music through changing the speed, adjusting the volume and playing the sounds backwards.

Nothing revolutionary occurred in the history of turntablism until 1973 when Bronx DJs like Afrika Bambaataa, Grandmaster Flash and Kool Herc showed what could be done with turntables. Kool Herc invented what is called "breakbeats," which is when a DJ fuses together different parts of songs into a musical collage. Grandmaster Flash is known as the DJ who expanded the most upon Kool Herc's breakbeats discovery to invent more turntablist techniques.

In 1977 an entertaining piece of turntablism history took place when Grand Wizard Theodore invented the "scratch." 14-year-old Theodore was messing around with his record player when his mom walked in his room. As she walked in, Grand Wizard stopped the record with his fingers to listen to her while unconsciously moving the record back and forth over the same drumbeat, creating a scratch. The "scratch" has been a major component of turntablism ever since.

From 1979 through the early 80s, DJs drifted out of the spotlight as commercial hip-hop musicians chose to use live funk bands on their recordings instead of DJs. Turntablism reached the mainstream in 1983 when jazz musician Herbie Hancock performed the song, "Rockit," with Grandmaster D.S.T. Kids across the world were captivated by Grandmaster D.S.T. as he scratched a record to the beat of the song.
The next step in the evolution of turntablism took place throughout the 1980s as turntablists showed off their skills at DJ battles. At these battles, extremely competitive DJs showed how fast and creative they could DJ while maintaining the rhythm.

DJ Cynsere, a Lawrence turntablist, said in the 1980s DJs were the spotlight of hip-hop and MCs (rappers) were in the background. He said nowadays this has unfortunately reversed to the point that the DJs are on the backburner while the rappers reach fame. Cynsere said DJs have been able to get a little more exposure over the last few years.

In the late 1980s popular culture lost interest in turntablism, and it went back underground where it continued to thrive with its truly devoted fans and musicians.

Throughout the 1990s turntablists continued to expand the vocabulary of the instrument by inventing new techniques. According to Kembrew McLeod, a turntablism historian, the amount of full-length turntablist albums by individual artists has grown exponentially since 1996. Presently turntablists like DJ Q-bert, Rob Swift, DJ Shadow, Mix Master Mike, DJ Spooky and Cut Chemist experiment with their turntables and demonstrate the dynamic range of the unique instrument.

Turntablism's history continues to sculpt itself. Without a doubt, turntablism has much room to expand into the future, and it will be interesting to see how it continues to evolve.


Mixmaster Mike- photo by Chris Taylor

Musical revolutions are often the result of the most mundane circumstances.
Sometime in the late-'70s at a housing project in the Bronx, a teen-ager was in his room playing records a bit too loudly. As most parents are wont to do, his mom started banging on his door, telling him to turn his music down. When she walked in, he stopped the record with his fingers, listening partially to what she was telling him while unconsciously moving the record back and forth over the same drumbeat.
A few years down the line, that teenager morphed into Grand Wizard Theodore.

"I wanted to get that same groove I was on," the veteran DJ explained to director John Carluccio in his film, "Battle Sounds," a soon-to-be released documentary on turntablists that was recently screened at the Whitney Museum in New York. "So I was, like, back and forth and I said to myself, 'Hey, this sounds pretty good!' Ya know?"

"So I started practicing it and it became a scratch," he said.
Whether this story is a fanciful bit of myth-making or straight-up fact, it is nonetheless a good illustration of turntablism's haphazard evolution -- a series of events built around mistakes that sounded good and so were further developed.

When hip-hop emerged in the Bronx during the early-to-mid-'70s, DJs were central figures in its evolution. The DJs who inspired Grand Wizard Theodore - - Kool Herc, Afrika Bambaataa and Grandmaster Flash -- often plugged their massive sound systems into street-lamp outlets in local parks, dug deep into their crates full of records and, quite literally, kept the party rocking till the cops came a-knocking.
At these house and block parties, it was the DJ's job to spin popular records that kept the party alive and people dancing. It was this push to keep everybody dancing that led to the development of techniques that became staples of turntablism.

The most popular of these early Bronx DJs was Jamaican immigrant Kool Herc, whose name is certainly not as recognizable as Bambaataa's and Flash's, but who is credited with one innovation that laid an essential blueprint for hip-hop. With two turntables, Herc fused together the chunks of songs that were the most popular with dancers, segueing them into one long musical collage.

These fragments were composed of the percussion breaks within the songs and came to be known as "breakbeats," the percussive foundation for virtually all hip-hop songs (some of the better known breaks are the Incredible Bongo Band's "Apache," James Brown's "Give It Up or Turn It Loose" and even Aerosmith's "Walk This Way").

Other DJs took this concept and began expanding on the possibilities that two turntables could offer. One of the first DJs to pick up on the breakbeat technique was Grandmaster Flash, who went further than Kool Herc in his turntable wizardry.

With a couple of turntables Flash was able to take two copies of the same record and keep the beat going by playing a short percussion break on one turntable while he alternately moved the needle back to the beginning of that break on the other, a technique called "needle dropping." Flash also claims that he first developed "scratching," as well as "punch phasing," in which he added percussive bursts of noise from one turntable on top of what was playing on the other turntable.

Herc, Flash and Bambaataa inspired a number of up-and-coming Bronx DJs during the late-'70s, including Grand Wizard Theodore, Grandmaster D.S.T., DJ Afrika Islam, DJ Charlie Chase and DJ Breakout. In the early days of hip-hop, it was the DJ who was the focus of a hip-hop crew, and many group names reflected that (Grandmaster Flash and the Furious Five, Grand Wizard Theodore and the Fantastic Five, etc.).

But one event changed hip-hop and altered the DJ's role for years to come: the release of the Sugarhill Gang's "Rapper's Delight" in 1979, which defined the sound of commercial hip-hop during the early-'80s. This song, and virtually all recorded raps that immediately followed, used a live funk band as the instrumental track, eliminating the role of the DJ altogether. Even subsequent records by Grandmaster Flash ("The Birthday Party") and Grand Wizard Theodore ("Can I Get a Soul Clap") contained little or no actual turntablism.

A few documents of turntablism from this period survived, most notably Afrika Bambaataa's "Death Mix" (essentially a bootleg of a live Bambaataa turntable jam) and Grandmaster Flash's "Adventures of Grandmaster Flash on the Wheels of Steel."

Released in 1981, this seven-minute song captures Grandmaster Flash cutting it up on the turntable with songs by Blondie, Chic, Queen, Spoonie Gee, Sequence, the Sugarhill Gang and his own Furious Five. While it sounds relatively primitive compared to the lightning-fast dexterity of Mix Master Mike or the multilayered density of Cut Chemist, "Adventures of Grandmaster Flash on the Wheels of Steel" demonstrated the turntablist technique for record- buyers and aspiring DJs outside New York.
Other records -- such as Davy DMX's "One For the Treble," released in 1984; and Grandmaster D.S.T.'s "Megamix," released in 1983 -- were instrumental jams popular with breakdancers that showed off those artists' turntablist skills.

But it was the Grammy-winning Herbie Hancock/Grandmaster D.S.T. collaboration, "Rockit," that brought turntablism to a much wider audience in 1983, with its orchestrated turntable scratches cut up over an electro beat.

"That record changed everything for me," said Q-bert, co-founder of the San Francisco turntablist crew Invisibl Skratch Piklz, which also includes current Beastie Boy DJ Mix Master Mike, along with Shortcut and D-Styles. "That song is still hella fresh. It's one of the things that inspired me to be a turntablist."
As far back as 1985, the seeds were planted for the future renaissance of turntablism in the mid- to late-'90s. In San Francisco, Q-bert, Mix Master Mike, DJ Apollo and other DJs battled each other. Said Mike: "DJ Apollo and I created the first orchestrated scratch band with our 'Peter Piper' routine back in 1985."

"Peter Piper," by RUN-DMC, is important in its own right because it was one of the first recorded hip-hop songs to heavily feature the crackles and pops of vinyl as part of the song, thus focusing the spotlight on the turntable as an instrument.

Throughout the second half of the '80s, full-length hip-hop albums often contained "DJ tracks," instrumental songs that showcased the DJ's skills, such as Public Enemy's "Terminator X Speaks With His Hands" and "Terminator X To the Edge of Panic"; Gang Starr's "DJ Premier in Deep Concentration"; and Cash Money & Marvelous' "The Music Maker."

Occasionally a DJ-produced single such as Mark the 45 King's "The 900 Number" and Original Concept's "Can You Feel It?" broke through. But, for the most part, the DJ was increasingly pushed further into the background.

"Rappers stopped having DJs in their shows and on their albums, and labels didn't want to clear samples from DJ tracks for tracks on rap albums," said Dave Paul, the owner of Bomb Hip-hop, the label that has put out the acclaimed Return of the DJ series. But DJing and turntablism didn't die, Paul said. It just went underground and thrived, especially in places like San Francisco.

Turntablism flourished despite the lack of commercial support and artists were driven by good-natured competition and a desire to advance the art. DJ battles, like Clark Kent's DJ Battle for World Supremacy, continued to grow during the '80s and into the '90s. And in 1987, the DMC (Disco Mix Club) held the first in an annual series of popular DJ competitions.

The DMC is also significant for introducing the world to what would later become the Invisibl Skratch Piklz, known in 1992 as Rocksteady DJs (comprised of Q-bert, Mike and Apollo). Many other turntablist crews followed the Piklz path, including BulletProof Scratch Hamsters, The Fifth Platoon, The Beat Junkies, Space Travelers, STA, The 1200 Hobos and the X-Men (now the X-ecutioners for copyright reasons).
The DMC, the largest competition of its kind, has since spawned a successful rival, the International Turntablist Federation (ITF), which held its first world championship battle in 1996.

For various reasons -- including hip-hop's historical pattern of sexism -- female DJs are rarer than female MCs. Baby D of the Bronx-based, late-'70s, all- female hip-hop crew Mercedes Ladies (which included Sha-Rock, later of the Funky Four + One), was one of the first female DJs.

Other female DJs -- such as Kim B, MK, Spinderella, Coco Chanel and Jazzy Joyce -- followed Baby D's lead, but few female DJs of the '80s got much respect or were allowed entry into the boys' club of DJ battles, such as the DMC or ITF.

There are exceptions, and in the late-'90s more female turntablists are making waves, such as DJ Snowhite and Kuttin Kandi, who is a member of both the 5th Platoon and the all-female crew, Anomalies. Kuttin Kandi is one of the few female turntablists who has entered and placed in both the ITF and DMC competitions.

In 1993, DJ Shadow released a single, "In/Flux," that would slowly reverberate throughout the music scene and, in retrospect, can be seen as a harbinger of the turntablist renaissance of the late-'90s. "In/Flux" was unique in a number of ways. It's subtle sonic layers were drawn from Shadow's record collection, and, most importantly, the album was all-instrumental.

Last year, numerous turntablist compilation records were released, and many more have been released this year. Among them are Axiom's Altered Beats, Bomb Hip-hop's Return of the DJ volumes, Bill Laswell's Vallis compilations, OM's Deep Concentration series, and Wordsound's Subterranean Hitz series.
Also, the number of full-length turntablist albums by individual artists has grown exponentially since 1996. In addition to a handful of albums by DJ Spooky, DJ Cam, DJ Krush and Rob Swift in 1996 and 1997, there have been a number of these releases in 1998, including the X-ecutioners' X-pressions, DJ Faust's Man or Myth?, Mix Master Mike's Anti-Theft Device and DJ Spooky's second album, Riddim Warfare, among others. With sales of the Techniques 1200 (the turntable of choice for DJs) skyrocketing, it appears that the momentum that turntablism has gained throughout the decade will not subside but, rather, increase exponentially.


The Hip-Hop DJ

by DXT formerly known as Grandmixer D.ST



Like the Jedi in Star Wars fighting against evil, enduring strenuous training, accepting a life-long commitment to obtaining the knowledge of the universe and being heard but never seen, the hip-hop DJ has very much the same destiny. The hip-hop DJ has to endure the process of obtaining a vast knowledge of music and rhythm (the force), be able to synchronize the grooves and beats, and continually search for new sounds to maintain his status in the culture. Much as the Jedi is rumored to be the ultimate warrior of the universe, the hip-hop DJ has become just that, a rumor. Nevertheless, the DJ will always play a major role in hip-hop culture despite its ever-changing nature.

In Star Wars, becoming a Jedi meant that a warrior had to feel the "force," know it and always recognize it. The hip-hop DJ has to do the same. As a DJ, a person has to feel the rhythms and identify them as being a natural part of their existence. Either the force is with you or it isn't. Feeling rhythm is a skill that cannot be taught. This was a sign of a true beat hunter - someone who could instantly feel the rhythms. DJs listened to all genres of music from rock, Latin, country, opera … whatever, but their main inspiration came from funk and R&B.



Funk/R&B music is the closest source of music that resembles the original drum sounds from Africa. No matter what, funk always moved a crowd. Somehow, after 400 years of displacement out of Africa, the true hip-hop DJ can still feel the rhythm of the drums of Africa. Once you've established a vast music collection, now you have to know how to work it! Not only does a DJ have to know the music on the record; a DJ must also know exactly where the rhythm is on the record. Developing DJ skills requires hours of practice and listening. Techniques such as needle dropping, cueing records, backspinning, scratching and the like are skills that have evolved out of pure hard work and creativity. Developing your own style is key in making your mark in the culture.



The Tri-Force Kool DJ Herc had a style of playing oldies but goodies and only playing the dopest part of the records. He also traveled with a massive sound system that was impressive in its own right. Grandmaster Flash was a technician about his work. He went against the rules of the disco DJs and left behind smooth mixes. He went straight to the cut. However, despite the equipment and technique, a DJ has to be in total harmony with the rhythms. That means being at one with the force (the rhythms of Africa), and the one who understood that overall was Afrika Bambaataa. He played rhythms that would penetrate your soul and make you move. In the old African tribes he would have been known as the medicine man!

Other DJs during that time (early 1970s) were Kool DJ Dee, DJ Smoke and the Smokatrons, Mean Jean, Disco King Mario (Chuck Chuck City), Pete DJ Jones, Grand Master Flowers and DJ Hollywood, just to name a few. However, Herc, Flash and Bambaataa had the most profound influence on the development of hip-hop culture. These three men represent the Tri-Force of the hip-hop DJ: Kool DJ Herc (presence), Afrika Bambaataa (energy) and Grandmaster Flash (technique). Their examples inspired young teenagers from all over the Bronx to become hip-hop DJs.

Out of the hundreds of DJs spawned from the spirit of the Tri-Force, sweeping through the parks and clubs of the Bronx, only a few stood out, for they had truly harnessed the power: DJ Jazzy Jay, DJ Charlie Chase, Tony Tone, DJ Lil Quick, Imperial Jay Cee, Whiz Kid, DJ Breakout & DJ Baron, DJ Tyrone, Grand Wizard Theodore (inventor of record scratching), DJ Africa Islam (the son of Bambaattaa) and Grandmixer D.ST. whose turntable skills mutated the turntable into a musical instrument. These young men along with Herc, Bam, Flash (the Tri-Force) and their MCs are the Jedi Knights of hip-hop culture. From them you have all the DJs and MCs you see and hear today. In fact, hip-hop culture has disseminated the force from the ghettos of the Bronx, New York to almost every culture in the world.

Looking for the Perfect Beat

The hip-hop DJ's original mission overall was rocking the house, and to do this he or she needed an arsenal of beats (records). The DJ's ability to keep a dance floor packed relied on his selection of records. Not only did he have to have radio favorites, he also had to keep up with the latest beats the other DJs had. In addition, he had to have his own collection of obscure beats and this wasn't an easy task. It was only a matter of time before the other DJs would find out the names of your beats. So, to keep your uniqueness, you had to constantly search for new beats. Thus begun, "The Quest for Beats!"

Other than the development of the MCs, the "quest" was one of the most important events in hip-hop culture because of the demands of maintaining the codes of discipline. First, you had to develop a vast understanding of music - this required much research. You had to listen to all forms of music, no album or album cover was too serious or silly. Nothing was excluded. If it was on vinyl, it had potential. So the more you researched, the more your knowledge of music grew along with your record collection. Second, always travel alone - and if you were with someone, they had to be part of your crew. Any rare recording found was declared top secret and no one outside of your crew could know its name.

Everyday, DJs would head out into the streets of New York to find beats. They would look for thrift shops with large collections of used records. The major record stores were next, to find the latest radio hits. However, the best stores were the small mom and pop record shops throughout the five boroughs of the city. Unlike the bigger commercial stores, the mom and pop record shops would have the old and the new. There wasn't any place that the hip-hop DJ wouldn't dig for beats. It could be mom's, dad's, aunt's, uncle's, cousin's, neighbor's or friend's. No one's record collection was excluded. If there were mountains with caves full of vinyl, you would find a DJ mining for hip-hop gold.

Once you collected enough beats, sometimes just hours before your next party, you had to remove any part of the record label that revealed the artist or the name of the song. Then, you had to subconsciously find where the new beats would fit in your set. Next was practicing - the new beats had to be played in a way that wouldn't give away the artist. If it was just a drumbeat, it was hard for other DJs to know who made the record. So cutting the beat before the other instruments or singers came in was critical. This meant that you had to be fast and precise, and the fastest way to go from one part of a song to another is needle dropping (placing the needle in the same groove at will). This was the ultimate hip-hop DJ skill and was truly mastered by only a few.



The next best thing was Grandmaster Flash's "Clock Theory" which later became known as backspinning. This technique proved to be very useful and allowed DJs to create more new tricks. However, there is a down side to this technique: the more you backspin the more you destroy that part of the record, and some records are too rare to be used like that. As time went on, hip-hop DJs began to incorporate other instruments (for example, Flash's beat box and D.ST.'s synthesizer) into their sets. Always finding something new to mesmerize the crowd. This competition was key to the growth of hip-hop culture, as each DJ's skills increased, the threshold of hip-hop perfection was raised.

The Empire Strikes Back

The hip-hop DJ now had power throughout the city. People would come from miles around just to see Bronx DJs git down. More people became interested in the culture, because they recognized the true spirit in the expression of hip-hop and its magnetizing effect on people. Some hated it because of its universal potential and some only saw one thing: MONEY. Unfortunately, they all played a part in the decay of the culture and the DJ's transition into the shadows.

First, the DJs themselves made a critical mistake. They allowed non-DJs to learn the names of songs that were secret. People from outside hip-hop culture would come to parties to meet DJs so they could discuss records. Sometimes DJs would need new copies of some of their rare beats, and these men would provide them. In return they would ask for the name of a beat that you played. At the same time they'd offer the name of a beat that they got from your competition that you did not have. By doing this, the hip-hop DJ was breaking his own code of secrecy, unaware that their sacred collection of records (their energy) was being consolidated into what we now know as Super Disco Breaks and Break Beat records. So now without the knowledge that could only be acquired through research and hard work, anyone who wanted to be a DJ had access to the sacred beats. This caused a great disturbance in the Tri-Force, and was the beginning of the hip-hop DJ's transition to obscurity.

Second, record companies began signing hip-hop groups with no true interest in the culture to record deals. The DJ and his MC were the two components of one unit, each complimenting the other. Their presence on stage would create energy levels that would leave crowds in awe. However, this was not important to record executives; they only cared about record sales and the MCs were all they needed to sell records. Record companies began to push the MCs into the spotlight, pulling them away from their DJs (the foundation of hip-hop) and pushing the DJs further out of the picture.

Third, the MC now had his own power, but this power was false because he received it from record company executives through their perversion of hip-hop culture and not from the Tri-Force. (And this is still the problem today.) In this perversion, the MC could easily be programmed to think that he or she was still representing hip-hop, even if he or she replaced a DJ with a DAT tape.

Return of the Jedi (The DJ)

We now see a new genre of music: A distorted by-product of true hip-hop culture called rap music (really rap-u-sic) where the MC has been transformed into something called a "rapper." Where money is energy, jewelry and expensive cars are his presence and he possesses no technique at all. For in his blindness he has been used to destroy everything hip-hop culture stands for. Within this madness, the DJ, who has become nothing more than a sidekick to the rapper, continues to struggle, doing everything he can to bring hip-hop from the underground to the service where it belongs.

The hip-hop DJ now spends more time with samplers, computers, synthesizers and drum machines than with turntables. Now some DJs just call themselves producers and the rap artist depends on them to make up beats with the new technology. So it seems that everything happens for a reason, because now that sampling is the main process of rap music, the producer has to find new sounds to sample. He must educate himself like the original hip-hop DJs did because the only way to compete is to practice hard and research (the new "Quest for Beats"). In this quest/search you will find hip-hop culture; it's there, it's always been there, and it will always be here. For it is truly the cosmic rhythm of the universe and its beacon on this planet is … AFRICA THE ORIGIN OF ALL HUMANITY.

This article was commissioned for the Rock-N-Roll Hall of Fame and appeared on their website in 1999: rockhall.com


DJ QBERT


1850's - The phonoautograph is developed by French Researchers. The device records sound waves on a rotating cylinder
1870's - Thomas Edison begins to develop a tinfoil phonograph or speaking machine. The machine included a cardboard cylinder wrapped in tinfoil on a threaded axle. A mouthpiece and diaphragm were connected to a stylus that embossed the sound waves on the tinfoil. To play back the recording, a reproducer replaced the mouthpiece. To test the invention for the first time, Edison recited "Mary Had a Little Lamb" into the mouthpiece.
1876 - Elisha Gray invents the Musical Telegraph. Alexander Graham Bell beats him to the patent office and patents the technology, calling it a graphophone.
1877 - Edison unveils the first hand-cranked phonograph.
1878 - Edison patents the phonograph and intends it to be an office machine.
1887 - Bell's graphophone used wax cylinders and included a floating stylus for clearer sound. Edison improves the phonograph by using a solid wax cylinder and a battery-driven motor to replace the original hand crank.
1890 - Musicians begin recording their music. The cylinders of the phonograph had the ability to record 2-4 minutes of audio. Around 1890, musicians began to record their sessions by setting up several phonographs to record at the same time.
1892 - Flat recording discs are invented; the first of which is called the gramophone disc.
1895 - Edison begins mass production of the phonograph and continues to improve the original design by adding a large horn to amplify the sound.
1901 - The Victor Talking Machine Company of New Jersey is incorporated, and the first Victor gramophones is introduced.
1906 - A new Victor gramaphone was introduced, which featured a concealed (inside) horn. It was dubbed the Victrola.
1919 - Invention of the Theremin, by Leon Theremin (Lev Sergeivitch Termen).
The Theremin is considered the predecessor to the Moog Synthesizer. It is unique in that it is the first musical instrument that can be played without being touched.
1920's - The first electronic instruments appear. Theremin, Ondes Martenot and Trautonium
1925 - Electrical amplification (the microphone) was introduced. This invention forced engineers to re-design reproducers.
The Victor Company's answer to this revolution in sound was the Orthophonic Sound Box, which was very sensitive to high and low frequencies.
1931 - EMI researcher Alan Dower Blumlein invents stereophonic sound for recording.
1939 - Invention of the magnetic tape.
John Cage composes imaginary Landscape #1: the first piece to use electronic reproduction. The piece was performed on variable-speed turntables with RCA test tones and other sounds.
1940s - The first DJs emerge as entertainers for troops overseas.
During WWII, persons armed with a turntable, an armful of records, and a basic amplifier would entertain troops in mess halls, spinning Glen Miller, the Andrews sisters, and Benny Goodman. It was much easier than sending an entire band overseas.
1950s - Invention of the 45 RPM 7 inch records.
45 RPM records were cheaper to make and easier for American youths to carry to parties.
In Jamaica, as popularity of Jazz and R'n B increases, sound systems are used to promote the music. Sound systems developed from enterprising record shop disc jockeys with reliable American connections for 45s. They would load a pair of hefty PA speakers into a pickup truck and tour the island from hilltop to savannah, spinning the latest hits.
1951 - John Cage composes imaginary Landscape #4: the first piece to use radios as instruments.
1956 - Ska develops in Jamaica, which makes the sound system explode in popularity.
Karlheinz Stockhausen's 'Gesang der Junglinge' uses both natural sounds and electronically generated noises.
Duke Reid and Clement Dodd emerge as sound system operators in Jamaica.
1958 - Invention of the E-Piano
1959 - Artist begin conducting recording sessions that center on sound systems.
Duke Reid held his first recording session. This included the duo Chuck and Dobby, and the Jiving Juniors. He also recorded Derrick Morgan and Eric Morris for sound system play. Clement Dodd also held his first recording session recording over a dozen tracks with artists like Alton (Ellis) and Eddie (Perkins), Theophilius Beckford, Beresford Ricketts and Lascelles Perkins.
1960's - During the 1960's, modern electronics enters the music domain.
The first Moog Synthesizer hits the market created by Robert Moog.
New concepts and sounds begin to be used in music composition, such as mathematically based compositions by Arnold Schonberg and Erik Satie and "machine" sound by Luigi Russolo.
The late 1960's brought the birth of Dub music and the first remixes pioneered by King Tubby.
1960 - The "afterbeat" and "syncopation" concepts are born.
Prince Buster and Voice of the People begin to emphasize the afterbeat, which became the essence of Jamaican syncopation.
1966 - Rocksteady comes onto the scene in Jamaica.
1967 - Stockhausen Telemusik uses shortwave radio as instruments to create a "world music."
Late 60's - reggae takes over Rock Steady
Foundations for remix and rap music emerge.
Lee "Scratch" Perry, Edward "Bunny" Lee and Osbourne Ruddock (King Tubby) begin operating multi-track studios; they become major reggae producers.
1968 - King Tubby develops cutting
In his position as master cutter for Duke Reid, King Tubby regularly cut acetates (soft wax discs) that were designed exclusively for his own, and a few other, sound systems. When he left out portions of the vocal on a 'dub plate', (the local term for the acetate disc) he effectively created a new 'version' of a song.
1969 - Kool Herc, considered to be the first hip-hop DJ develops "Cutting Breaks." Kool Herc adapted his style by chanting over the instrumental or percussion sections of the day's popular songs. Because these breaks were relatively short, he learned to extend them indefinitely by using an audio mixer and two identical records in which he continuously replaced the desired segment. His particular skill, later copied by many others, was to meld the percussion breaks from two identical records by playing the break over and over switching from one deck to the other. Hip hop derived from "hip hoppin" on the turntable.
"Toasting" begins in dance halls - considered to be a direct link to rap music.
Technics introduced the Direct Drive System, SP-10
Early 70's - Technics released the original SL-1200 as a hi-fi turntable.
Giorgio Moroder is considered to be the pioneer of pro-synthesizer electronic disco music.
1971 - Ralf Hutter, Florian Schneider & Co. form Kraftwerk - the first electronic band.
1975 - Grand Wizard Theodore discovers the scratch.
1979 - Sugarhill Gang's "Rapper's Delight" is released. While they didn't really utilize the skills of a DJ, this song had a profound influence on the sound of commercial hip-hop during the early 1980's.
Late 70's - Technics does some work on 1200s turntables by improving the motor, redesigning the casing, and adding a separate ground wire and pitch control. It releases it as the sl-1200.
1980's - While playing at a club called the Warehouse, DJ Frankie Knuckles lays down drum machine-generated 4/4 beats on top of soul and disco tunes.
Marshall Jefferson develops a deep, melodic sound that relied on big strings and pounding piano. The result was 'Move Your Body' which became the house record of 1986.
12" disco records that included long percussion breaks (ideal for mixing) contribute to the emergence of House Music.
Grandmaster Flash is one of the first DJs to utilize the "breaks" of certain songs which when looped in a table to table fashion created the "breakbeat".
1980 - Roland introduces the TB-303 bassline machine and the TR-808 drum machine.
1981 - Grandmaster Flash's 1981 single "The Adventures of Grandmaster Flash on the Wheels of Steel" was Grandmaster Flash and The Furious Five's first record to demonstrate hip-hop deejaying skills
1982 - Afrika Bambaata's "Planet Rock" samples Kraftwerk and creates electro.
Grandmaster Flash and the Furious Five's "The Message" becomes a hit. "The Message" is seen by many as the first serious rap record.
1982 - Davy DMX's "One For the Treble" is released
1983 - Grandmaster D.S.T.'s "Megamix" is released
Herbie Hancock's "Rockit" featuring cuts and scratches by Grandmaster D.S.T. brings turntablism to a much wider audience
mid 80s - First affordable samplers (Akai s900) hit the market, which enable musicians to capture and manipulate existing sounds.
Other Hip-hop DJs in New York begin to use the spinback capabilities of the Technics 1200 turntable for "scratching", and to extend grooves and "breaks" by cutting back and forth between 2 copies of the same record as first exhibited by Grandmaster Flash.
1987 - The DMC (Disco Mix Club) holds its first annual DJ Competition
1989 - The rave scene develops.
The rave scene came out of Acid House and became so big that promoters came up with the idea of putting on huge events in the countryside outside London - events that held thousands of people and went on all night.
early 90s - Breakbeat emerges and produces Drum 'n Bass and Trip Hop.
Breakbeat, a descendent of Techno, has origins of Hip-Hop frenetic beats and high pitch samples. There are many variations of breakbeats: Darkside, Jungle and the most popular, Drum 'n Bass.
Trip Hop has roots in breakbeat and ambient and is a montage of beats, vocals, guitar & bass strings, and jazzy elements.
Steve Dee, strongly influenced by DJ Barry B. "The Cut Professor" from the Get Fresh Crew begins experimenting with "The Funk" which further develops and comes to be know as "beat juggling", or "remixing right before your eyes." He later founds the X-men who begin utilizing the style and take beat juggling to a higher level.
1990 - Mix Master Mike, and DJ Apollo form the first all turntable skratch band called "Shadow of the Prophet". They were the DJs for a rap group named F.M.2.0. and performed at various, radio shows and venues in the Bay Area.
1991 - Scratch DJ Innovator/Perfectionist DJ QBert gains worldwide attention in the Technics DMC DJ Championships
1992 - DJ Flare introduces the "Flare" skratch
QBert, Mix Master Mike, and Apollo dubbed as the "Rocksteady DJ's" by Crazy Legs.
92' also marks the year of the first skratch / battle record that was designed for ease of kutting and tricks because of the samples being on beat one after the other with no pause or lag time. It was called "Battlebreaks". The idea was then given to Darth Fader and the rest is history.
1994 - Shortkut, Disk, and QBert form the band, "Tern Tabel Dragunz" and perform at local Hip-Hop events around the Bay during 94'.
Shortkut Wins the Rap Pages DJ Battle in L.A. Strongly influenced by Steve Dee and the X-men, he also introduces his patented "Strobe" juggling technique and later in 94', wins the Technics DMC west coast championships.
Qbert's mixtape "Demolition Pumpkin Squeeze Musik" (dubbed by Rap Pages as the greatest Mixtape of all time) ignites the fire of the experimental skratch / mixtape revolution.
DJ Shadow releases "In/Flux" further fueling the movement towards a more sampler oriented movement in turntablism
1995 - Perhaps the winningest competition DJs ever, Qbert and Mix Master Mike retire from the DMC to become judges and enter a new challenge, the creation of music with turntables.
Mix Master Mike and Disk unknowingly create the name "Invisibl Skratch Piklz" for the crew by jokingly throwing out hundreds of goofy names for bands.
1995 also marked the birth of the first "all samples skratched song" by QBert entitled, "Invasion of the Octopus People" which starts another phase in turntablist culture.
With the help of Shortkut's initial introduction to them in 95', ISP became the first DJ band to be sponsored by, then a small manufacturer of DJ products, "Vestax". With the help of ISP's designs like the PMC 05 pro, 06 pro, 07 Pro and 05 Pro ltd., Vestax has now captured first place as the world's leader in sales of mixers and become the biggest DJ product company.
1996 - The I.T.F. (International Turntablist Federation) holds it's first world champioinship competitions
Showcasing the new era of turntablism, the historic battle at the Rocksteady Reunion between ISP and the X-Men (now called the X-Ececutioners) took place.
QBert gets filmed as a starring role in the movie, "Hang the DJ", which gets picked up by Miramax and plays in theatres in Europe, Canada, and the U.S.
ISP recorded the classic turntable orchestrated piece, "Invisibl Skratch Piklz Vs. Da Klamz uv Deth", on Vinyl.
1997 - Turntable T.V. was born on March 23, 1997 (the day of the Lunar Eclipse) and has now turned into an international turntablist video magazine featuring the Piklz practicing and hanging out with DJs from all over the world showing off their talents, skills, tips, tricks, and other turntable entertainment.
ISP filmed the educational and hilarious "Turntable Mechanics Workshop" for Vestax (check out tracoman.com). In this video, skratches were more publicly defined and given names so that turntablists may now share a mutual "skratch Language".
1998 - Yogafrog creates and gives away the first ISP music grant to aspiring artists in the Bay Area.
Mix Master Mike Joins the "Beastie Boys" in 98' and brings skratching to the eyes of the mainstream.
QBert Receives a lifetime acheivement award from the DMC
mid to late 90s - Individual DJs and crews such as the Invisibl Skratch Piklz, BulletProof Scratch Hamsters/Space Travelers, Allies, Supernatural Turntable Artists, Fifth Platoon, Beat Junkies, 1200 Hobos, Scratch Perverts, X-Men/X-Ecutioners, Cosmic Crew, and many others continue to expand on the frontiers of turntablism as an artform.

Fri 26 May 06 @ 1:16 pm

Armature
1882 by Thomas Watson

Dynamic
1874 by Ernst Siemens

Direct Radiator
1925 by Rice-Kellogg

Acoustic Suspension
1954 by Edgar Villchur

Flat Panel
1929 by E. W. Kellogg


1874 - Ernst W. Siemens was the first to describe the "dynamic" or moving-coil transducer, with a circular coil of wire in a magnetic field and supported so that it could move axially. He filed his U. S. patent application for a "magneto-electric apparatus" for "obtaining the mechanical movement of an electrical coil from electrical currents transmitted through it" on Jan. 20, 1874, and was granted patent No. 149,797 Apr. 14, 1874. However, he did not use his device for audible transmission, as did Alexander G. Bell who patented the telephone in 1876. After Bell's patent was granted, Siemens applied for German patent No. 2355, filed Dec. 14, 1877, for a nonmagnetic parchment diaphragm as the sound radiator of a moving-coil transducer. The diaphragm could take the form of a cone, with an exponentially flaring "morning glory" trumpet form. This is the first patent for the loudspeaker horn that would be used on most phonographs players in the acoustic era. His German patent was granted July 30, 1878 and his British patent No. 4685 was granted Feb. 1, 1878.

1898 - Oliver Lodge filed for British patent No. 9712 on Apr. 27, 1898, for an improved loudspeaker with nonmagnetic spacers to keep the air gap between the inner and outer poles of a moving coil transducer. This was the same year he applied for a patent on his famous radio tuner. A model of his loudspeaker is in the British Science Museum in South Kensington, and a photo was published in Wireless World Dec. 21, 1927. This improvement was later claimed by Pridham and Jensen in the Magnavox application for patent No. 1,448,279 filed Apr. 28, 1920, and granted Mar. 13, 1923.



1901 - John Stroh first described the conical paper diaphragm that terminated at the rim of the speaker in a section that was flat except for corrugations, filed for the British patent No. 3393 on Feb. 16, 1901, granted Dec. 14, 1901.

1908 - Anton Pollak improved the moving-coil loudspeaker with a voice-coil centering spider, filed for U.S. patent No. 939,625 on Aug. 7, 1908, granted Nov. 9, 1909.

1911 - Edwin S. Pridham and Peter L. Jensen in Napa, California, invented a moving-coil loudspeaker they called the "Magnavox" that was used by Woodrow Wilson in San Diego in 1919.

1915 - Harold Arnold began program at Bell Labs to improve phonographic sound recording. The first priority was the electronic amplifier using the new vacuum tube, second was the microphone, and third was the loudspeaker that would improve the "balanced armature" units developed for public address. After WWI, J. P. Maxfield led this project that produced E. C. Wente's moving coil speaker by 1925, the Orthophonic phonographic player by 1925, and Vitaphone talking motion pictures by 1926.



1918 - Henry Egerton on 1918/01/08 filed patent for balanced-armature loudspeaker, used in the Bell Labs No. 540AW speakers developed by N. H. Ricker Oct. 6, 1922, that became the 540 commercial speaker by 1924; was based on the balanced armature telephone patent of Thomas Watson granted Oct. 24, 1882, similar to devices also developed by Siemens and Frank Capps.

1921 - The Phonetron based on patent No. 1,847,935 filed Apr. 23, 1921, by C. L. Farrand, was the first coil-driven direct-radiator loudspeaker to be sold in the U.S. and was well-received, competing with the horns used by table radios

1923 - The Thorophone was a gooseneck loudspeaker with a voice-coil driver

1925 - The research paper of Chester W. Rice and Edward W. Kellogg at General Electric was important in establishing the basic principle of the direct-radiator loudspeaker with a small coil-driven mass-controlled diaphragm in a baffle with a broad midfrequency range of uniform response. Edward Wente at Bell Labs had independently discovered this same principle, filed patent No. 1,812,389 Apr. 1, 1925, granted June 30, 1931. The Rice-Kellogg paper also published an amplifier design that was important in boosting the power transmitted to loudspeakers. In 1926, RCA used this design in the Radiola line of a.c. powered radios.



1925 - Victor Orthophonic acoustic phonograph player had a folded exponential horn that was later used as model for the Klipsch speaker of the hi-fi era. Within a year, the Orthophonic faced competition from all-electric phonographs with an electromechanical pickup, vacuum-tube amplifier, and moving-coil loudspeaker, such as the Brunswick Panatrope sold by the Brunswick-Balke-Collender Company.

1926 - Vitaphone sound system for motion pictures used a new speaker developed at Bell Labs. Wente and Thuras designed the Western Electric 555-W speaker driver that was coupled with a horn having a 1-in. throat and a 40-sq. ft. mouth; it was capable of 100-5000 hz freq. range with an efficiency of 25% (compared to 1% today) needed due to low amp power of 10 watts. The power amps were 205-D. Older loudspeakers were balanced armature type, but the newer 555-W speakers of the Vitaphone were moving coil type.

1928 - Herman J. Fanger filed patent No. 1,895,071 on Sep. 25, 1928, granted Jan. 24, 1933, that described what came to be known as the coaxial speaker, composed of a small high frequency horn with its own diaphragm nested inside or in front of a large cone loudspeaker, based on the variable-area principle that made the center cone light and stiff for high frequencies and the outer cone flexible and highly damped for lower frequencies.



1929 - E. W. Kellogg filed patent No. 1,983,377 on September 17, 1929, granted December 4, 1934, that described an electrostatic speaker composed of many small sections able to radiate sound with out magnets or cones or baffles. This patent, as well as the 1932 British patents of Hans Vogt, influenced Peter Walker to build the Quad ESL flat panel speaker in 1957.

1929 - J. D. Seabert of Westinghouse developed a horn-type loudspeaker that directed the sounds of human speech toward the audience better than cone speakers that were intended for the over-all sound including music to fill the entire theater. These "directional baffle" horns had an opening 3 ft. by 4 ft. and were different from small-throat horns.
Thuras bass-reflex patent

1930 - Albert L. Thuras filed patent No. 1,869,178 on Aug. 15, 1930, granted July 26, 1932, for the bass-reflex principle while working at Bell Labs. Early cabinets used a passive baffle to direct sound to the front, allowing the back of the cabinet to be open for the low sounds. The bass-reflex enclosure kept the low-frequency sounds from being lost from the rear of the diaphragm.



1931 - Bell Labs developed the two-way loudspeaker, called "divided range" for the demonstration by H. A. Frederick in December of vertically cut records. The high frequencies were reproduced by a small horn with a frequency response of 3000-13,000 hz, and the low frequencies by a 12-inch dynamic cone direct-radiator unit with a frequency response within 5db from 50-10,000 hz. By 1933, a triple-range speaker had been developed for the Constitution Hall demo in April, adding Western Electric No. 555 driver units as the mid-range speaker. For the low frequency range 40-300 hz, a large moving coil-driven cone diaphragm in a large baffle expanding from a 12-in throat to a 60-inch mouth over a total length of 10 ft. This 3-way system was introduced in motion picture theaters as "Wide Range" reproduction.

1932 - RCA demonstrated a dual-range speaker of its own design for theaters, using three 6-inch cone diaphragms with aluminum voice coils in divergent directions, with a response of 125-8000 hz, and 10-ft. horns 40-125 hz.

1933 - "Progress was such that a demonstration of the new system - called "stereophonic" because of its ability to give a spatial sense corresponding to stereoscopic vision - was given before the National Academy of Sciences and many invited guests at Constitution Hall, Washington in the spring of 1933. Transmission was over wire lines from the Academy of Music in Philadelphia and three channels were used with microphones respectively at left, center and right of the orchestra stage and loud speakers in similar positions in Constitution Hall." This transmission of music "was carried out with special loud speakers developed for the purpose by Dr. Wente and the late A. L. Thuras. The objectives in the design of these loudspeakers were uniform response over the whole tonal range of the orchestra, an enhanced sound power output capacity without noticeable non-linear distortion and uniform distribution of the emitted sound at all frequencies throughout a wide solid angle. For the receiving unit and the multicellular horn which were developed for this demonstration, Dr. Wente, jointly with the Bell Telephone Laboratories, was awarded a gold plaque by the Academy of Motion Picture Arts and Sciences in 1936." (Bell Labs, 1953)



1935 - Douglas Shearer and John Hilliard at MGM developed a standard theater speaker system, starting with the Loews 5000-seat Capitol Theater on Broadway. James Lansing and Dr. John F. Blackburn of Cal Tech designed a 2-way speaker system; the high frequency driver had a 3-inch aluminum diaphragm and throat size of 1.4 inches; the low frequency baffled cone unit was 15 inches. ERPI provided speakers from Fletcher's hi-fi experimental equipment to help design the speakers. The low frequency horn used four 15-in. Lansing cone drivers and Lansing 284 drivers for multicell horns of different sizes. The system was installed in 12 theaters for the opening of "Romeo and Juliet" with Norma Shearer, sister of Douglas,

then installed in all Loews Theaters, then became the standard established by the Academy.

1940 - Paul W. Klipsch filed patent No. 2,310,243 on Feb. 5, 1940, granted Feb. 9, 1943, for the corner horn speaker.



1941 - Altec Lansing Corp. was formed when Altec bought Lansing; Altec Service Corp. (from "all technical") had been formed in 1938 by M. Conroe and George Carrington to manage ERPI installations after ERPI was dissolved. John Hilliard worked at Altec Lansing in 1943 on magnetic airborne sub detection and in 1945 put on the market the 2-way "Voice of the Theater" speaker system with improved horns and magnet drivers. See Lansing Heritage for images and a detailed history.

1949 - W. E. Kock and F. K. Harvey at Bell Labs developed the acoustical lens, and reported findings in 1949 JAES. These lenses are used in James B. Lansing theater speakers and home hi-fi speakers

1953 - Arthur Janszen was granted patent No. 2,631,196 on March 10, 1953, for an electrostatic high-frequency speaker



1954 - Acoustic Research introduced the small AR-1 bookshelf loudspeaker that used the acoustic suspension principle developed by company co-founder Edgar Villchur. This was soon followed by the $89 AR-2 and by the AR-3 with improved domed tweeters in 1958.

1957 - Quad ESL marketed as the first full-range electrostatic loudspeaker, designed by Peter Walker and David Williamson, based on Edward W. Kellogg's patent No. 1,983,377 filed September 17, 1929 and granted December 4, 1934.

1974 - Earthquake premiered Nov. 15 in the Chinese Theater in Hollywood with Universal Picture's Sensurround process developed by W. O. Watson and Richard Stumpf at Universal. Four large low-frequency horns were located behind the screen, two in each corner. The Model W horn in each corner was 8 ft. long, 4 ft. wide, 4 ft. high. The Model C horn in each corner was a modular unit 1 ft. wide and 5 ft. high. Two additional horns were located on a platform in the rear of the theater. Each horn was driven by a 1000-watt amplifier controlled by inaudible tones on a special optical control track along with the normal 4-track magnetic soundtrack of the 35mm Panavision filmstrip.



1982 - Return of the Jedi was the first movie exhibited on the THX sound system designed by George Lucas and Tomlinson Holman; THX "is comprised of customized acoustical design work for each auditorium, a

special screen speaker installation method, a proprietary electronic crossover network, and rigorous audio equipment specifications and performance standards."

1996 - The Verity Group in Britain formed New Transducers Ltd, now known as the NXT company , to develop the Distributed-Mode Loudspeaker (DML) based on the 1991 patent by Dr Ken Heron of Britain's Defence Evaluation & Research Agency (DERA)

1998 - Benwin marketed the first DML flat panel loudspeakers

Fri 26 May 06 @ 12:50 pm

Finally i did it! After long searching on net and putting articles together :)
Here is everything you should know about loudspeakers if you are a real DJ ;)
Enjoy reading!



Dynamic Loudspeaker Principle



A current-carrying wire in a magnetic field experiences a magnetic force perpendicular to the wire.


Loudspeaker Basics

The loudspeakers are almost always the limiting element on the fidelity of a reproduced sound in either home or theater. The other stages in sound reproduction are mostly electronic, and the electronic components are highly developed. The loudspeaker involves electromechanical processes where the amplified audio signal must move a cone or other mechanical device to produce sound like the original sound wave. This process involves many difficulties, and usually is the most imperfect of the steps in sound reproduction. Choose your speakers carefully. Some basic ideas about speaker enclosures might help with perspective.
Once you have chosen a good loudspeaker from a reputable manufacturer and paid a good price for it, you might presume that you would get good sound reproduction from it. But you won't --- not without a good enclosure. The enclosure is an essential part of sound production because of the following problems with a direct radiating loudspeaker.


Loudspeaker Details

An enormous amount of engineering work has gone into the design of today's dynamic loudspeaker. A light voice coil is mounted so that it can move freely inside the magnetic field of a strong permanent magnet. The speaker cone is attached to the voice coil and attached with a flexible mounting to the outer ring of the speaker support. Because there is a definite "home" or equilibrium position for the speaker cone and there is elasticity of the mounting structure, there is inevitably a free cone resonant frequency like that of a mass on a spring. The frequency can be determined by adjusting the mass and stiffness of the cone and voice coil, and it can be damped and broadened by the nature of the construction, but that natural mechanical frequency of vibration is always there and enhances the frequencies in the frequency range near resonance. Part of the role of a good enclosure is to minimize the impact of this resonant frequency.





Types of Enclosures

The production of a good high-fidelity loudspeaker requires that the speakers be enclosed because of a number of basic properties of loudspeakers. Just putting a single dynamic loudspeaker in a closed box will improve its sound quality dramatically. Modern loudspeaker enclosures typically involve multiple loudspeakers with a crossover network to provide a more nearly uniform frequency response across the audio frequency range. Other techniques such as those used in bass reflex enclosures may be used to extend the useful bass range of the loudspeakers.




Use of Multiple Drivers in Loudspeakers

Even with a good enclosure, a single loudspeaker cannot be expected to deliver optimally balanced sound over the full audible sound spectrum. For the production of high frequencies, the driving element should be small and light to be able to respond rapidly to the applied signal. Such high frequency speakers are called "tweeters". On the other hand, a bass speaker should be large to efficiently impedance match to the air. Such speakers, called "woofers", must also be supplied with more power since the signal must drive a larger mass. Another factor is that the ear's response curves discriminate against bass, so that more acoustic power must be supplied in the bass range. It is usually desirable to have a third, mid-range, speaker to achieve a smooth frequency response. The appropriate frequency signals are routed to the speakers by a crossover network.




Crossover Networks for Loudspeakers

Most loudspeakers use multiple drivers and employ crossover networks to route the appropriate frequency ranges to the different drivers.




Two-Way Crossover

Combinations of capacitors, inductors, and resistors can direct high frequencies to the tweeter and low frequencies to the woofer. This amounts to filter action. A two-way crossover network divides the frequency range between two speakers.




Three-Way Crossover



Combinations of capacitors, inductors, and resistors can direct high frequencies to the tweeter and low frequencies to the woofer. This amounts to filter action. A three-way crossover network divides the frequency range between three speakers.
A capacitor has lower impedance for high frequencies. In series with the high frequency speaker (tweeter), it acts to block low frequencies and let high frequencies through.



The inductor has a lower impedance for low frequencies. In series with the low-frequency speaker (woofer), it acts to block high frequencies and let low frequencies through.


Inductors

Inductance is typified by the behavior of a coil of wire in resisting any change of electric current through the coil.

Arising from Faraday's law, the inductance L may be defined in terms of the emf generated to oppose a given change in current:




Ported Bass-Reflex Enclosure

The bass-reflex enclosure makes use of a tuned port which projects some of the sound energy from the back of the loudspeaker, energy which is lost in a sealed enclosure. But care must be taken to avoid the back-to-front cancelation of low frequencies which characterizes unenclosed loudspeakers. This is avoided by tuning the cavity resonant frequency of the enclosure to the free-cone resonant frequency of the loudspeaker. This has the effect of projecting bass frequencies from the port in phase with the sound from the front of the cone, at least at the resonant frequency. The overall effect is the increasing of bass efficiency and the extension of the bass response to lower frequencies.




Enclosure Effects on Resonance

Putting a loudspeaker in a closed box will eliminate the back-to-front cancelation effect, but will shift the ouput curve upward in frequency compared to the infinite baffle. A bass reflex enclosure can extend the bass response significantly below the loudspeaker resonance.


Back-to-Front Cancelation



While the front surface of the cone of a loudspeaker is pushing forward to create a sound wave by increasing air pressure, the back surface of the cone is lowering the air pressure. Since the wavelengths of low frequency sound are large compared to the size of the speaker, and since those low frequencies readily diffract around the speaker cone, the sound wave from the back of the cone will tend to cancel that from the front of the cone. For most bass frequencies, the wavelength is so much longer than the speaker diameter that the phase difference approaches 180°, so there is severe loss of bass from this back-to-front cancelation.

This is one of the reasons why even the best cone-type loudspeaker must have an enclosure to produce good sound.


Coupling Loudspeaker to Air

How hard can you punch a handkerchief? Not very hard, because it offers so little resistance. A loudspeaker has a similar problem when it tries to punch sound energy into the air. The usual language is that the speaker has a poor "impedance match" to the air.

A loudspeaker without an enclosure does a very poor job of producing sounds whose wavelengths are longer than the diameter of the loudspeaker. For an 8-inch speaker, diameter of speaker equals wavelength at about 1700 Hz. Even for a 16-inch speaker, the diameter equals the wavelength at 850 Hz.
Besides the severe bass loss, the overall efficiency of such loudspeakers is low, about 3-5% compared to 25-50% for well designed horn type loudspeakers.


Impedance Match to Air

One of the reasons for the low efficiency of direct-radiating cone-type loudspeakers is the poor impedance match to the air that they are driving. The impedance which the air offers to the motion of a speaker cone can be mechanically modeled. The specific acoustic impedance of free air is approximately 42 ohms per square cm. For optimum efficiency the radiation resistance of the speaker cone should also be 42 ohms/cm2 , but for sound wavelengths longer than the diameter of the speaker, this impedance drops rapidly . The smaller the speaker, the poorer its low frequency production.

A loudspeaker without an enclosure does a very poor job of producing sounds whose wavelengths are longer than the diameter of the loudspeaker. For an 8-inch speaker, diameter of speaker equals wavelength at about 1700 Hz. Even for a 16-inch speaker, the diameter equals the wavelength at 850 Hz.

This is one of the reasons why even the best cone-type loudspeaker must have an enclosure to produce good sound. The enclosure increases the effective size of the loudspeaker.


Loudspeaker Resonance

Direct-radiating cone-type loudspeakers must be mounted so that they are free to vibrate. This mounting is elastic, so there is an inherent resonant frequency of the speaker cone assembly -- like a mass on a spring. This free cone resonant frequency distorts the sound by responding more strongly to signals near its natural vibration frequency. This non-uniform response changes the frequency content in terms of the relative intensities of the harmonics and thus changes the timbre of the sound. Since the cone is undamped, it tends to produce "ringing" or "hangover" with frequencies near resonance. If the resonance is in the bass range, the bass will be "boomy".


Audible Sound

Usually "sound" is used to mean sound which can be perceived by the human ear, i.e., "sound" refers to audible sound unless otherwise classified. A reasonably standard definition of audible sound is that it is a pressure wave with frequency between 20 Hz and 20,000 Hz and with an intensity above the standard threshold of hearing. Since the ear is surrounded by air, or perhaps under water, the sound waves are constrained to be longitudinal waves. Normal ranges of sound pressure and sound intensity may also be specified.

Frequency: 20 Hz - 20,000 Hz (corresponds with pitch)

Intensity: 10-12 - 10 watts/m2 (0 to 130 decibels)

Pressure: 2 x 10-5 - 60 Newtons/m2 2 x 10-10 - .0006 atmospheres

For an air temperature of 20°C where the sound speed is 344 m/s, the audible sound waves have wavelengths from 0.0172 m (0.68 inches) to 17.2 meters (56.4 feet).


Sound Intensity

Sound intensity is defined as the sound power per unit area. The usual context is the measurement of sound intensity in the air at a listener's location. The basic units are watts/m2 or watts/cm2 . Many sound intensity measurements are made relative to a standard threshold of hearing intensity I0 :



The most common approach to sound intensity measurement is to use the decibel scale:



Decibels measure the ratio of a given intensity I to the threshold of hearing intensity , so that this threshold takes the value 0 decibels (0 dB). To assess sound loudness, as distinct from an objective intensity measurement, the sensitivity of the ear must be factored in.


Sound Pressure

Since audible sound consists of pressure waves, one of the ways to quantify the sound is to state the amount of pressure variation relative to atmospheric pressure caused by the sound. Because of the great sensitivity of human hearing, the threshold of hearing corresponds to a pressure variation less than a billionth of atmospheric pressure.

The standard threshold of hearing can be stated in terms of pressure and the sound intensity in decibels can be expressed in terms of the sound pressure:



The pressure P here is to be understood as the amplitude of the pressure wave. The power carried by a traveling wave is proportional to the square of the amplitude. The factor of 20 comes from the fact that the logarithm of the square of a quantity is equal to 2 x the logarithm of the quantity. Since common microphones such as dynamic microphones produce a voltage which is proportional to the sound pressure, then changes in sound intensity incident on the microphone can be calculated from


where V1 and V2 are the measured voltage amplitudes .


Threshold of Hearing

Sound level measurements in decibels are generally referenced to a standard threshold of hearing at 1000 Hz for the human ear which can be stated in terms of sound intensity:



or in terms of sound pressure:



This value has wide acceptance as a nominal standard threshold and corresponds to 0 decibels. It represents a pressure change of less than one billionth of standard atmospheric pressure. This is indicative of the incredible sensitivity of human hearing. The actual average threshold of hearing at 1000 Hz is more like 2.5 x 10-12 watts/m2 or about 4 decibels, but zero decibels is a convenient reference. The threshold of hearing varies with frequency, as illustrated by the measured hearing curves.


Annotated Equal Loudness Curves




Loudness

Loudness is not simply sound intensity!

Sound loudness is a subjective term describing the strength of the ear's perception of a sound. It is intimately related to sound intensity but can by no means be considered identical to intensity. The sound intensity must be factored by the ear's sensitivity to the particular frequencies contained in the sound. This is the kind of information contained in equal loudness curves for the human ear. It must also be considered that the ear's response to increasing sound intensity is a "power of ten" or logarithmic relationship. This is one of the motivations for using the decibel scale to measure sound intensity. A general "rule of thumb" for loudness is that the power must be increased by about a factor of ten to sound twice as loud. To more realistically assess sound loudness, the ear's sensitivity curves are factored in to produce a phon scale for loudness. The factor of ten rule of thumb can then be used to produce the sone scale of loudness. In practical sound level measurement, filter contours such as the A, B, and C contours are used to make the measuring instrument more nearly approximate the ear.


"Rule of Thumb" for Loudness

A widely used "rule of thumb" for the loudness of a particular sound is that the sound must be increased in intensity by a factor of ten for the sound to be perceived as twice as loud. A common way of stating it is that it takes 10 violins to sound twice as loud as one violin. Another way to state the rule is to say that the loudness doubles for every 10 phon increase in the sound loudness level. Although this rule is widely used, it must be emphasized that it is an approximate general statement based upon a great deal of investigation of average human hearing but it is not to be taken as a hard and fast rule.



Why is it that doubling the sound intensity to the ear does not produce a dramatic increase in loudness? We cannot give answers with complete confidence, but it appears that there are saturation effects. Nerve cells have maximum rates at which they can fire, and it appears that doubling the sound energy to the sensitive inner ear does not double the strength of the nerve signal to the brain. This is just a model, but it seems to correlate with the general observations which suggest that something like ten times the intensity is required to double the signal from the innner ear.

One difficulty with this "rule of thumb" for loudness is that it is applicable only to adding loudness for identical sounds. If a second sound is widely enough separated in frequency to be outside the critical band of the first, then this rule does not apply at all.

While not a precise rule even for the increase of the same sound, the rule has considerable utility along with the just noticeable difference in sound intensity when judging the significance of changes in sound level.



Threshold of Pain

The nominal dynamic range of human hearing is from the standard threshold of hearing to the threshold of pain. A nominal figure for the threshold of pain is 130 decibels, but that which may be considered painful for one may be welcomed as entertainment by others. Generally, younger persons are more tolerant of loud sounds than older persons because their protective mechanisms are more effective. This tolerance does not make them immune to the damage that loud sounds can produce.



Some sources quote 120 dB as the pain threshold and define the audible sound frequency range as ending at about 20,000 Hz where the threshold of hearing and the threshold of pain meet.

Fri 26 May 06 @ 11:38 am

These tips and tricks have been tested out and have proved successful in many occasions. If you follow this document to the "T" you will have your system in true optimum performance. They are listed in descending importance.

By the way, there is a list of acronyms used in this guide at the bottom of the guide.

1) It is recommended for all Windows XP machines to change your pro audio card's buffer size to approximately 128 in its control panel. Typical defaults are 512 and 1024. Check your manual for where this setting is. There is a very desirable side effect of this as well - it drastically lowers latency in the audio system. If you are using Cakewalk Sonar, you'll want to re-run the Wave Profiler after this step.

* Pro Tools may require a buffer size of 256 or 512.

2) Make sure you don't have ANY IRQ conflicts of any sort with your audio/MIDI card. (Windows will not always tell you if you do.)

Right-Click MY COMPUTER, select PROPERTIES.

Select the HARDWARE tab, and DEVICE MANAGER.

Go to VIEW, select RESOURCES BY CONNECTION, expand INTERRUPT REQUEST.

This will show you a complete list of all the IRQs in use. If you see more than 2 devices on the same number, you might have an IRQ conflict.

Any device can, without problem, share with IRQ holder for PCI steering, or ACPI IRQ holder for PCI IRQ steering. Some other devices that will not usually cause problems are System Management Bus or SMBus, and SCI IRQ used by ACPI Bus.

Devices to watch out for (especially when they're sharing the sound card's IRQ) are:

Other sound cards, SCSI controllers, USB controllers, Network cards, Video cards, and many others.

If you have a conflict and you want to fix it, the best way to move the IRQs around is to move the card in question to another PCI slot. This may mean trying 2,3, or even 4 different slots. You may need to move another card to a different slot to allow the audio card to use that PCI slot. All unused PCI cards should be PHYSICALLY REMOVED (or built-in devices should be disabled from BIOS), not just disabled in Windows, as this does not always truly free up the resource. Remember to power down your system before changing hardware; you should NEVER install or remove hardware while the computer is powered ON, except for USB and Firewire devices (or PCMCIA Cardbus devices on laptops).

Some BIOS manufacturers may allow you to assign a specific IRQ to a PCI slot (from within the BIOS under PCI configuration - see below), but you should try to avoid IRQ 9 because it is cascaded to IRQ 2. It will allow you to assign PCI slot 2, for example, to IRQ 5.

Some more tricks for freeing up IRQs are to disable one or more COM (serial) ports, printer ports and USB (if you're not using them), or on-board audio (Sometimes called AC97 or Legacy Audio) from within the BIOS (see below). You must check with your motherboard/computer manufacturer to find the correct way to do this. Disabling devices within the BIOS will remove them from the system, and possibly allow a device or two to jump to a free IRQ, reducing the amount of troubleshooting you must do.

To get into your computer's BIOS, you'll need to reboot (or turn on) the machine. Immediately when you see the bootup logo or memory/hard drive check, press the appropriate key (which varies from computer to computer - check your computer's documentation if you're unsure) until it enters the setup. This will be before Windows boots, and typically it only waits for 1-2 seconds for you to press the key. The most likely keys are DEL , F1, and F2, but could also be any of the F-keys (you can usually press multiple keys at the same time if you're unsure.) Within the BIOS, you should select Integrated Perhiperals. If you don't see this option, try Advanced, and within this menu, look for I/O Configuration, Perhiperal Configuration, or something of the like- it varies from system to system. This is where you'll see the options to enable/disable hardware. This is also where you'll change the parallel port mode if you are using a parallel port midi interface. Then follow on-screen instructions on how to exit and save changes.

Another option within the BIOS, which may be under almost any menu, again depending on the computer, is PLUG AND PLAY BIOS or INSTALLED OS. This will have two choices, one being Windows, and one being non-Windows (the wording may be different.) This setting determines whether the BIOS or Windows will control resources. You may want to try switching the option to the other choice. If it doesn't work as well, it can always be changed back.

* If you're using a USB device (either audio or midi interface), you'll want to have the USB on its own IRQ. Most new computers actually have 2 or 3 USB devices: One pair on the back and one which is either connected to the front or not connected at all. Resolving IRQ sharing with USB is more difficult because there is no way to control which IRQ it uses, so usually you must move devices off of the USB IRQ by moving or removing the PCI card in conflict. Also, if you can avoid it, don't use any other USB device on that pair of USB ports. This will decrease the bandwidth for your audio/midi device and possibly create conflicts. Especially avoid hubs and high-bandwidth items like modems, ethernet controllers, hard drives or cd-rom drives, printers, and scanners.

* If you're using a Firewire audio device, you'll want to have the Firewire IEEE 1394 controller on its own IRQ just like it is your sound card. All firewire ports on a card share the same resources, so it shouldn't make a difference where they are plugged in unless there is a physical problem with the card.

2a) Also, while in Device Manager, you may want to check for multiple driver installations. Go to Start - Run, and type CMD and click OK. Type exactly:

set devmgr_show_nonpresent_devices=1

start devmgmt.msc

(The latter command actually launches your device manager- you can use this elsewhere or make a shortcut to this file if you desire.) Go to View - Show Hidden Devices. Don't worry- you'll see alot of other devices which you didn't see before- this is normal. For instance, if you have a network card, it may now show as 5-10 devices, these are background pieces of it required for normal operation. The same goes for Sound, Video, and Game Controllers- there will be 6-12 new devices, mostly Microsoft devices. Do not remove these, they are required for Windows use. You can now see if you have 2 or 3 (or more) of the exact same device driver installed of your audio/midi interface (unused devices show as a light-grey color). If you do, right-click on the extra devices, and click Uninstall. Reboot after removing the extra devices, and go back and check again. You may need to remove all of a particular device to get the driver installed just once. This step is typically not needed, but in some cases where you have gone through driver installation several times, it is worth checking.

2b) LAPTOPS

Laptops often are a special case for troubleshooting hardware conflicts because of the inability to move hardware to different slots. If your laptop does not exhibit IRQ sharing, start with disabling hardware you don't use (like a modem if you use ethernet, firewire if you're not using it, etc.) If your laptop has all devices sharing IRQ11, you'll likely need to disable all non-critical devices and add them back one at a time. The best way to disable devices is to disconnect them from your computer, and for built-in devices to disable them in the BIOS (see above for how to get into the BIOS). However, most laptops do not have the option to disable all devices in the BIOS, so you may need to disable the devices in Windows Device Manager (see above again). If this is the only way to disable the devices on your laptop, it will usually work fine.

You will always see a Cardbus or PCMCIA controller in your device manager. This is the PC card slot(s) on the side of your computer and is not necessary unless you are using it. Although it does not take much in the way of resources, it's best to disable it if you're not using them. With most laptops today, everything is built-in and these slots are not in high demand.

4-pin vs. 6-pin Firewire - what's the difference? The 4-pin Firewire connector on your laptop is compatable with standard 6-pin devices as long as 1) you have a cable to convert between 4-pin and 6-pin 2) your Firewire device does not draw its power from the Firewire bus. The 4-pin Firewire connector does not provide power to the Firewire bus like the 6-pin does. Most Firewire devices have external power supplies and this will not cause any problems.

2c) Some other BIOS settings to try

With many new features appearing on current motherboards, you should disable these and then re-enable them one-at-a-time to find out if they are causing a problem.

Hyperthreading - Pentium 4s of 2.4gHz and higher only

SATA (serial ATA) - on many motherboards

Intel built-in ethernet - Only on Intel boards, known to sometimes cause problems

IDE or SATA RAID - on many motherboards

3) Multimedia Settings. ** Very important for SONAR & PRO TOOLS (only if you do not have DIGI ASIO drivers installed) **

Go to START MENU, SETTINGS, CONTROL PANEL, SOUNDS & AUDIO DEVICES.

Select the SOUNDS tab, select No Sounds and answer NO to "Save current scheme?"

Select HARDWARE tab, select your pro audio card wave driver & hit PROPERTIES.

Go to the PROPERTIES tab, expand the device, double-click the newly listed device and checkmark the box "DO NOT MAP THROUGH THIS DEVICE" and hit OK.

This will disallow Windows from using your audio card while you're in a recording program. If you have a Soundblaster-compatable card still installed, Windows will now use it for Windows sounds.

4) Check for Windows XP Service Pack 1 & 2.

Right-click on MY COMPUTER, select PROPERTIES.

This will display Microsoft Windows XP Home Edition and on the 2nd line, Service Pack 1 & 2 if you have it.

5) Disable Virtual Memory, Visual Effects, System Restore, Auto Updates, Remote Control, and Error Reporting.

Right-click MY COMPUTER, select PROPERTIES.

Click the ADVANCED tab, click ERROR REPORTING (at the bottom).

Checkmark DISABLE ERROR REPORTING and hit OK.

Under Performance, click SETTINGS.

Select ADJUST FOR BEST PERFORMANCE.

Click the ADVANCED tab, under Processor Scheduling, select Background Services.

Under Virtual Memory, click CHANGE.

Select each drive letter shown, and select NO PAGING FILE and press SET.

Hit OK twice.

Click the SYSTEM RESTORE tab, and checkmark the DISABLE SYSTEM RESTORE.

Click the AUTOMATIC UPDATES tab, and uncheck Keep My Computer Up to date.

Click the REMOTE tab, and uncheck both boxes that say Allow (this may already be done).

Hit OK and reboot.

6) Make sure no extra programs are started up when you launch Windows.

Go to your START MENU, RUN, type MSCONFIG and hit OK.

Select STARTUP tab, and deselect everything.

If there is a question about an item, disable it. You can always re-enable it simply by checking the box in the future and rebooting.

7) Check your sound hardware/software manufacturers' websites for updates. Many times an updated driver is the simple solution for an error. If possible, you should keep your old driver just in case, and be aware of beta (not fully tested) drivers and updates.

8) Motherboard chipsets - VIA and INTEL

The chipset on your motherboard can be as important as the actual processor because all the data to and from the processor goes through this chipset.

If you're unsure which chipset your motherboard has, you can go into Device Manager and open System Devices, and look for your CPU-to-AGP or CPU-to-PCI bridge. The brand name of this device (Intel, AMD, VIA, SIS, ALI, or nVIDEA) will tell you what type of motherboard you have.

If you have a VIA chipset on your motherboard (VIA makes chipsets for both Athalon and Intel PIII/P4/Celeron systems), you should download the 4-in-1 drivers which include updates for Windows at http://www.viatech.com/.

If you have an INTEL chipset, you may want to update the chipset drivers for Windows. Go to Intel.com, Support & Downloads, Chipsets, Chipset software, and download and install the Intel Application Accelerator and Intel Chipset Software Installation Utility. Select your version of Windows and download these to a new folder (recommended). The updates will only install if needed and if they are compatible with your motherboard. This will update the Windows access to the features of the motherboard.

9) Parallel port midi interfaces - * no need to follow these if you don't have a midi device on your printer port. These are several extra steps which must be done to get them working properly.

First, Right-Click MY COMPUTER, select PROPERTIES.

Select the HARDWARE tab, and DEVICE MANAGER and go to Ports (COM & LPT).

Double-click on Printer Port (LPT1), go to Port Settings, select Use any interrupt assigned to this port and Enable legacy plug and play detection, hit OK.

If the Parallel Port says ECP Parallel Port , the mode must be set in your computer's BIOS to EPP (default is ECP.) See step #1 for steps to get into the BIOS.

* If you do not have EPP mode, there are several things to try. Check to see what the I/O address of the printer port is (in BIOS), it should be 378. If not, EPP mode may be unavailable. Also, performing a BIOS update (from your motherboard manufacturer's website) can yield extra features and settings. Some computers simply do not have this mode. In this case, you may need to try several settings, but Bi-Directional is the second-best option.

Legacy SoundBlaster emulation:

It is known that many Soundblaster cards will share the parallel port IRQ without telling you.

Right-Click MY COMPUTER, select PROPERTIES.

Select the HARDWARE tab, and DEVICE MANAGER and look for Creative Misc. Devices.

If found, double-click anything that says Legacy Emulation, and checkmark the Disable. Hit OK.

If not found, check in your Sound, Video, and Game controllers for anything that says Legacy Emulation or Legacy Audio and disable that.

10) Disable CD Auto Play.

Go to Start Menu - Run, type GPEDIT.MSC and hit Return.

Under Computer Configuration - Administrative Templates - System, double-click Turn Off Autoplay.

Select Enabled (enabling turning this feature off) and select Turn off Autoplay on: All Drives.

Click OK and close the window.

11) Hard drive DMA.

Direct Memory Addressing, or DMA, allows a device to access your RAM directly without taking CPU recources. For hard drives, this will greatly increase disk throughput and reduce CPU load, and usually causes a very noticable increase in overall system speed.

Most Windows XP systems will have this enabled and working by default, but it is worth checking to achieve maximum performance.

Right-Click MY COMPUTER, select PROPERTIES.

Select the HARDWARE tab, and DEVICE MANAGER and look for IDE ATA/ATAPI Controllers and double-click on Primary IDE channel, select Advanced Settings. Check your Transfer Mode, it should say DMA if available. Current Transfer Mode Setting should say DMA and then some mode number, higher numbers indicate faster transfer speed. If you see PIO mode in the Current Transfer Mode but you have DMA if available selected, you likely have a cabling problem with your hard drives. Some possible cures are (and you may have to consult your local computer guru): changing from Cable Select to Master/Slave designation, moving your hard drives to different cables, or even cable replacement. In a newer system (PIII or greater), you should have the ATA-66 cable, which has twice as many conductors as a standard ATA or IDE cable. This will ensure the fastest possible DMA speed your hard drive and hard drive controller support. Generally, the newer your system and hard drive are, the faster DMA mode it will support.

12) ACPI mode * for VERY Advanced users *

* This step is recommended to be one of the LAST RESORT options to try since it drastically changes your system's IRQ configuration. Current computers RARELY need this step done.

If you have all your device on IRQ 9, 10, or 11 and changing PCI slots doesn't change this, or if you have devices on IRQs higher than 15 in Windows XP it's because you are running in ACPI mode. This can cause problems due to sharing and IRQs the hardware doesn't know how to handle. Here's how you can switch back to the normal Win98-BIOS-controlled IRQs.

* This step is not recommended for DUAL PROCESSOR machines - disabling ACPI will disable dual-processor support.

* WARNING: YOU MUST HAVE ALL HARDWARE DRIVERS AVAILABLE AT RESTART. This will re-detect ALL your hardware, and any hardware drivers needed will be asked for. THERE IS A (slight) CHANCE THIS WILL RENDER YOUR SYSTEM UNBOOTABLE, do this at your own risk.

* If you have a USB mouse, you may need to find an old PS/2 (round connector) mouse. USB is one of the last devices detected in this process, and therefore your mouse pointer will be unusable for part of this process if it is USB.

To disable ACPI, you'll need to change the system setting from 'ACPI-PC' to 'Standard-PC'. Right-Click on My Computer -> Properties -> Hardware -> Device Manager -> Computer -> Advanced Configuration and Power Interface (ACPI) PC -> Driver -> Update Driver -> Install from a list or specific location -> Don't Search.. -> Standard-PC.

Note that changing this means that all drivers of your hardware are re-installed (keep the driver disks available). Additionally, make sure that PNP OS INSTALLED in BIOS is set to NO (very important). Also note that disabling ACPI mode may cause your computer to not power off when you perform a Windows shutdown. Simply push the power button after performing a shutdown.

Reversing this step - re-enabling ACPI - can be as simple asrepeating this step and choosing ACPI-compliant PC instead of Standard. Again, this will require drivers to be reloaded.

These settings have been tried and tested, and are known to solve most problems. Some more tips are to:

* Windows XP recommends 256mb of RAM or higher to run properly, check to see if you have at least 256mb.

* Limit the programs installed on the computer to as few as possible. If at all possible, limit use of this computer to audio.

* Defrag your hard drives regularly (depending on use)

We recommend purchasing a system utility to do this such as Norton because it is faster and does a better job.

* Use a separate hard drive for audio (7200 rpm minimum)

* Make sure if you're recording FROM a digital source to your computer, you have word clock on the audio card set to external, sp/dif, ADAT, TDIF, Word clock, or whatever source from where you are recording.

* Always remember to BACKUP, even if it is only onto another hard drive in the system.

ACRONYMS USED:

CPU - Central Processor Unit - basically the brains of your computer, Pentium, Athalon, etc.

BUS - A communication line that can have one or more devices attached

PCI - Perhiperal Connection Interface - the white slots in your computer

IRQ - Interrupt request - basically a communcation line from a device to a processor

BIOS - Built-In Operating System - Allows hardware configuration, handles memory, basic hardware, drives, features, etc.

ACPI - Advanced Connection Perhiperal Interface - Power management, IRQ assignment, developed by Microsoft

SMBUS - System Management BUS - Micromanages device status and sleep mode.

SCSI - Small Computer System Inteface/Interconnect - type of hard drive interface & drive, typically associated with high transfer rates, can have up to 7 or 15 devices on one bus

IDE - Integrated Drive Electronics - 90% of consumer hard drives and the interface that connects them, only 2 per bus

DMA - Direct Memory Addressing - Many applications of this, hard drives and PCI cards can access RAM directly and quickly

USB - Universal Serial Bus - Connects one or many devices to a single communication line, conceptually similar to Firewire

COM - Communication port (serial) - 9 pin connector on the back of your computer, rarely used now

OS - Operating System - Windows 98, 2000, XP, Linux, BeOs, MacOS, VMS, Unix, and many many more, they run your computer.

Fri 05 May 06 @ 1:35 pm

Balanced Audio


This tutorial explains how balanced audio systems work. It is suitable for people who have a basic understanding of audio cables and connectors (see my previous tutorilas), as well as simple wave interactions (such as how waves from different sources interfere with each other).


What is Balanced Audio?

Balanced audio is a method of minimising unwanted noise from interference in audio cables. The idea is that any interference picked up in a balanced cable is eliminated at the point where the cable plugs into a sound mixer or other equipment.
Balanced audio works on the principle that two identical signals which are inverted 180° out of phase will cancel each other out. The cables used in such systems are designed to carry two versions of the signal and manipulate the relative phases of these signals to eliminate noise.
This will make more sense when we look at how balanced cables work, but first we need to take a step backwards and look at unbalanced audio cables.


Unbalanced Audio Cables

Traditional unbalanced cables use two lines to transmit the audio signal - a hot line which carries the signal and an earth line. This is all that is required to transmit audio and is common in short cables (where noise is less of a problem) and less professional applications.



Note: Internal componentry (in sound mixers etc) is also unbalanced.


Unbalanced Audio Connectors

Unbalanced audio cables are commonly associated with the 1/4" phono jack connector and the RCA connector. However any single-pin connector used for audio is unbalanced. 3-pin XLRs can also be used for unbalanced cables. For more information about these connectors, including how to wire them, see my previous tutorials.




Balanced Audio Cables

Balanced audio cables use an extra line, and consist of a hot line (positive), cold line (negative) and earth. The audio signal is transmitted on both the hot and cold lines, but the voltage in the cold line is inverted so it is negative when the hot signal is positive. These two signals are often referred to as being 180 degrees out of phase with each other. This terminology can be confusing — it does not mean one signal is delayed until it is out of phase, it means one signal is effectively flipped upside down.





When the cable is plugged into an input (on a mixer or other equipment) the hot and cold signals are combined. Normally you would expect these two signals to cancel each other out, but at the input stage they are put "back in phase" (i.e. the inversion is reversed) before being merged together, so they actually combine to form a stronger signal.


Removing Noise

Along the length of the cable, noise can be introduced from external sources such as power cables, RF interference, etc. This noise will be identical on both hot and cold lines. This is known as a common mode signal - a signal which appears equally on both conductors of a two wire line.

So the hot and cold lines carry two signals: A desirable audio signal which has an opposite voltage on each line, and unwanted noise which is the same on both lines.

This is where the trick of balanced audio kicks in. At the input stage when the inverted audio signal is re-inverted to make both desirable audio signals the same, the unwanted noise is inverted (i.e. put out of phase). Viola - all the unwanted noise is cancelled out, leaving only the combined original signal.


Combining Balanced Cables

The standard connector for balanced audio is the 3-pin XLR. For details on wiring various configurations and connectors see my previous tutorials.

Unfortunately there is no official standard for wiring balanced audio cables, but the most common configuration is:

Pin 1: Shield (Ground)
Pin 2: Hot
Pin 3: Cold


Mixing Wiring Configurations

Using cables or equipment with different wiring configurations in the same system is a recipe for trouble. You may well find that audio signals start canceling each other out and leave you with nothing.

Many sound mixers have a "phase invert" switch on each channel. This swaps the phasing of the hot and cold pins to solve the mismatch problem.

Obviously the best plan is to keep your wiring consistent. Use the configuration above and you shouldn't experience too many problems.


Main rule:
The rule of thumb for audio systems is: Connect all shields, ground everything, balance wherever possible.

Thu 20 Apr 06 @ 9:29 pm

How Do Microphones Work?


The Basics

Microphones are a type of transducer - a device which converts energy from one form to another. Microphones convert acoustical energy (sound waves) into electrical energy (the audio signal).
Different types of microphone have different ways of converting energy but they all share one thing in common: The diaphragm. This is a thin piece of material (such as paper, plastic or aluminium) which vibrates when it is struck by sound waves. In a typical hand-held mic like the one below, the diaphragm is located in the head of the microphone.

Location of Microphone Diaphragm


When the diaphragm vibrates, it causes other components in the microphone to vibrate. These vibrations are converted into an electrical current which becomes the audio signal.
Note: At the other end of the audio chain, the loudspeaker is also a transducer - it converts the electrical energy back into acoustical energy.


Types of Microphone

There are a number of different types of microphone in common use. The differences can be divided into two areas:

(1) The type of conversion technology they use
This refers to the technical method the mic uses to convert sound into electricity. The most common technologies are dynamic, condenser, ribbon and crystal. Each has advantages and disadvantages, and each is generally more suited to certain types of application. The following pages will provide details.

(2) The type of application they are designed for
Some mics are designed for general use and can be used effectively in many different situations. Others are very specialised and are only really useful for their intended purpose. Characteristics to look for include directional properties, frequency response and impedance (more on these later).


Mic Level & Line Level

The electrical current generated by a microphone is very small. Referred to as mic level, this signal is typically measured in millivolts. Before it can be used for anything serious the signal needs to be amplified, usually to line level (typically 0.5 -2V). Being a stronger and more robust signal, line level is the standard signal strength used by audio processing equipment and common domestic equipment such as CD players, tape machines, VCRs, etc.

This amplification is achieved in one or more of the following ways:

• Some microphones have tiny built-in amplifiers which boost the signal to a high mic level or line level.
• The mic can be fed through a small boosting amplifier, often called a line amplifier or preamplifier
• Sound mixers have small amplifiers in each channel. Attenuators can accommodate mics of varying levels and adjust them all to an even line level.
• The audio signal is fed to a power amplifier - a specialised amp which boosts the signal enough to be fed to loudspeakers.



Dynamic Microphones


Dynamic microphones are versatile and ideal for general-purpose use. They use a simple design with few moving parts. They are relatively sturdy and resilient to rough handling. They are also better suited to handling high volume levels, such as from certain musical instruments or amplifiers. They have no internal amplifier and do not require batteries or external power.

How Dynamic Microphones Work

As you may recall from your school science, when a magnet is moved near a coil of wire an electrical current is generated in the wire. Using this electromagnet principle, the dynamic microphone uses a wire coil and magnet to create to create the audio signal.

The diaphragm is attached to the coil. When the diaphragm vibrates in response to incoming sound waves, the coil moves backwards and forwards past the magnet. This creates a current in the coil which is channeled from the microphone along wires. A common configuration is shown below.



Earlier we mentioned that loudspeakers perform the opposite function of microphones by converting electrical energy into sound waves. This is demonstrated perfectly in the dynamic microphone which is basically a loudspeaker in reverse. When you see a cross-section of a speaker you'll see the similarity with the diagram above. If fact, some intercom systems use the speaker as a microphone. You can also demonstrate this effect by plugging a microphone into the headphone output of your stereo, although I don't recommend it!


Technical Notes:

Dynamics do not usually have the same flat frequency response as condensers. Instead they tend to have tailored frequency responses for particular applications.

Neodymium magnets are more powerful than conventional magnets, meaning that neodymium microphones can be made smaller, with more linear frequency response and higher output level.



Condenser Microphones


Condenser means capacitor, an electronic component which stores energy in the form of an electrostatic field. The term condenser is actually obsolete but has stuck as the name for this type of microphone, which uses a capacitor to convert acoustical energy into electrical energy.

Condenser microphones require power from a battery or external source. The resulting audio signal is stronger signal than that from a dynamic. Condensers also tend to be more sensitive and responsive than dynamics, making them well-suited to capturing subtle nuances in a sound. They are not ideal for high-volume work, as their sensitivity makes them prone to distort.

How Condenser Microphones Work

A capacitor has two plates with a voltage between them. In the condenser mic, one of these plates is made of very light material and acts as the diaphragm. The diaphragm vibrates when struck by sound waves, changing the distance between the two plates and therefore changing the capacitance. Specifically, when the plates are closer together, capacitance increases and a charge current occurs. When the plates are further apart, capacitance decreases and a discharge current occurs.

A voltage is required across the capacitor for this to work. This voltage is supplied either by a battery in the mic or by external phantom power.


Phantom Power

Phantom power is a means of distributing a DC current through audio cables to provide power for microphones and other equipment.

The supplied voltage is usually between 12 and 48 Volts, with 48V being the most common. Individual microphones draw as much current from this voltage as they need.

A balanced audio signal connected to a 3 pin XLR has the audio signal traveling on the two wires – usually connected to pin 2 (+ve) and pin 3 (-ve). Pin 1 is connected to the shield, which is earthed. The audio signal is an AC (alternating current), whereas phantom power is DC (direct current).

The DC phantom power is transmitted simultaneously on both pin 2 and 3, with the shield (pin 1) being the return path. Since the DC voltage on the ‘hot’ and ‘cold’ pins (2 & 3) is identical, it is seen by equipment as “common mode” noise and rejected, or ignored, by the equipment.

If you put a volt meter on pins 1 & 2, or pins 1 & 3, you will see the 48v DC phantom power, but if you meter pins 2 & 3 (the audio carrying wires) you will see no voltage.

The DC voltage can be harnessed however, and used to power mics, mic-line amps, or indeed a video camera (in this case the DC voltage would travel up the video cable – and would need special equipment to filter this voltage).

Phantom powering is defined in DIN standard 45 596 or IEC standard 268-15A

Note: Audio signals transmit as AC current, whereas powered equipment requires DC current to operate. Phantom power is a clever way of using one cable to transmit both currents.

How is Phantom Power Generated?
Phantom power can be generated from sound equipment such as mixing consoles and preamplifiers. Special phantom power supplies are also available.

Does Phantom Power Affect the Audio?
No, it does not affect the quality of audio at all and is quite safe to use. However it is recommended that you do not supply phantom power to microphones which do not require it, especially ribbon microphones.


Condenser Microphone


The Electret Condenser Microphone
The electret condenser mic uses a special type of capacitor which has a permanent voltage built in during manufacture. This is somewhat like a permanent magnet, in that it doesn't require any external power for operation. Therefore you don't need to worry about batteries or phantom power.

Other than this difference, you can think of an electret condenser microphone as being the same as a normal condenser.


Technical Notes:

Condenser microphones have a flatter frequency response than dynamics.

A condenser mic works in much the same way as an electrostatic tweeter (although obviously in reverse).



Directional Properties

Every microphone has a property known as directionality. This describes the microphone's sensitivity to sound from various directions. Some microphones pick up sound equally from all directions, others pick up sound only from one direction or a particular combination of directions. The types of directionality are divided into three main categories:

1. Omnidirectional
Picks up sound evenly from all directions (omni means "all" or "every").
2. Unidirectional
Picks up sound predominantly from one direction. This includes cardioid and hypercardioid microphones (see below).
3. Bidirectional
Picks up sound from two opposite directions.

To help understand a the directional properties of a particular microphone, user manuals and promotional material often include a graphical representation of the microphone's directionality. This graph is called a polar pattern. Some typical examples are shown below.


Omnidirectional
Uses: Capturing ambient noise; Situations where sound is coming from many directions; Situations where the mic position must remain fixed while the sound source is moving.

Notes:
Although omnidirectional mics are very useful in the right situation, picking up sound from every direction is not usually what you need. Omni sound is very general and unfocused - if you are trying to capture sound from a particular subject or area it is likely to be overwhelmed by other noise.




Cardioid
Cardioid means "heart-shaped", which is the type of pick-up pattern these mics use. Sound is picked up mostly from the front, but to a lesser extent the sides as well.
Uses: Emphasising sound from the direction the mic is pointed whilst leaving some latitude for mic movement and ambient noise.

Notes:
- The cardioid is a very versatile microphone, ideal for general use. Handheld mics are usually cardioid.
- There are many variations of the cardioid pattern (such as the hypercardioid below).




Hypercardioid
This is exaggerated version of the cardioid pattern. It is very directional and eliminates most sound from the sides and rear. Due to the long thin design of hypercardioids, they are often referred to as shotgun microphones.
Uses: Isolating the sound from a subject or direction when there is a lot of ambient noise; Picking up sound from a subject at a distance.

Notes:
- By removing all the ambient noise, unidirectional sound can sometimes be a little unnatural. It may help to add a discreet audio bed from another mic (i.e. constant background noise at a low level).
- You need to be careful to keep the sound consistent. If the mic doesn't stay pointed at the subject you will lose the audio.
- Shotguns can have an area of increased sensitivity directly to the rear.




Bidirectional
Uses a figure-of-eight pattern and picks up sound equally from two opposite directions.
Uses: As you can imagine, there aren't a lot of situations which require this polar pattern. One possibility would be an interview with two people facing each other (with the mic between them).




Variable Directionality
Some microphones allow you to vary the directional characteristics by selecting omni, cardioid or shotgun patterns.

This feature is sometimes found on video camera microphones, with the idea that you can adjust the directionality to suit the angle of zoom, e.g. have a shotgun mic for long zooms. Some models can even automatically follow the lens zoom angle so the directionality changes from cardioid to shotgun as you zoom in.

Although this seems like a good idea (and can sometimes be handy), variable zoom microphones don't perform particularly well and they often make a noise while zooming. Using different mics will usually produce better results.



Microphone Impedance


When dealing with microphones, one consideration which is often misunderstood or overlooked is the microphone's impedance rating. Perhaps this is because impedance isn't a "critical" factor; that is, microphones will still continue to operate whether or not the best impedance rating is used. However, in order to ensure the best quality and most reliable audio, attention should be paid to getting this factor right.

If you want the short answer, here it is: Low impedance is better than high impedance.

If you're interested in understanding more, read on....


What is Impedance?
Impedance is an electronics term which measures the amount of opposition a device has to an AC current (such as an audio signal). Technically speaking, it is the combined effect of capacitance, inductance, and resistance on a signal.

Impedance is measured in ohms, shown with the Greek Omega symbol Ω or the letter Z. A microphone with the specification 600Ω has an impedance of 600 ohms.


What is Microphone Impedance?
All microphones have a specification referring to their impedance. This spec may be written on the mic itself (perhaps alongside the directional pattern), or you may need to consult the manual or manufacturer's website.

You will often find that mics with a hard-wired cable and 1/4" jack are high impedance, and mics with separate balanced audio cable and XLR connector are low impedance.

There are three general classifications for microphone impedance. Different manufacturers use slightly different guidelines but the classifications are roughly:

(1) Low Impedance (less than 600Ω)
(2) Medium Impedance (600Ω - 10,000Ω)
(3) High Impedance (greater than 10,000Ω)

Note that some microphones have the ability to select from different impedance ratings


Which Impedance to Choose?
High impedance microphones are usually quite cheap. Their main disadvantage is that they do not perform well over long distance cables - after about 5 or 10 metres they begin producing poor quality audio (in particular a loss of high frequencies). In any case these mics are not a good choice for serious work. In fact, although not completely reliable, one of the clues to a microphone's overall quality is the impedance rating.

Low impedance microphones are usually the preferred choice.


Matching Impedance with Other Equipment
Microphones aren't the only things with impedance. Other equipment, such as the input of a sound mixer, also has an ohms rating. Again, you may need to consult the appropriate manual or website to find these values. Be aware that what one system calls "low impedance" may not be the same as your low impedance microphone - you really need to see the ohms value to know exactly what you're dealing with.

A low impedance microphone should generally be connected to an input with the same or higher impedance. If a microphone is connected to an input with lower impedance, there will be a loss of signal strength.

In some cases you can use a line matching transformer, which will convert a signal to a different impedance for matching to other components.



Microphone Frequency Response


Frequency response refers to the way a microphone responds to different frequencies. It is a characteristic of all microphones that some frequencies are exaggerated and others are attenuated (reduced). For example, a frequency response which favours high frequencies means that the resulting audio output will sound more trebly than the original sound.

Frequency Response Charts
A microphone's frequency response pattern is shown using a chart like the one below and referred to as a frequency response curve. The x axis shows frequency in Hertz, the y axis shows response in decibels. A higher value means that frequency will be exaggerated, a lower value means the frequency is attenuated. In this example, frequencies around 5 - kHz are boosted while frequencies above 10kHz and below 100Hz are attenuated. This is a typical response curve for a vocal microphone.




Which Response Curve is Best?
An ideal "flat" frequency response means that the microphone is equally sensitive to all frequencies. In this case, no frequencies would be exaggerated or reduced (the chart above would show a flat line), resulting in a more accurate representation of the original sound. We therefore say that a flat frequency response produces the purest audio.

In the real world a perfectly flat response is not possible and even the best "flat response" microphones have some deviation.

More importantly, it should be noted that a flat frequency response is not always the most desirable option. In many cases a tailored frequency response is more useful. For example, a response pattern designed to emphasise the frequencies in a human voice would be well suited to picking up speech in an environment with lots of low-frequency background noise.

The main thing is to avoid response patterns which emphasise the wrong frequencies. For example, a vocal mic is a poor choice for picking up the low frequencies of a bass drum.


Frequency Response Ranges
You will often see frequency response quoted as a range between two figures. This is a simple (or perhaps "simplistic") way to see which frequencies a microphone is capable of capturing effectively. For example, a microphone which is said to have a frequency response of 20 Hz to 20 kHz can reproduce all frequencies within this range. Frequencies outside this range will be reproduced to a much lesser extent or not at all.

This specification makes no mention of the response curve, or how successfully the various frequencies will be reproduced. Like many specifications, it should be taken as a guide only.


Condenser vs. Dynamic
Condenser microphones generally have flatter frequency responses than dynamic. All other things being equal, this would usually mean that a condenser is more desirable if accurate sound is a prime consideration.

Thu 20 Apr 06 @ 3:44 pm

XLR to 1/4" Mono Jack
The most comon way to wire a 3-pin XLR to a 1/4 inch mono jack (or 6.5mm jack), is to join the -ve and shield together.
This can be done by either soldering the shield and -ve wires to the sleeve of the jack.



Or by soldering a jumper on the XLR.




XLR to 1/4" Stereo Jack (wired for balanced mono)

• XLR pin 1 to jack sleeve
• XLR pin 2 to jack tip
• XLR pin 3 to jack ring




XLR to 1x RCA

When connecting a 3-pin XLR to one RCA, you use the same wiring as if you were connecting an XLR to a 1/4" jack.
The -ve and shield of the XLR are joined together, either at the XLR end or the RCA end. The easiest way is to solder a link between pins 1 and 3 (shield and -ve) of the XLR, rather than trying to solder the shield and -ve wire to the sleeve contact of the RCA.



This produces an unbalanced audio cable.


XLR to 2x RCA

A 3-pin XLR with a stereo signal can be split into left and right by wiring pin 2 of the XLR to the tip of one RCA plug, and pin 3 of the XLR to another RCA tip. Pin 1 of the XLR connects to the sleeve of both RCA plugs.




Stereo Jack to 2x RCA

When a stereo 1/4" jack is being used for a stereo signal (as opposed to a balanced mono signal), the left and right parts of the stereo signal can be split off to two seperate connectors. For example, a stereo headphone output can be split into left and right connectors, and one possible use for this would be to use these two independant connectors to feed left and right monitoring speakers.

Sun 16 Apr 06 @ 8:45 pm

So here are described most common audio plugs used to connect audio equipment components like mixers, amplifiers, speakers and so on...

First is here RCA connector, called Chinch connector too. This is the most common plug designed as standard for audio cables. You can see it usually in pair for stereo usage. Left channel is usually black or white, right channel is usually red. But they are the same cables.
You will find yellow RCA plug too, but it is for Composite video. This one is the same as audio connectors, but yellow color is specified for video.





Second the most common plug known as “quarter-inch” plug is called Jack too. These plugs are usually used for microphones, headphones and sometimes for speakers, but you rather do not use them for this because it is easy to plug them out.
We know 2 different types of this plug. Stereo and mono.

Mono:



Stereo:




Third and actually the best plug is called XLR plug, sometimes Cannon too. This is the most professional plug ever made. It has security button to unplug it. It is used for microphones and speakers. It has also a ground cable that is used to alleviate the hum produced by long cables.





The newest connector at audio side is Neutrik Speakon plug. It is used as speaker’s plug only.





Plug for amplifiers is called Binding Post connector, more known as Banana plug. Many amplifiers have holes for these plugs at the back side as their main out. On the other side of cable go speakers with (usually) Neutrik Speakon or XLR connectors.






We know also 3,5mm MiniJack plug, also available in stereo and mono. It is usually find at headphones, many times are used for connecting computer sound cards to speakers or external sound devices. It has actually the lowest quality for use for DJ equipment.

Mono:



Stereo:

Sun 26 Feb 06 @ 12:23 pm

ATTENTION!
All these settings were find on different internet pages and they actually help to run VDJ faster, but you set all these settings on your own and nobody is responsible if these settings will damage your system.

1. Processor scheduling should be set to Programs and not to background services. Start > Settings > Control Panel > System > Advanced > Performance Settings > Advanced Tab > Programs.

2. Visual effects should be set to a minimum. Start > Settings > Control Panel > System > Advanced > Performance Settings > Visual Effects Tab > Adjust for best performance.

3. Switch Off Desktop Background Image Right Click Desktop > Properties > Desktop Tab > Background None.

4. Disable Screen Saver Right Click Desktop > Properties > Screen Saver > None.

5. Disable Fast User Switching Start > Settings > Control Panel > User Accounts > Change the way users log on or off > Untick Use Fast User Switching.

6. Switch Off Power Schemes Start > Settings > Control Panel > Power Options > Always On > Turn off monitor and turn off hard discs to Never.

7. Switch Off Hibernation Start > Settings > Control Panel > Power Options > Hibernate > Untick Hibernation.

8. Activate DMA on Hard Discs/CD ROMS Start > Settings > Control Panel > System > Hardware > Device Manager > IDE ATA/ATAPI Controllers > Right Click Primary IDE channel and Secondary IDE channel > Properties > Advanced Settings Tab > Transfer Mode to "DMA if available" for both devices.

9. Disable System Sounds Start > Settings > Control Panel > Sounds and Audio Devices > Sounds Tab > Sound Scheme to None.

10. Do Not Map Through Soundcard Start > Settings > Control Panel > Sounds and Audio Devices > Hardware Tab > (highlight your soundcard from the list) > Properties > Audio Devices > (highlight your soundcard from the list) > Properties, and check the "Do not map through this device" checkbox.

11. Disable System Restore Start > Settings > Control Panel> System > System Restore Tab. Tick the "Turn off System Restore on all Drives".

12. Disable Automatic Updates Start > Settings > Control Panel> System > Automatic Updates > Turn off automatic updating. I want to update my computer manually.

13. Startup and Recovery Options Start > Settings > Control Panel> System > Advanced > Startup and Recovery Settings > Untick Automatically Restart.

14. Disable Error Reporting Start > Settings > Control Panel> System > Advanced > Error Reporting > Disable Error Reporting.

15. Disable Remote Assistance Start > Settings > Control Panel> System > Remote > Untick Allow remote assistance invitations to be sent from this computer.

16. Fix Swap File (Virtual Memory) Start > Settings > Control Panel > System > Advanced > Performance Settings > Advanced > Virtual Memory Change > Custom Size. Set initial and maximum size to the same value.

17. Speed Up Menus Start > Run > Regedit > HKEY_CURRENT_USER > Control Panel > Desktop Folder. Set MenuShowDelay to 1.

18. Disable Offline Files Start > Settings > Control Panel > Folder Options > Offline Files Untick "Enable Offline Files".

19. Disable Remote Desktop Start > Settings > Control Panel > System > Remote > Untick "Allow users to connect remotely to this computer".

20. Disable Internet Synchronise Time Start > Settings > Control Panel > Date and Time > Internet Time > Untick "Automatically synchronize with internet time server".

21. Disable Hide Inactive Icons Start > Settings > Taskbar and Start Menu > Taskbar TAB > Uncheck "Hide Inactive Icons".

22. Disable Automatic Desktop Cleanup Wizard Start > Settings > Control Panel > Display > Desktop > Customise Desktop > Untick "Run Desktop Cleanup Wizard every 60 days".

23. Make sure the NTFS Last Access Time Logging (NTFS File Systems Only) is enabled. Start > Run > regedit > HKEY_LOCAL_MACHINE > System > CurrentControlSet > Control > Filesystem. Add a new DWORD value - "NtfsDisableLastAccessUpdate" and set value to 1.

24. Disable Notification Area Balloon Tips Start > Run > regedit > HKEY_CURRENT_USER > Software > Microsoft > Windows > CurrentVersion > Explorer > Advanced. Create a new DWORD value called EnableBalloonTips and set to 0.

25. Disable CDROM Autoplay To Disable CD Autoplay in Windows XP Pro -- [1] Start/Run/GPEDIT.MSC -- [2] Computer Configuration/Administrative Templates/System. -- [3] Locate the entry for Turn Off Autoplay and modify(enable). Start > Run > regedit > HKEY_LOCAL_MACHINE > System > CurrentControlSet > Services > Cdrom. Set autorun to 0.

26. Disable Disc Indexing Service Right Click Start > Explorer > Right Click Each Disc > Properties > Untick "Allow Indexing Service to index this disc for fast file searching".

27. Delete unused icons on your desktop.

28. Make sure you always have empty Trash.

29. Close all unused programs running in background and located in System Tray.

30. Uninstall all unused programs and delete their folders after uninstall.

31. Do the Defragmentation of your disks at least once a month.

32. Do not put laptops to sleep, rather completely turn them off.


... and happy Mixing ;)

Wed 13 Apr 05 @ 4:25 pm

xeonneutral@yahoo.com