The Tracer Store
Personal Forensics Services
Recording Restoration Service
DC 8 Software Training DVD
Audio Mentor
Audio Restoration DVD
Audio: The Movie
Audio Performer
Noise Reduction Software
Tape Repair Products 
Audio Training DVDs 
DC LIVE/Forensics

Forensics Audio Hardware
DCAT-3 Test Signal CDs
Record Cleaning Supplies 
Record Storage 
Custom Built PCs 
Professional Soundcards
Sign Up For Newsletter 
Free Demo Downloads
Basics Of Restoration 
What Might I Need?
Audio Examples
Compare Our Products
Forensic Training Classes
Technical Support
Current Versions/Patches
Reading Materials
Knowledge Database
Contact Us
Topics Of Interest
45 Record Restoration
78 Record Restoration
Archiving Audio

Audio Enhancement
Audio Forensics
Audio Restoration
Audio Sweetening
Audio Testing
Buzz Removal

Cassette Tape Restoration
CD Making
Cell Phone Noise Removal
Cleaning Audio
Click and Pop Removal
Distortion Removal
8 Track Tape Restoration

Forensic Audio

Garbled Recordings
Hidden Mic Recordings
Hiss Removal
Hum Removal
LP Record Restoration
Noise Reduction
Noise Removal Software
Records To CD
Scratchy Records
Tape Restoration
Tapes To CD
Vinyl Restoration

Voice Analysis Software
Voice Recordings
Voice Removal
Warped Records

Glossary of Terms

The following is an extensive list of terms that you may or may not be familiar with that you may encounter at our site or throughout your audio experience.  If nothing else...use a few every now and then so everyone thinks you're really smart!  This Glossary is borrowed from the Diamond Cut Millennium/LIVE manual...any reference to those programs is purely intentional. 

Acoustical Impedance

Acoustical impedance is the total opposition provided by acoustical resistance and reactance to the flow of an alternating pressure applied to a system.  More specifically, it is the complex quotient of the alternating pressure applied to a system by the resulting volume current.  The unit is the acoustical ohm.

Acoustical Reactance

Acoustical reactance is the imaginary part of the acoustical impedance.  Energy is not dissipated by acoustical reactance; it is only stored there. The unit is the acoustical ohm.

Acoustically Mastered

Acoustically mastered record recordings utilized only the energy of the sound waves created by the sound source to modulate the master cutting lathe stylus.  This recording technique had none of the benefits which signal amplification can provide to the recording process.  This is the method that was utilized from the time of the invention of the phonograph by Thomas Edison in 1876 up until around 1925, when vacuum tube amplifiers and microphones began to be employed in the mastering process.  By 1929, all of the major record companies had switched over to the "electrical process" of record mastering.

Acoustical Resistance

Acoustical Resistance is the real term of the acoustical impedance relationship.  This is the term responsible for the dissipation of energy.  The unit is the acoustical ohm.

A-D Converter

A device used to convert analog signals into digital (discrete time) signals, so that they can be signal processed by a computer algorithm.  The sound card in your computer contains an A-D converter and also a D-A (Digital to Analog) converter.  To be compatible with DC Millennium/LIVE, it must have at least 16-bit resolution.  However, the software does support 8 through 24 bit resolution sound cards.

(2^16 = 65,536).  In other words, your sound card must be able to divide the amplitude of audio signals into numerically sampled representations, the smallest division being one part in 65,536.  16 bit audio has the same resolution as “red book” CD Audio.  


The Audio Engineering Society


A step-by-step procedure for solving a mathematical problem.


The unit of electric current that is equal to one coulomb flowing per second.  Also, I = V / R, wherein V = Voltage in Volts, and R = Resistance in Ohms (also, see Ohms Law).


An electronic system which enables an input signal to control power from a source independent of the signal and thus be capable of delivering an output that bears some relationship to, and is generally greater than the input signal.  An audio amplifier performs this function producing a relatively linear relationship between the input signal and the output signal.  For more information on audio amplifiers, refer to Pre-Amplifier and Power Amplifier in the Glossary section of the Help File.


The loudness (or intensity) of a sound at any given moment in time, which is represented on the vertical axis of the audio workspace areas.  Amplitude in audio terms is usually expressed in relative terms (the ratio of two levels) in dB (decibels), although; sometimes it may be represented in absolute terms such as volts, or sound pressure level.


An electronic system in which signals are represented, amplified, and processed utilizing continuous voltages and/or currents (whose value could be expressed as an irrational number at any point in time) which are not quantized.  DC Millennium/LIVE utilizes several digital simulations of analog systems in its algorithms.   


Solvents that are made up of cyclic hydrocarbons which, after they evaporate, tend to leave little or no residue on the surface on which they were used.


Attenuation is the process of signal reduction, which is the opposite of the process of signal amplification.  Most filters attenuate signals outside of their passband and feed signals through with no attenuation (or amplification) within their passband.  Some filters, such as parametric and graphic equalizers are configured to provide either amplification or attenuation at any given frequency.  Devices such as volume control potentiometers, “L” Pads, “T” Pads and “H” Pads are used to attenuate signals independent of frequency, i.e. flat.  “L” Pads hold either the input or the output impedance constant as the attenuation factor is modified.  “T” Pads hold both the input and the output impedance constant as the attenuation factor is changed. “H” Pads perform the same function as “T” Pads, only for balanced line systems.  Here is a table of resistance multipliers for a symmetrical (equal input and output impedance) “T” Pad attenuator: 

(Note: R1 = R3)

R1 = Attenuator Input Resistor 

R2 = Attenuator Shunt Resistor

R3 = Attenuator Output Resistor


Attenuation (dB)

 R1 & R3 (ohms)                                       (Normalized)               

R2 (ohms) (Normalized)

















































To use this table, multiply the input (or output) impedance of your circuit by the numbers associated with the attenuation which you desire.  Remember, this table of values requires that the input terminating impedance and the output terminating impedance of the circuits on each side of the attenuator be present and of the same value.  To obtain values of attenuation which are not in this table, merely cascade "T" sections adding up to the value (in dB) which you desire.  For example, to achieve 23 dB, cascade a 20 dB section with a 3 dB section.


Audio Connection Standards         

1. Balanced  "XLR"  Standard:

     A. Pin # 1 =  Shield (Common)

     B. Pin # 2 =  + (Hot)

     C. Pin # 3 = - (Cold)


 2. 1/4 inch Stereo Phone Plug  (TRS)  for Balanced Audio Circuits 

      A. Tip = + (Hot)

      B. Ring = - (Cold)

      C. Sleeve =  Shield (Common)


 3. 1/4 inch Mono Phone Plug  (TR)  for Unbalanced Audio Circuits

     A. Tip = + (Hot)

     B. Sleeve = - (Shield)


4. RCA / Phono Plug

     A. Tip = + (Hot)

     B. Sleeve = - (Shield)


5. Amphenol 3 Pin Balanced Microphone Connector

     A. Pin #1 =  Shield (Common)

     B. Pin #2 =  + (Hot)

     C. Pin #3 = - (Cold)


6. Amphenol 4 Pin Microphone Connector (Balanced and


A.      Pin #1 =  Shield (Common)

B.       Pin #2 = + (Hot) Unbalanced  (Note: Unbalanced output is with respect to Shield)

C.       Pin #3 = + (Hot) Balanced

D.      Pin #4 = - (Cold) Balanced


7. DIN 5 Pin Connector (Tape Deck I / O Connector)

A.      Pin #1 =  Right Channel Record Input

B.       Pin #2 =  Shield (Common)

C.       Pin #3 =  Right Channel Playback Output

D.      Pin #4 =  Left Channel Record Input

E.       Pin #5 =  Left Channel Playback Output


8. 1 / 8 inch Mono Phone Plug (TR)

     A. Tip = + (Hot)

     B. Sleeve - - (Shield)


9. 1 / 8 inch Stereo Phone Plug (TRS) {The type used on most Sound Cards}

     A. Tip = Left Channel + (Hot)

     B. Ring = Right Channel - (Hot)

     C. Sleeve = Shield (Common)


10. Modular Phone Jack (- 48 volt, 4 terminal, 2 line system / United States)

     A. Red* or Blue or Blue with White Stripe = Line #1

         -  (Hot)

     B.   Green* or White or White with Blue Stripe = Line #1 + (Common)

     C.   Yellow* or Orange or Orange with White Stripe = Line #2 - (Hot)

     D.   Black* or White or White with Orange Stripe = Line #2 + (Common)

          * Denotes the most standard color code


Audio Frequency Spectrum

The range of frequencies between 20 Hz and 20 KHz.  A very high quality audio system capable of reproducing this frequency range should be able to do so within +/- 3 dB.

 The following is a listing of some common audio sources and the portion of the audio spectrum which they typically occupy including their harmonics:


Audio Source                                             

Fundamental plus Harmonics       

Fundamental only


130 Hz to 15 KHz

130 Hz to 1.8 KHz

Bass Drum:

50 Hz to 5 KHz


Bass Tuba:

40 Hz to 7 KHz

40 Hz to 375 Hz


70 Hz to 14 KHz

70 Hz to 900 Hz

Clarinet (Soprano):

150 Hz to 14 KHz

150 to 1.7 KHz

Cymbals (14 inch):

300 Hz to 17 KHz


Female Speech:

180 Hz to 10 KHz



250 Hz to 14 KHz

250 Hz to 2.5 KHz

Foot Steps:

80 Hz to 15 KHz


Hand Clapping:

100 Hz to 15 KHz



450 Hz to 15 KHz

450 Hz to 1.3 KHz

Jingling Keys:

1.5 KHz to 14 KHz


Male Speech:

100 Hz to 8 KHz



250 Hz to 15 KHz

250 Hz to 1.7 KHz


30 Hz to 6 KHz

30 Hz to 4.2 KHz


500 Hz to 15 KHz

500 Hz to 3.8 KHz

Pipe Organ:

32 Hz (sometimes 16 Hz) to 15 KHz

32 Hz to 8 KHz

Room Noise:

30 Hz to 18 KHz


Snare Drum:

80 Hz to 15 KHz


Timpani Drums:

50 Hz to 4.5 KHz



80 Hz to 8 KHz

80 Hz to 500 Hz


180 Hz to 9 KHz

180 Hz to 900 Hz


190 Hz to 15 KHz

190 Hz to 3 KHz





When analog magnetic tapes are recorded or reproduced, the gap of the respective head (recording or playback) should ideally be perfectly normal (perpendicular) to the direction of the tape movement.  If, in either of the two mentioned processes, the respective head gap is off-normal (off-azimuth,) two types of signal degradation will occur.  The first phenomenon results in the loss of the high-end of the audio spectrum frequency response. The second effect produces a phase shifting of one channel with respect to the other, thereby “smearing” the stereophonic image. A similar phenomenon occurs when a monophonic half-track reel-to-reel tape is reproduced on a quarter track machine.   Azimuth problems can be corrected by utilizing the “Time Offset” feature found in the File Converter, which is under the Filter Menu.

Bandpass Filter

A filter which only allows a range of frequencies to be passed without attenuation.  A wide bandpass filter is one in which an upper and a lower corner frequency need to be defined, and often several octaves will be passed in between without attenuation.  A narrow bandpass filter is one in which only a center frequency needs to be defined, and often has a bandwidth of an octave or less.  The center frequency for a narrow bandpass filter is sometimes referred to as its resonant frequency.  There are different bandpass shapes which can also be defined for narrow bandpass filters.

Berliner, Emile

Emile Berliner is widely know as the one who commercialized the lateral cut disc record format.  He first introduced his products into the market place in 1895, although he had spent the previous 10-year period developing his product.  However, Emile Berliner was not the inventor of the disc format or the lateral cut method for creating the undulations on a surface.  These principles were outlined in the earlier Edison phonograph patents.


"Blast" is a term which is used to describe a passage of sound on a recording which is disproportionately louder than the rest of the recording.  "Blasts" can be created by poor instrument placement on acoustic recordings, poor mixes on electrical recordings, or by poor planning of microphone placement in live recordings.  The term "blast" was used by recording engineers at least as early as the 1920's.     

Blue Amberol

The third (and final) generation of cylinder record which the Edison Company commercialized which was about 4 minutes in length.  These records were made of a celluloid recording surfaced mounted on a plaster of Paris core.  They were an improvement on the Edison Gold Molded black wax cylinders, which were only two minutes in length.  The rotational speed for Blue Amberols (and black wax cylinders) is 160 RPM.


A buffer is a memory sector which is used as a temporary storage location during input and output operations.  The "preview buffer" length is programmable in Diamond Cut, and is found in the preferences section of the Edit Menu.

Butterworth Filter

A Butterworth Filter produces a maximally flat amplitude characteristic in the pass band or the reject band (depending on whether it is used as a bandpass or a notch filter).  It has a critically dampened response at the corner frequencies, having no ripple, and therefore it introduces little distortion into the signal which is feeding it.  The Butterworth poles of signal transmittance are uniformly spaced on a semicircle, having its center on the imaginary axis.  Its half-power frequencies are those at which the circle intersects the imaginary axis.


Buzz usually refers to a series of harmonics related to the frequency of the AC power mains.  It differs from “Hum” in sound, because it usually contains a large number of higher frequency harmonics.  Buzz is best eliminated using the Harmonic Reject filter. 


An eight-bit word.  Each sample of a monophonic wave file is represented by two eight-bit bytes.  Two eight-bit bytes are used to represent all of the integer numbers between 0 to 65,535, which is the total dynamic range of your Diamond Cut Audio editor when it is operating in 16 bit mode.

1 kilobyte (Kbyte) = 1,024 bytes


Capacitance is the ratio of the electric charge given a body compared to the resultant change of potential.  It is usually expressed in coulombs of charge per volt of potential change and its basic unit is the Farad.  Energy is only stored (but not dissipated) in theoretical capacitance.  Time constants for audio filters are created with a combination of resistors and capacitors in various configurations.  High pass, low pass, bandpass, and notch filters can all be created with the appropriate combinations of resistors, capacitors, and operational amplifiers.  The corner frequency for a simple first order RC filter = 1 / 2 pi (R x C).  The principle of capacitance (and conservation of charge) is involved in the operation of condenser and electret microphones and electrostatic loudspeakers and headphones.

F (Farad) and uF (micro Farad or 1 X 10 ^ -6 Farad)

Note:  pi = 3.141592654 (approximately)

Cassette Tape Equalization Time Constants

Compact Cassette tapes (which operate at 1 7/8 ips) commonly utilize one of the following two equalization time constants based on the tape type:

1. Normal (IEC Type 1) (Usually Ferrous Oxide based):

   120 uSec

2. High (IEC Type 2) Usually Chromium Oxide based):

   70 uSec

Classification of Amplifiers 

Audio Amplifiers can be broken down into several classifications based on their degree of conduction relative to its input signal.  The VVA Virtual Valve Amplifier utilizes two of the following classifications.  The others are included in the description for completeness*:

Class A: The device or devices conduct for a full 360 degrees of the input signal.  These amplifiers can be wired either in single-ended or push-pull configurations.  Class A Audio amplifiers are usually used in pre-amplifier stages, or low power amplifier applications.  This circuit has the poorest electrical efficiency, but produces predominantly even order distortion.

Class B: Two devices are operated out of phase with respect to one another.  Each device conducts for only 180 degrees of the input signal.  When the two amplified signals are combined, the full input waveform is represented, only amplified. This type of circuit is plagued by a phenomenon known as “crossover distortion” at low signal levels. This configuration is reserved for low performance PA amplifiers or AM (Amplitude Modulated) communications modulators.  It is electrically efficient, but produces relatively large values of harmonic distortion especially at small signal levels.


Class AB: Two devices are operated out of phase with respect to one another, just the same as the Class B configuration.  However, each device conducts for more than 180 degrees of input signal, but less than 360 degrees.  This configuration produces a reasonable tradeoff between electrical efficiency and low distortion.  It is commonly found used in high – high power audio power amplifiers.  Since the circuit is symmetrical, distortion levels can be quite low.

 Class C: This configuration can consist of one or two devices which are conducting for anywhere between 90 to 180 degrees of the applied input signal.  It is reserved for RF circuits only.

*Note: There are additional classifications of amplifiers involving tap switching, multiple rail, and pulse width modulation techniques, which have not been included in this listing. 


Clipping is a phenomenon, which occurs when a signal (or numerical value) exceeds a system’s headroom.  This concept applies to both analog and digital systems.  The result of clipping is distortion.  The amount of distortion produced depends on the amplitude of the over-driven signal.  In Diamond Cut, clipping will occur anytime a signal or calculation produces a numerical value greater than 2^16 (or 65,536 counts or LSB’s).  Clipping can be observed as a flattening of the slope (horizontal line) of a signal at its peak on the Source or Destination workspace displays. 

Co-Axial Cable

A coaxial cable is one constructed in a manner in which the signal conductor is located in the center of the return conductor with a dielectric located in-between. This provides three notable characteristics for the cable:

1. The center conductor is shielded from the effects of “E” fields which may be present.  “E” field coupled current is returned back to signal ground with little effect on the signal itself.

2. The loop area formed between the two conductors is very small compared to other types of conductors thereby minimizing inductance and also susceptibility to “H” field coupling.

3. The cable exhibits a characteristic impedance which is independent of cable length (after past a few wavelengths) which is of a constant value related to its ratio of distributed inductance and capacitance.  This makes the cable suitable for carrying RF (radio frequency) signals over long distances.

Co-Axial cables are often used to carry low level signals from one audio device to another because of the first two mentioned characteristics.

Comb Filter

A comb filter (or Harmonic Reject filter) is a wave reject filter whose frequency rejection spectrum consists of a number of equi-spaced elements resembling the tines of a comb.  This filter is useful for getting rid of “Hum” type noise containing more than just the line frequency fundamental component.  In Diamond Cut, it is called the Harmonic Reject Filter, and for more details, please refer to the same.


An electronic device which is used to reduce the dynamic range of an audio signal.  They are often used to prevent overloading on certain mixer inputs (i.e. drums and vocals) in live performance applications.  Radio stations often use them to make themselves "sound louder" when tuning across the radio band without violating any FCC regulations on maximum % modulation or modulation index. 

Corner Frequency

The corner frequency of a filter is the frequency at which the signal has been attenuated by 3 dB relative to the pass band region of the filter. 


Crackle is a term used to describe relatively low levels of impulse noise found on old phonograph recordings.  It is very similar to impulse noise, only the peak amplitude is much smaller in comparison. Crackle sort of sounds like Rice Krispies just after you pour the milk in the dish.  Crackle is usually caused by slight imperfections in the record playing surface due to the use of coarse grain fillers in the record composition.  Sometimes, crackle is caused by gas bubbles that occur in the surface as the record "cured" after the stamping process.  Crackle can be filtered out most effectively with the Impulse or Median Filter.  Very old acoustic recordings may be even more effectively de-Crackled (and de-Hissed at the same time) with the Average Filter.


1/10 of a bel.  A bel is the basic unit for the measurement of sound intensity.  It is a log scale measurement system used for relating the ratio of two acoustical or electrical parameters.   Since electrical voltage, current, and power are used to represent sound through audio signals, the following mathematical relationships may be found to be useful when relating them in terms of outputs and inputs:

         dB (voltage) = 20 log V output / V input

        dB (current) = 20 log I output / I input

        dB (power) =  10 log P output / P input

 Note:  A doubling of a voltage or current represents a 6 dB change.  A doubling of power represents a 3 dB change. The following table shows the relationship between Voltage, Current, and Power ratios and Decibels:

 Current or              Decibels                   Power Ratio              Decibels

Voltage Ratio


1                              0              |                               1                              0

2                              6.0           |                               2                              3.0

3                              9.5           |                               3                              4.8

4                              12.0         |                               4                              6.0

5                              14.0         |                               5                              7.0

6                              15.6         |                               6                              7.8

7                              16.9         |                               7                              8.5

8                              18.1         |                               8                              9.0

9                              19.1         |                               9                              9.5

10                            20            |                               10                            10

100                          40            |                               100                          20

1,000                       60            |                               1,000                       30

10,000                     80            |                               10,000                     40

100,000                   100          |                               100,000                   50

1,000,000                120          |                               1,000,000                60                                                                           


dBm is the power level of a signal expressed in dB, and referenced to 1 milliwatt (0.001 watt).


V = (P x Z) ^ 1/2

where V = Voltage in Volts,  P = Power in Watts,  and  Z = Impedance in ohms.

Therefore, in a 500 ohm audio line distribution system,

V at 0 dBm  =  0.707 Volts 


dBv is the voltage level of a signal expressed in dB, and referenced to 1 volt peak to peak.  If a pure sine wave is the reference signal, its value would be 0.35 volts RMS.

D-A Converter

A device to convert digital signals back into analog signals so that they will be compatible with analog sound reproduction equipment.  Diamond Cut requires a D-A converter with 16-bit resolution (2^16 = 65,536).  It also supports sound cards capable of up to 24-bit resolution.

DC Offset

A DC offset is a fixed value of voltage which may have been added to a signal inadvertently.  It contains no audio information.  It can be eliminated by feeding the signal through the Diamond Cut high-pass filter set to 20 Hz and a slope of 6 dB/ Octave.


The reversal of a pre-emphasis process.  See Pre-Emphasis for more information.


De-essing is the process of decreasing the sibilance of over-modulated “ess” sounds produced by the human voice.  The Diamond Cut dynamics processor contains an algorithm for De-essing a signal containing this particular anomaly.

Diamond Discs

The trade-name for the records in the disc format produced by the Edison Company was "Diamond Disc."  These records were cut vertically (hill and dale) and could only be played on Edison Diamond Disc phonographs designed for this purpose.  They rotate at a speed of 80 RPM.  To extract the vertical component of a signal provided by a stereo cartridge when transferring Diamond Discs, use the Diamond Cut Mono (L-R) File Conversion feature.


In control loops, dither is the addition of a useful oscillation or noise signal into the system to overcome friction or hysteresis.  This improves the response of the control loop to very small changes of the system reference signal.  This principle has been extended to digital audio.  In this case it implies the addition of a random noise signal intermodulated with the LSB of the audio signal, effectively increasing the resolution of the system.  


Drive refers to the amplitude of a signal which is applied to an amplification device such as an electron tube or transistor.  It represents the ac component, rather than the dc (or quiescent) component applied to the input.  Since the above-described devices are intrinsically non-linear with regard to their transfer function, the larger the value of drive applied to the device, the greater will be the harmonic by-products.  The Diamond Cut Virtual Valve Amplifier allows you to adjust the drive level to the various amplifiers to vary the degree of “tube warmth”.  The program automatically compensates the output level, so that large values of drive do not produce substantial changes in overall system gain.


“Dry” is the term used to describe the signal output of a special effects generator (such as the Reverb) which contains only the non-processed signal. “Wet,” on the other hand, refers to the effect signal alone.  Like most special effect generators, the Reverb has an output mix control which allows you to transfer a signal from the effects generator which ranges from completely dry, to completely wet (no source signal), or to some mixture in between.

Dynamic Filter

A filter in which its corner frequency is varied as a function another parameter associated with the signal content of a sound source.  Most often the corner frequency is that of a low pass filter that is modulated by the rectified output of a high pass filter, although other schemes are possible. This sort of system changes bandwidth on the fly, and in co-ordination with the occurrence of high frequency content present in the source.  It can be done either in a feedback or in a feed forward manner, with advantages and disadvantages attendant to each technique.

Dynamics Processor

An electronic device used to modify the characteristic dynamic amplitude response of an audio signal.  These circuits can compress, expand, and de-ess (remove overly sibilant “esses”.)


Your “ear” is the most critical piece of equipment which you will be using in the audio restoration process.  If you are only restoring audio for yourself, the audio restoration process is much less demanding as compared to situations where you may be performing the job for broad-based public consumption.  In the second case, there are two very critical aspects of your “ear” which must be considered:

 A.      You must have good hearing.  If you have a hearing deficiency, you may have a difficult time making the subjective judgments that are critical to the production of a commercially viable product, which will be acceptable to the “ear” of the general public.   For example, if the “top-end” of your hearing is missing, it is more likely that you will produce restorations that seem harsh, hissy, and containing too many digital artifacts as far as the general public is concerned. 

B.      Even if you have an exceptional sense of hearing, you will need to develop a good “ear” for what the general public expects in terms of audio restoration.  This requires good judgment, and a great deal of experience.

Edison, Thomas Alva

Thomas A. Edison was the inventor of the phonograph in 1877 at his laboratory in Menlo Park, New Jersey.  Edison also invented the Carbon and the Condenser Microphone, and the "Edison Effect" which is the principle behind the early rectification and amplification devices that were used to develop the field of modern electronics.

Electrical Recording

A process which was commercialized around 1925 for mastering records in which microphones and electrical signal amplification was utilized to supply the energy required to modulate the cutting head stylus of the recording lathe.  Prior to the invention of electrical recording, the acoustic energy of the various sound sources in the recording studio was the only source of energy which modulated the cutting head stylus. Electrical recording allowed more of the subtlety and detail of music to be captured on the wax master.

Electron Tube

The predecessor to the modern transistor was the Electron Tube (also sometimes referred to as an electron “Valve”).  Dr. Lee DeForest invented the device around 1906.  He took a Fleming diode (a derivative of the Edison Effect light bulb - 1883), and installed a grid between its cathode and anode.  He observed that small voltage signals applied to the grid with respect to the cathode produced large current changes in the devices plate current. This device became known as the “triode”, having three active elements within it.   Thus was born the key device that became the foundation building block for the development of modern electronics, as we know it today.  Electron tubes are basically amplification devices, which can be used in a myriad of applications.  The Virtual Valve Amplifier uses the measured characteristics of real electron tube triodes and pentodes in various amplifier and rectifier circuit models to produce a versatile array of “tube-warmth” effects. 

Elliptical Stylus

The shape of the tip of certain phonograph record playing styli which improves the high frequency response as compared to standard conical styli.


The average amplitude of a Wave file as displayed in the Diamond Cut workspace when zoomed-out.


A device that performs the opposite function of a Compressor.  These devices increase the dynamic range of an audio signal source.  When the process of compression is used in the recording or transmission process, and the process of expansion is used in the playback or reception process, the technique is known as companding.  It is sometimes employed because it increases the signal-to-noise ratio of the analog recording or transmission process.


(Full Frequency Range Recording)  This is the equalization curve used on records marketed by London (Decca) Records.  It claimed a frequency response of 50 to 14,000 Hz as early as the mid 1930's.


 (Finite Impulse Response)

A digital, not recursive method for producing filters, which can produce a phase linear response characteristic.  FIR filters are always stable.


A relatively rapid frequency modulation of the information on a recording due to rapid changes in the velocity of the record, tape, or the soundtrack of the source.  Flutter is the rapid counterpart to Wow, occurring at a deviation rate in the range of 6  to 250 Hz.  This distortion could have been introduced in the mastering process, or the playback process, or a combination of both.  Diamond Cut is not capable for correcting this sort of problem in a sound recording at this time. 

Fractional Speed Mastering

Fractional speed mastering is the process of transferring a record at a slower speed to a new media, and then converting it to the proper speed at a later time.  This has two potential benefits:

1.       It allows persons who do not own 78 or 80 RPM turntables to make transfers of those types of records using their 45 RPM speed

2.       It allows severely warped records to be transferred without skipping. 

Frequency Response

The range of frequencies which a system will pass through without attenuation.  The frequency response of audio equipment is generally specified with the upper and lower corner frequencies defined at the -3dB points.  For most high performance audio system electronics, the frequency response will be at least as good as 20 Hz to 20 KHz +/-3dB.  However, loudspeaker systems rarely are able to reproduce the same spectrum within the specified linearity band.  

Fourier Transform

A set of mathematical relationships which allow complex waveforms to be resolved into a series of fundamental frequencies, plus a finite number of terms which describe the waveforms harmonics. Fourier transforms are said to allow signals in the time domain to be represented in the frequency domain. Certain mathematical manipulations are more easily performed in the frequency domain as compared to the time domain, and Diamond Cut takes advantage of this characteristic.  After the mathematical manipulations have been completed, the resultants are re-converted back into the time domain via Inverse Fourier Transforms in order to re-create the processed version of the original time domain waveforms.  This method is utilized in the Continuous Noise Filter.


In the context of audio restoration, full-duplex refers to a sound card which is capable of performing input and output functions simultaneously.  For example, an analog sound card which has full duplex capability will be able to take an analog signal and convert it into a digital signal, at the same time that it is converting a separate digital signal into an analog form.  The Live version of Diamond Cut  will require a full-duplex card in order for it to operate.


The amplification effect of an electronic system which is often expressed in decibels (dB).  For example, an amplifier which has a voltage gain of 20 dB produces an output voltage signal which is 10 times greater in amplitude compared to its input.  Many special effects audio processors produce "unity" gain. This implies that its output voltage will be equal to the input voltage (X 1 gain).  Unity gain allows many signal processors to be placed in cascade without concern that the last processor in the chain will become overloaded due to the amplification build-up through each previous processor in the chain.

 In general, Voltage Gain = Av = Vout / Vin


Voltage Gain in dB = Av (dB) = 20 log Vout / Vin

Total System Gain in dB = Subsystem #1 Gain in dB + Subsystem #2 Gain in dB + Subsystem #N Gain in dB (when the subsystems are connected in a cascaded configuration.)

Note: If the subsystem gains are not given in dB, the total system gain is the product of the various subsystem gain values. For example, the total Gain = (Subsystem #1 Gain) X (Subsystem #2 Gain) X (Subsystem #N Gain).

The gain of an electrical system can be given in terms of any of the following:

1.        Voltage:   (Av = Vout / Vin)  ---  (Voltage Gain in dB = 20 log Vout / Vin)

2.        Current:   (Ai = Iout / Iin)      ---  (Current Gain in dB = 20 log Iout / Iin)

3.        Power:     (Ap = Pout / Pin)    ---  (Power Gain in dB = 10 log Pout / Pin)

Generation Loss

Each time an audio signal is transferred from one medium to another, it will suffer some degree of “generational loss.”  These losses include noise buildup, distortion, phase jitter, quantization errors, etc.   In analog systems, generation loss is much more of a significant factor in signal degradation compared to that which will be found in digital systems.  In practical terms, analog signal transfers should be minimized in audio restoration work.  The best results are produced if the analog to digital conversion process is performed only once.  Ideally, the only analog process would be the original A-D transfer.  Once in the digital domain, all processing, including the final transfer to DAT or CD-R, can be performed by your computer and the Diamond Cut program.  The only future conversion back to the analog world would occur during the playback process of the CD or the DAT through your audio system.

Graphic Equalizer

A Graphic Equalizer is a signal processor in which the audio band is divided into smaller spectral bands (portions).  Each spectral band can be adjusted in terms of either the gain or the attenuation of the frequencies which fall within that band.  Most Graphic Equalizer are Octave based, and contain about 10 bands.  However, some are 1 /3 Octave based and have about 30 bands.  Octave based graphic equalizers (including the one contained within the Diamond Cut application) typically break the audio spectrum down into bands with the following center frequency values:

 31 Hz, 62 Hz, 125 Hz, 250 Hz, 500 Hz, 1 KHz, 2 KHz, 4 KHz, 8 KHz, 16 KHz.  

Ground Loop

A potentially detrimental loop formed when two or more points in an electronic system that are nominally at ground potential are connected by a conducting path.  The term usually is employed when, by improper design or by accident, unwanted noisy signals are generated in the common return of relatively low-level (audio) signal circuits by the return currents or by magnetic fields generated by relatively high powered circuits or components.

Harmonic Exciter

A Harmonic Exciter is an electronic device or algorithm, which synthesizes odd and/or even harmonics of the upper end of the audio spectrum presented to it, and then re-inserts them back into the signal path.  This device will “liven-up” olde recordings in which the upper musical registers are missing due to generational losses or lack of response to begin with.  It can also be used to enhance vocals, or stringed instrument recordings. The Exciter is found under the Virtual Valve Amplifier (VVA) system located under the effects Menu.  It uses real models of Electron Tube rectifier and amplifier circuits to accomplish its synthesis.


Harmonics are the odd and even multiples of a fundamental frequency.  In music, it is the distribution of these harmonics which provides the characteristic (or timbre) which gives each musical instrument or human voice a unique sound. 

Harmonic Distortion

Harmonic Distortion results from the interaction of a non-linear transfer function of a system on a signal. The non-linearity of the system create undesirable harmonic products (except in rock and roll) which modify the sound of the original signal.  Devices like transistors, vacuum tubes, microphones, phonograph cartridges, loudspeakers, and A to D converters all have non-linearities to some degree.  In some cases, feedback is used to correct for non-linearity and in other cases using the device only in a very limited portion of its total dynamic range is the method for minimizing the production of harmonic distortion.

Diamond Cut can produce signal distortion when one of the algorithms attempts to drive the system to full scale or beyond.  It is therefore necessary to be careful when applying the Gain Change or the Graphic Equalizer algorithms, both of which can increase the gain of the system causing signals to exceed the programs dynamic range.  The distortion produced as a by product of this mechanism is called clipping.

Harmonic Reject Filter

Please refer to “Comb” or “Multiple Notch Filter.


A unit for the measurement of frequency.  1 Hertz = 1 cycle per second.


When two frequencies mix with one another through a non-linear system, sum and difference signals are produced.  These signals are referred to as Heterodynes and are also sometimes referred to as “beat frequencies.”  Early audio oscillators constructed before 1935 used the heterodyne technique for producing output signals in the audio range by “beating” two RF (radio frequency) oscillators against one another.  They were referred to as Beat Frequency Oscillators (BFO’s,) and were eventually replaced with a gain-stabilized form of the Wien Bridge oscillator.  Spurious heterodyne signals are produced on the various AM radio bands due to adjacent channel inteference.  In the US, these signal are 10 KHz, and in Europe, they are 9 KHz.  The Diamond Cut notch filter can be used to remove them.

Hill and Dale

See Vertical Cut.

High Pass Filter

A filter which attenuates all frequencies which fall below its corner frequency.  The degree of attenuation of a signal outside of the filters pass band depends on the frequency of interest, and the corner frequency and slope (order) of the high pass-filter.  This type of filter is often used to reduce rumble and muddy bass on a recording. 


Random noise at the top end of the audio frequency spectrum is often referred to as Hiss.  Generally this is considered to be the random noise which is heard above 5 KHz.  A good example of "hiss" is the sound you will hear if you tune a FM tuner to the top or bottom end of the band where there are no stations transmitting with the "mute" button disabled.  (This is a form of limited bandwidth "white noise.")


Noise introduced into a recording or sound system which is harmonically related to the power line frequency.  In the US, this will be 60 Hz and in Europe, this will be 50 Hz., and in both cases, it will include harmonics of the line frequency.  The most common "hum" frequencies are the fundamental (usually due to ground loops) and /or its second harmonic (due to defective power supply filter capacitors in electronic equipment).  To attenuate Hum on a recording, use the Diamond Cut Notch filter set to either 50 or 60 Hz, depending on the hum frequency.  Start with a bandwidth setting of around 0.2 Octave.  Adjust the bandwidth to the minimum value required to effectively attenuate the Hum.  This will minimize the Notch filters effect on all other frequencies.

Impedance (Z)

The total opposition including resistance and reactance which a circuit element(s) offers to the flow of an alternating current, measured in ohms.  Z = ((R^2) + (Xc^2) + (Xl^2)) ^ 1/2

Wherein ---

 Z  = Impedance in ohms

R  = Resistance in ohms

Xc = Capacitive Reactance in ohms

Xl = Inductive Reactance in ohms

Some standard Input and Output Impedance values which you will encounter are as follows:

1.        1 ohm - The basic unit of measurement for Electrical Resistance,  Impedance, or Reactance

2.        2 ohms (sound re-enforcement systems), 3.2 ohms (antique audio),  4,  8, 16, and 32,  ohms - Standard Loudspeaker

Impedance's  (8 ohms is the most common in 1996 in the United States).

3.        50 ohms - Standard Unbalanced Co-Axial impedance for RF signal transmission

4.        75 ohms - Standard Unbalanced Co-Axial impedance for Television and FM signal transmission

5.        300 ohms - Standard Balanced impedance for Television and FM signal transmission

6.        377 ohms - Impedance of Free Space

7.        500 ohms - Standard Balanced Microphone impedance

8.        600 ohms - Standard Telephone Exchange Audio line impedance

9.        2,000 ohms - Antique Audio (headphones & 1920's vintage horn loudspeakers)

10.     20,000 ohms - Common single ended input impedance found on Professional Audio Equipment

11.     47,000 ohms - Common Magnetic Phono Cartridge Loading Impedance

12.     50,000 ohms - Standard Unbalanced High Impedance Microphone Impedance.

13.     100,000 ohms - Common Input Impedance on Audio Equipment

14.     1 M. ohms - De-Facto Standard, Oscilloscope Input Impedance

15.     10 M.ohms - De-Facto Standard, True RMS Voltmeter Input Impedance


Mathematically, an impulse function is an event of infinite amplitude, and infinitesimal time duration. In Diamond Cut terms, an impulse is a transient that begins and ends within somewhere between 50 uS to 1 mS, with amplitudes which are generally higher than the average program material in a wave file.


The inductance of a circuit component (most often a coil) is the rate of increase in magnetic linkage with an increase of current.  The unit of measurement of inductance is the Henry which corresponds to a rate of linkage increase of 10 ^ 8 Maxwell-turns or one Weber-turn per ampere of current.  Energy is stored (but not dissipated) in theoretically ideal inductors.  The principle of inductance is a strong element in the operation of electronic transducers such as loudspeakers, magnetic phono cartridges, dynamic microphones, and transformers.  Resonant circuits can be created utilizing a combination of capacitors and inductors.  The basic resonant frequency of such a circuit is given by Fr = 1 / 2 pi (L X C) ^1/2.  This principle can be used to create narrow bandpass and notch filters.

The unit of measurement of inductance = H  (Henry)

Note: pi = 3.141592654 (approximately)

Intermodulation (Distortion)

Intermodulation distortion is that which results from the modulations of the frequency components of a complex wave by each other due to system non-linearities.  The result of this process is the production of frequency components which are equal to the sums and differences of integral multiples of the components of the original complex wave.


The property by which matter which is at rest will tend to remain at rest, and matter which is in motion will tend to remain in motion (in the absence of friction). 


Input / Output refers to the ports into which electronic signals are fed to an electronic device and the ports from which electronic signals are derived from an electronic device.  Diamond Cut allows you to choose between several I / O ports, provided you have the sound cards to support the feature.

IPS (Inches per Second)

The linear velocity of magnetic tape moving past a recording or playback head is referred to in terms of its IPS (inches per second) value.  The following is a listing of common speeds used by tape recorders:

       Pro Reel to Reel    Home Reel to Reel      Comp. Cassette           Micro-Cassette



30            x                              -                               -                               -


15            x                              -                               -                               -


7 1/2        -                               x                              -                               -


3 3/4        -                               x                              -                               -


1 7/8        -                               -                               x                              x


15/16       -                               -                               -                               x


15/32*    -                               -                               -                               x


* This speed is also used by reel to reel analog data recorders.

- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -

Rotary Head Tape Recorder Speeds:

1.       DAT:

0.321 ips (8.15 mm / sec)

2.       VHS:

1.31 ips - SP (Standard Play)

0.66 ips - LP (Long Play)

0.44 ips - EP (Extended Play)

3.       Beta:

1.58 ips (4.0 cm / sec) - Beta I

0.797 ips (2.0 cm / sec) - Beta II

0.524 ips (1.33 cm / sec) - Beta III

KHz(Kilo Hertz)

The unit used in the measurement of frequency equal to 1000 Hertz.  In earlier times, this term was Kilocycles (per second.)


The unit used in the measurement of electrical resistance equal to 1000 ohms.


Latency is the delay time encountered when operating in Preview mode or in “Live” feed-through mode.  Maximizing the speed of your computer system minimizes latency.

Lateral Cut

A record recording technique in which the groove modulation (undulations) occurs in a side-to-side direction, as opposed to up and down.  This technique was popularized by Emile Berliner with his Victrola.


Computer ease for starting a program.  This is accomplished by double clicking on the appropriate Icon.

Least Significant Bit (LSB)

The smallest quantitized increment which an Analog to Digital or Digital to Analog Converter can resolve an analog voltage or current. (LSB’s are sometimes referred to as “counts.)  In Diamond Cut, this value is 1 part in 65,536 (or 1 part in + / - 32,768), for 16 bit system applications.  For 24 bit system applications, this value increases to 1 part in 16,777,216 (or 1 part in + / - 8,388,608.)  For other sampling rates use the formulae wherein SR = Sample Rate:

+ / - LSB’s = (2^(SR)) / 2


An electronic circuit or system consisting of non-linear elements that will not allow signals above a threshold value to pass through to its output.  An upward compressor will produce this effect when its ratio is set to a high value.  The Dynamics Processor can be used as a signal limiter when used in compressor mode when high values of “ratio” are selected.

Lissajous Figures

When two sine waves are displayed on an X-Y display, with one applied to the X-axis and the other to the Y-axis, the interacting vectors of the two waveforms are displayed.  These waveforms are referred to as Lissajous figures.  Signals having the same frequency but of differing phase (other than 180 degrees) will form elliptical patterns, the phase of which can be calculated from the intercepts of the waveform with the display axis.  This technique if often used to adjust the azimuth of tape recorder recording and playback heads.  A properly aligned tape head will produce no ellipse, but only a 45-degree line with a positive slope.  This can be done using the Time Offset feature found in the File Conversions Filter in conjunction with the X-Y plotter found under the View menu.

Low Pass Filter

A filter which attenuates all frequencies which fall above its corner frequency.  The degree of attenuation of a signal outside of the filters pass-band depends on the frequency of interest, the corner frequency, and slope (order) of the low-pass filter.  This type of filter is often used to reduce the hiss on a recording.  However, low-pass filters will also attenuate the "highs" on a recording at the same time, which make them generally undesirable for this application.

Magnetic Phono Cartridge

A device for converting the mechanical motion of a record stylus into electrical signals utilizing the properties of magnetic circuits.  There are three types of magnetic phono cartridges.  They are:

1.        Variable Reluctance (early magnetic cartridges)

2.        Moving Magnet (the most commonly used)

3.        Moving Coil (quite expensive and having costly stylus replacement)*

·         The output impedance of moving coil (MC) cartridges are in the 10 to 100 ohm range.  Therefore, they require special matching transformers or pre-pre-amplifiers in order to be able to drive a conventional magnetic cartridge input on an audio pre-amplifier. 


One million Bytes.  (Sometimes 1024 KBytes for disks)


 The median value of a series of numbers is the number which is in the center of the sorted string.  For example, in the series of numbers 2, 4, 7, 3, 8, 4, 4, 1, 9, the median value is 4. 


 The unit used in the measurement of time equal to 1/1000 of a second.


The unit used in the measurement of distance equal to 1/1000 of an inch.  The diameter of phonograph styli are generally specified in "mils."  (If the stylus is elliptical in shape, the larger of the two dimensions is generally given).


An electronic process in which one source modifies the characteristics of another signal source.  For example, an audio signal may be used to Amplitude, Frequency, or Phase Modulate a sine wave signal (called a carrier.)   The result would be an Amplitude Modulated carrier in the first case (AM). In the second case, the result would be a Frequency Modulated carrier (FM).  In the last case, the result would be a Phase Modulation (PM) carrier.  These are techniques used for transmitting radio, television, and data. Sometimes, in audio, one refers to the undulations on a record as record groove "modulation."


An audio signal or a wave file which contains only one unique channel of sound information.


Multi-path Distortion

Multi-path distortion is a phenomenon that can occur during FM broadcast reception.  It occurs when the receiving antennae pick up two signals from the same transmitter.  This dual pickup consists of the direct signal from the transmitter (usually a line of sight trajectory) and a second parasitic signal arriving at the antenna some time later.  The second signal is a reflected signal off of a mountain, building or other object, and arrives at the antennae some time after the main signal had arrived.  The time shift between the main signal and the reflected signal creates phase distortion of the de-modulated audio signal when these two signal mix together.  This phase distortion manifests itself in the last two octaves of the audio spectrum and sounds like “slurring” of the pronunciation of the letter “s” and general harshness.  It will sound worse on a stereo broadcast than on a monophonic one.  There are several cures for this problem.  Purchase a directional antennae (one with a high front to back ratio) and install it as high as possible, aiming it towards the transmitter of interest.  Secondarily, you can minimize the problem by switching over to monophonic during a particularly distorted broadcast.  And lastly, when all else fails, you can reduce the distortion by utilizing the “de-esser” found in the Dynamics Processor.

Multiple Notch Filter

The term used in the Diamond Cut program used to describe a comb filter. A comb filter is a wave reject filter whose frequency rejection spectrum consists of a number of equispaced elements resembling the tines of a comb.  This filter is useful for getting rid of “Hum” type noise containing more than just the line frequency fundamental component.  This type of noise is line frequency related noise and sometimes described as “Buzz.”  This results from the interaction of non-linear systems with the finite output impedance presented by the power line sine wave voltage waveform, adding harmonics to the same.  Buzz can also be introduced into and audio system through non-sinusoinal current waveforms producing “H” fields which couple into noise sensitive loop areas (or ground loops) in audio systems. 

Musical Scale

There are two relatively common musical scales.  They are the Scale of Just Intonation, and the Scale of Equal Temperament.  The Scale of Just Intonation requires at least 30 discrete frequencies for each octave, making it relatively impractical to build musical instruments with fixed tones to play in the Just Scale.  Therefore, the scale of Equal Temperament containing only 12 notes per octave is the one in general use. 

The following table provides the frequencies of two Octaves of the tempered scale (1/2 step between notes) rounded in integers:


A  (below middle C)

= 220Hz

A (above middle C)

= 440 Hz

A (above high C)

= 880 Hz

A#  (or B flat)

= 233 Hz

A # (or B flat)

= 466 Hz




= 247 Hz


= 494 Hz



C (middle C)

= 262 Hz

C (high C)

= 523 Hz



C#  (or D flat)

= 277 Hz

C# (or D flat)

= 554 Hz




= 294 Hz


= 587 Hz



D# (or E flat)

= 311 Hz

D# (or E flat)

= 622 Hz




= 330 Hz


= 659 Hz




= 349 Hz


= 698 Hz



F# (or G flat)

= 370 Hz

F# (or G flat)

= 740 Hz




= 392 Hz


= 784 Hz



G# (or A flat)

= 415 Hz

G# (or A flat)

= 831 Hz



Note: Standard Pitch is based on the tone “A” of 440 Hz.  With this standard, the frequency of Middle C should actually be 261.626 Hz.

NAB Equalization Curve(National Association of Broadcasters)

The NAB Curve is a set of equalization frequency response contours which are used by manufacturers of analog tape recorders to compensate for the inductive nature of a tape head.  The equalization time constants specified depend on tape speed.  One pair of time constants are specified for 1 7/8 ips (inches per second) and 3 3/4 ips.  Another pair of time constants are specified for 7 1/2 ips and 15 ips.  The low frequency breakpoint for all speeds is 50 Hz.  The high frequency breakpoint for 1 7/8 and 3 3/4 ips is specified as 1770 Hz.  The high frequency breakpoint for 7 1/2 and 15 ips is specified as 3180 Hz.


Unwanted disturbances superimposed upon a useful signal that tends to obscure its information content.  Also, refer to Signal-to-Noise ratio for more information.

Noise Gate

A noise gate is an electronic device, which turns off a signal path when an input signal is below a predetermined threshold value.  The Dynamics Processor produces a noise gate effect when you check the Expander/Gate function.  You must set the ratio to the highest number for the best noise gate effect.

Notch Filter

A filter which attenuates all frequencies close to the center frequency of the filter setting. The degree of attenuation and the range of frequencies which are attenuated by this filter are determined by the filters Q or bandwidth.  This type of filter is often used to minimize hum or acoustic feedback from a recording. This type of filter is sometimes referred to as a "band reject filter."


An octave is a group of eight musical notes and also a doubling of frequency.  For example, the range of frequencies from 440 Hz to 880 Hz is 1 octave. The next octave will end at 1760 Hz.  Note that in two octaves, the frequency has increased by a factor of four.


A DC value of voltage or current added into a circuit to shift the quiescent operating point of a device or display.  Offset is used in Diamond Cut to allow detail to be seen in a signal when the detail exists towards the top or bottom of the signal workspace display area.

Ohm (R)

The unit of electrical resistance equal to the resistance of a circuit in which a potential difference of 1 Volt produces a current flow of 1 ampere.

Ohms Law

V = I x R  wherein V = voltage in Volts, I = current in Amperes, and R = resistance (in Ohms)


When an audio signal is applied to an audio device which is greater than the device can handle in a linear transfer manner, this creates a condition of "over-modulation."   It results in a distorted sound in the output of the device being over modulated.  Sometimes, this condition is referred to as "clipping," meaning that the amplification devices of an electronic system are either cutting-off or saturating due to overdrive.

Parametric Equalizer

A variable electronic filter in which the following three parameters may be adjusted on each parametric channel:

1.       Frequency

2.       Level (attenuation or amplification)

3.       Bandwidth

Parametric equalizers are usually equipped with several parametric channels which can all be used simultaneously or each one can be individually bypassed.


Pathe Freres Phonograph Company was a European based record and phonograph company, who utilized a somewhat unique groove modulation technique.  Their method produced a vertical stylus displacement (like Edison Hill and Dale Diamond Discs and Cylinders) however; this was accomplished by a different mechanism.  The groove on these recordings is “width” modulated, and so when a conical stylus interacts with these groove width modulations, a vertical displacement is thereby produced. If you are transferring a Pathe 78 rpm recording with a stereophonic pickup cartridge, you will need to use the Diamond Cut Mono (L - R) file conversion algorithm.


A Pentode is an electron tube (or valve) containing five elements.  They include a cathode, anode, control grid, screen grid or beam deflector electrode, and a suppressor grid.  They are most commonly used in audio power amplifiers, but are sometimes found in microphone pre-amplifiers.  Typical beam power pentodes listed in ascending power levels include types 6BQ5/EL84, 6L6GC, 5881, 7591, KT-66, 6CA7/EL34, KT-88, and 6550. 

Phase Inversion

Phase inversion is the phenomena when one of two signals has become 180 degrees phase shifted with respect to the other.  This sometimes accidentally occurred on vinyl stereo recordings because the input leads to one of the two cutting lathe driver heads became “swapped” in location.  This can be corrected by using the File Converter, using the Left or Right Phase-Invert feature. 


Pi (Greek Letter) is the symbol which relates the ratio of the circumference to the diameter of a circle.

Pi = C / D wherein C = Circumference of a Circle; D = Diameter of a Circle.

Pi is approximately =  3.141592654

Pink Noise

Pink Noise is random noise, which is characterized as containing equal energy per unit octave.  When viewed on an octave based spectrum analyzer, it will produce a flat horizontal line on the display.  Pink Noise is useful for characterizing the frequency response of electronic systems and for analyzing room acoustic transmittance and resonance.  Pink noise can be created through a two-step process using Diamond Cut.  First, create white noise with the Makes Waves function.  Next, process the signal through the Paragraphic equalizer using the factory preset labeled “white to pink noise converter.”


Power is the time rate for the transfer of energy in any system.  In other words, Power = Energy / time.

In electrical terms, power is given in Watts and has the following relationships to Voltage, Current, and Resistance: 

 P = V x I


P = Power in Watts, V = Voltage in Volts, and  I = Current in Amperes.


P = (I ^ 2) R


P = (E ^2) / R


R = Resistance in ohms

Power Amplifier(Power Amp)

A power amplifier is a device that provides power amplification of an audio signal.  Generally, this is the device that is used to drive a loudspeaker, the cutting head of a record lathe, or an audio transmission line, and is the final stage of amplification in an audio system.  Audio power amplifiers generally develop somewhere between 10 to 1000 watts of output power, depending on make and model (although shake table audio amplifiers and AM radio transmitter modulators can be found which produce well over 50,000 watts).

 To minimize power loss in the transmission process, and to maximize system dampening factor, it is important to minimize voltage drops across loudspeaker distribution cables.  Poor dampening factor will produce an ill-defined bottom-end (bass).  Long distances between your power amplifier and your speaker system will require larger diameter cables.  To determine the correct cable for your application, refer to the Wire Table provided in this Glossary.


A device that provides voltage amplification of an audio signal.  Sometimes these devices also include equalization networks and/or tone (bass, treble, loudness, etc.) controls.  


The intentional added amplification which is sometimes applied to the top end of the audio spectrum during a recording or radio transmission process in order to raise the signal level at high frequencies substantially above the noise level of the system.  This process is reversed during the reproduction process of the signal in order to recreate an overall flat frequency response.  The result of this process is an improvement in the signal-to-noise ratio of the system.  For example, the third specified time constant of 75 uSec associated with the RIAA equalization curve is pre-emphasis.  Also, FM broadcast transmission utilizes a 75 uSec (or sometimes a 25 uSec) pre-emphasis to improve its signal-to-noise ratio.  This process is reversed at your receiver (de-emphasis.)  The Paragraphic equalizer contains 75uSec pre-emphasis and de-emphasis preset curves.


Most of the filters and effects have a plethora of descriptive presets.  Most often, the most efficient place to start when using a particular filter or effect would involve selecting one of the factory presets, and then tweaking the parameters to fine tune the system to your own personal taste.  If you desire to keep a separate copy of your presets on diskette, it can be found in the Windows directory under DCArtpresets.ini

Quiescent Point

The Quiescent point (or operating point) of an amplification device like an electron tube or a transistor, refers to the bias established on it’s linear portion of the transfer function curve when the device is “at rest” (ie. no signal input applied).  The Virtual Valve Amplifier allows you to adjust the Quiescent (operating) point of class A amplifiers anywhere from near cutoff to near saturation.

RAM Random Access Memory

A digital electronic device for storing binary information temporarily.  RAM performance is generally characterized in terms of its size in MBytes, and its access time in nanoseconds.  Your computer will need a minimum of 8 MBytes of RAM to run the Diamond Cut application correctly.

Real Time

A system which can process a signal and output the signal at the same rate at which it is being fed into the system is said to be a real-time processor.  The Diamond Cut algorithms can process signals in real-time or faster provided your platform is a 200 MHz Intel Pentium or higher.  The exception to this rule is the 200 MHz Intel Pentium-Pro processor.  Since it is not optimized for 16 bit applications, it cannot run all algorithms in real time or faster.

Real Time Analyzer (RTA)

A Real Time Analyzer is a form of spectrum analyzer used for the analysis of audio signals.  Unlike conventional spectrum analyzers, it does not use a single filter in a scanning mode to produce an amplitude vs. frequency display, which is a relatively slow process.  Instead, it processes audio signals in parallel, so that all frequency bands are displayed simultaneously.  Generally, RTA's have 31 bands (in 1 / 3 octave increments) covering the frequency spectrum from 20 Hz to 20 KHz.  They usually come with a calibrated electret microphone and a built-in pink noise generator for making acoustical measurements.

Rectified Voltage

A process wherein an alternating current signal is converted into direct current amplitude modulated envelope representation of the source. Often, some smoothing is applied to this signal with a set of time constants referred to as "attack" and "decay."  This signal is used in such devices as dynamic filters, companders, compressors, expanders, spectral enhancers, and is digitally simulated in some of the Diamond Cut algorithms.


The residue of a filtered signal is the algebraic difference between the filter output and its signal input.  Diamond Cut allows you to hear the “residue” of two of its filters by enabling the “Keep Residue” function.  The two filters that include this feature are the Continuous Noise Filter and the Harmonic Reject Filter.  This feature has been included because in some cases, it may be useful as an aid to hear what you are filtering out of the signal source.  This is particularly useful when adjusting the Harmonic Reject Filter when attempting to remove “Hum” or “Buzz” from a recording.  

Resistor / Resistance (R)(Ohms)

An basic electrical device which has electrical resistance, and is used to control the amount of current flow in a circuit.  The unit of measurement for a resistor is the ohm.

R = E / I  wherein, R = Resistance in ohms, E = Voltage in Volts, and I = Current in Amperes

Resistor Color Code

Standard RMA (Radio Manufacturers Association) Color Code:


Color                      Significant Figure           Decimal Multiplier


Silver                                      -                               0.01

Gold                                        -                               0.1

Black                                      0                              1.0

Brown                                    1                              10

Red                                         2                              100

Orange                                   3                              1,000

Yellow                                    4                              10,000

Green                                      5                              100,000

Blue                                        6                              1,000,000               

Violet                                      7                              10,000,000

Gray                                        8                              100,000,000

White                                     9                              1,000,000,000


The minimum amplitude increment into which the A-D converter of a discrete time system can divide an analog signal.  The resolution of Diamond Cut is usually 16 bits, which is 1 part in 65,536.  However, with the appropriate sound card, Diamond Cut does support up to with up to 24-bit I/O resolution.  Resolution can also refer to the minimum “time slice” into which a sampled data system is divided or displayed.

Reverse RIAA Curve 

Diamond Cut is equipped with a family of reverse RIAA curves, allowing you to use a standard RIAA phonograph pre-amplifier to perform your mastering of old acoustical and 78-RPM recordings.  A straight reverse RIAA curve is supplied for acoustical recordings, and a number of reverse RIAA curves with varying values of turnover frequency are supplied for electrically recorded 78’s.  These reverse curves can be found as several of the equalizer factory presets.


The process whereby the acoustical reflections of a room or concert hall are reproduced artificially, with devices such as tapped delay lines working in conjunction with mixing and phase shifting devices or algorithms.


Equalization Curve (Record Industry Association of America)

The RIAA Curve is an equalization frequency response contour which was utilized by manufacturers of LP records after around 1955.  It specifies three R*C time constants to be used by playback pre-amplifiers in order to invert the record cutter equalization.  The three time constants and their corresponding breakpoint frequencies are as follows:

1.       3180 uS  (50 Hz)

2.       318 uS  (500 Hz)  (turnover frequency)

3.       75 uS  (2120 Hz)  (rolloff frequency)

RIAA / IEC Equalization Curve

The RIAA / IEC equalization curve is defined in terms of the same time constants as the RIAA curve, with one additional time constant added of 7960 uS.  This provides 3 dB of attenuation at 20 Hz rolling off at -6 dB / Octave thereafter.  Below is a listing of all of the time constants associated with the RIAA / IEC Equalization Curve:

1.       7960 uS (20 Hz)

2.       3180 uS (50 Hz)

3.       318 uS (500 Hz)  (turnover frequency)

4.       75 uS (2120 Hz)  (rolloff frequency)

RMS Root Mean Squared

RMS is the square root of the average of the squared instantaneous values of a waveform taken over the waveforms time duration  (sometimes referred to as the "effective" value or the "heating" effect value).  In electrical terms, a.c. voltages and currents can be described in terms of their RMS value; in acoustical terms, sound pressure (acoustomotive force) can be described in terms of its RMS value.



For a low pass filter or for an equalization curve (such as the RIAA curve), the upper cutoff frequency is sometimes referred to as the Rolloff Frequency.

RPM (Revolutions Per Minute)

Some common record speeds are 33.33 RPM for LP's,  45 RPM for records with the same name, 78.26 RPM for most so called lateral 78's (like Victor), 78.8 RPM for Edison Lateral's, 80 RPM for Edison Diamond Discs, and 160 RPM for Edison Cylinder recordings.  Additional speeds such as 16 RPM will occasionally be encountered.

Here is a brief listing of some unusual speeds which may be encountered:

1.       White Wax Cylinders (1888 - 1892): 100 RPM

2.       Early Brown Wax Cylinders: 125 to 144 RPM

3.       Brown Wax Cylinders (1892 - 1899): 125 RPM

4.       Brown Wax Cylinders ("New Process" - 1900): 144 RPM

5.       Edison Concert Cylinders: 100 RPM

6.       Edison Gold Molded Cylinders: 160 RPM

7.       Pre 1900 Berliner Discs: 57 to 72 RPM

Fractional Speed (from 45 RPM) Change Speed ratio’s are as follows:

1.       45 RPM to 78.2 RPM - Use +73.7 % speed change

2.       45 RPM to 78.8 RPM - Use +75.1 % speed change

3.       45 RPM to 80 RPM - Use +77.1 % speed change


Low frequency noise, typically below 50 Hz which is often found on records.  This phenomenon can be caused by seismic effects during the mastering process or during playback.  On poor turntables or cutting lathes, it can also be produced by irregularities in the main thrust bearing race.  To attenuate turntable rumble using Diamond Cut, use the High Pass Filter.  Start with settings of 60 Hz and 18 dB / Octave, and adjust the frequency upwards or downwards until you are satisfied with the results.

Sample Rate

The rate at which an analog signal is converted to discrete numbers by an A-D converter.  For audio systems, sample rate is expressed in KHz.  Diamond Cut supports any number of standard sample rates including:

1.       KHz

2.       KHz

3.       44.1 KHz

4.       48.00 KHz

If your sound card supports intermediate sampling rates, you can also enter the numeric value of any sample rate you desire, between 8 KHz to 97 KHz for recording purposes.

Sampling Theorem

In a sampled data system (like the environment in which your Diamond Cut program is operating), Sampling Theorem tells us that regularly spaced sampling must occur at least at the Nyquist rate, which is twice the frequency of the highest frequency signal or noise component which is expected to be resolvable by the system (without aliases).  In other words, in a system expected to exhibit a frequency response up to 20 KHz, the minimum sample rate will have to be 40 KHz.  Because it is impossible to construct an ideal low pass filter, the sampling rate will have to be somewhat larger than 2X the desired maximum frequency response value.   In practice, a 44.1 KHz sampling rate is generally used in 20 KHz frequency response audio applications (although sometimes 48 KHz is also used).

Signal-to-Noise Ratio

The ratio of signal-to-noise (voltage, current, or acoustical sound pressure level) that is expressed in dB.  Signal-to-Noise ratio in dB = 20 log (signal  / noise).

Shielded Cables

Shielded cables are special cables which are designed to minimize stray noise fields (particularly E fields) from entering an audio system through the interconnection wiring from component to component due to extraneous sources.  Most often shielded cables are of the co-axial type so that loop area is also minimized, resulting in a minimization of  "H" field pickup.  However, some systems used a balanced pair of shielded wires which further minimizes pickup, provided the appropriate terminating transformers or differential amplifiers & line drivers are used on each end of the cable.


In the context of Diamond Cut and audio filter terminology, slope is the linear rate of change of amplitude vs. frequency of a filter past its corner frequency.  This is expressed in dB / Octave or dB / Decade.  6 dB / Octave = 20 dB / Decade, 12 dB / Octave = 40 dB / Decade, etc.

 Slot Filter

A slot filter is the compliment to the “notch” filter.  It is a variable narrow bandpass filter; capable of greater selectivity than a typical bandpass filter.  It is often used in Forensics work for isolating particular sounds like the ringing of a telephone on a recording in a crowded noisy bar situation, or anything similar.  By allowing only a very narrow “slot” of frequencies through the system, one can observe the “slotted-band” with a much improved signal to noise ratio compared to the wideband signal.  The Diamond Cut slot filter can be found under the Notch filter and is activated by checking the appropriate box.  Multiple slot filters can be run via the Multi-filter.  If the slots that are desired are harmonically related, you could use the Harmonic reject filter in “keep-residue” mode to produce up to 500 slots in one pass.


A Sone is a unit of measurement for sound loudness.  A simple tone of a frequency of 1 KHz and at a level 40 decibels above a listener's threshold of perception represents a loudness of 1 Sone.  A loudness of any sound which is judged by a listener to be n times greater that of the 1 Sone tone is n Sones.

Sound Level

Sound Level is a weighted sound pressure level obtained by the use of a metering system and any of three weighting standards as established in the American National Standard Specification for General Purpose Sound Level Meters.  The reference pressure is 2 X 10 ^ -5 Newton per meter ^2.  The two most common standards are the "A" and the "C" weighting factors.  The "A" weighting characteristic responds mostly to frequencies in the area of the greatest sensitivity of the human ear in the 500 to 10,000 Hz range.  The "C" weighting characteristic is nearly uniform over most of the audio spectrum.

The 0 dB reference sound pressure level (SPL) for a sound level meter is 0.0002 microbars using a simple tone of 1000 Hz.

The following is a list of some sound sources and their Acoustic Power and Sound Power Levels:

(Measured at 10 meters)

Sound Source                      Total Acoustic Power         Power Level

(A Weighted in Watts)             (dB re 10^-12 watts)  


Very soft Voice                     1 Nanowatt                            30 dB

Conversational Voice          10 Microwatts                       70 dB

Shouting Voice                     1 Milliwatt                             90 dB

Auto on Highway                10 Milliwatts                         100 dB

Blaring Radio                        100 Milliwatts                       110 dB

Piano                                      1.0 Watts                               120 dB

Small Aircraft Engine           3.0 Watts                               125 dB

Pipe Organ                            100 Watts                              140 dB

75 Piece Orchestra               100 Watts                              140 dB

4 Propeller Airplane             1000 Watts                            150 dB

Turbojet Engine                   10,000 Watts                         160 dB

Ram jet Engine                      100,000 Watts                      170dB     

Sound Wave Velocity

Sound Wave Velocity in air as a function of temperature is given by the following:

      c = 33,100 (1 + 0.00366t)^1/2


     c = Sound Wave Velocity in air in centimeters per second


      t = temperature in degrees centigrade

Therefore at 70 degrees C, sound will travel at 37,098.6 centimeters per second, or around 830 miles per hour.

Sound Wavelength

The Wavelength of a sound wave is given by the following equation:

     L = c / f


     L (lambda)  =  wavelength in centimeters


     c = Sound Wave Velocity


     f = frequency in Hz (cycles per second)


Speech Filter

A filter which typically has a bandpass only between the frequencies of 300 Hz to 3 KHz, and which is used to improve the basic intelligibility of speech.  Often, this type of filter uses slopes of -12 dB / Octave.

This characteristic can be replicated with the Bandpass filter.  An alternative speech filter that is sometimes useful is called the Steep Slope Speech filter.  Its response is 250 Hz to 3.5 KHz with a slope of 18 dB / Octave. 


A band or range of frequencies as in the audio spectrum, the light spectrum, or the electromagnetic spectrum.

Spectrum Analyzer

A device for analyzing and displaying the Amplitude versus Frequency characteristic of a portion of a spectrum.  They fall into two general categories:

1. Swept Bandpass Filter (a serial process of analysis)

2. Real Time Analyzer (a parallel process of analysis)

Spectral Enhancer

An electronic device which is used to expand the dynamic range of the upper and/or the lower octaves of the audio frequency spectrum, leaving the mid-band portion of the spectrum unprocessed.  This has the effect of increasing the "definition" of a recording without continuously amplifying hiss and rumble which may be present on the source material.  It is a form of dynamic filter which uses the principle of "upward expansion" to improve dynamic range.  The Dynamic Noise Filter contains a Spectral Enhancer mode of operation which can be enabled.


A device which indicates the RPM speed of a turntable by creating an optical illusion of the slowing-down, freezing, or speeding-up of a pattern when illuminated by a pulsating light source operating at a known frequency.  You can create your own stroboscope disc by dividing a circle evenly into black and white segments. Use the following formulae to calculate the number of segments required per 360 degrees (1 rotation of the disc) into which the disc must be marked:

60 Hz power systems:  # of segments = 7,200 / RPM*

50 Hz power systems: # of segments = 6,000 / RPM*

For example, assume that you want to construct a strobe for use in the United States where the power system operates at 60 Hz in frequency.  We want to design it “to freeze” at 78.2 RPM.  7,200 / 78.2 = 92.07.  Round the number to 92 segments.  Divide your circle into 92 evenly spaced segments, and voila, you have your strobe. Because of the rounding error, the strobe you constructed will be in error by 0.08 %.  Your strobe will have to be used under a flourescent or neon light connected to the power line in order to function.  Incandescent lamps will not work.

The following is a chart which you can use to create your own strobe using common line frequencies and RPM values:



# of Divisions for 50 Hz

# of Divisions for 60 Hz
















*Note: Actually, two pulses of light are produced per cycle of the line.  But, for improved visibility, it is better to use every other pulse to light up the strobe.

 Note: The Diamond Cut program provides two bitmaps which you can download and use as phonograph strobes covering the important speeds.


The following is a listing of some of the more common record types and the styli that they require:

A.      Modern LP's:    0.7 mil elliptical

B.      Early LP's:   1.5 mil truncated elliptical

C.      Transcription Recordings:   2.3 mil truncated elliptical

D.      Narrow Groove 78's such as Polydor:   2.4 mil truncated elliptical

E.       Late 1930's Lateral 78 RPM Discs:   2.8 mil truncated elliptical

F.       Standard Groove 78 RPM Discs:   3.0 mil truncated elliptical

G.      Pre-1935 Lateral Cut Electrical 78's:   3.3 mil truncated elliptical

H.      1931 to 1935 RCA Pre-Grooved Home Recordings:   5.0 mil spherical

I.         Edison 80 RPM Diamond Discs:   3.7 mil spherical or non-truncated conical

J.       Edison Blue Amberol Cylinders: 3.7 to 4.2 mil non-truncated spherical

K.      Wide Groove Acoustical 78 Lateral Disc:   3.8 mil truncated elliptical

L.       Edison Wax Amberol Cylinders:   4.2 mil Spherical

M.     Edison White Wax, Brown Wax, Concert, and Gold Molded Cylinders:   7.4 mil Spherical

N.      Pathe 78’s: 3.7 mil truncated conical

O.      Metal Stampers: Biradial of appropriate dimensions *

P.       Late 16 inch transcription discs: 2.0 mil truncated elliptical

Q.      Very early acoustical lateral cut discs: 4.0 truncated elliptical

R.      Etched-label Pathes up to 14 inches in diameter: 8.0 mil spherical

S.       Etched-label Pathes greater than 14 inches in diameter: 16.0 mil spherical

T.       Acetate and aluminum “instantaneous” discs: 6.0 mil elliptical or truncated elliptical

* Note: When stampers are played on a normal turntable equipped with a Biradial stylus, you will need to use the File Reversal feature so that it can be converted to forward play.

 Tape Recorder Speeds


Time Constant

Time constants are exponential amplitude vs. time functions, which are realized with resistors and capacitors, or resistors and inductors.  Tau = R x C   or  Tau = L / R  wherein Tau = time constant in seconds,  R = resistance in Ohms,  C = capacitance in Farads, and L is inductance in Henries.

 The relationship between a simple first order filters corner frequency (Fc) and time constant is as follows:

Fc = 1 / (2 x pi x Tau)

 Note that the higher the value of time constant, the lower the corner frequency created.  Some common time constants found in audio applications are as follows:

25 uSec – Dolby based FM de-emphasis

70 uSec – Type 1 (Normal Bias) Cassette Tape Eq

75 uSec – Standard FM Broadcast de-emphasis

120 uSec – Type 2 (High Bias) Cassette Tape Eq

 Additional audio time constants can be found under RIAA and NAB in this glossary.

Time Derivative

The instantaneous rate of change of a parameter (such as voltage amplitude or sound pressure level) with respect to time.    (i.e. dV / dt,  dP / dt, etc.)


An alternating current device used to impedance match transducers and electronic circuits to one another.  Sometimes, these devices are used with a unity turns ratio to provide isolation from one circuit to another rather than to impedance match the two.  This is useful in audio applications when it is necessary to break a ground loop source of noise in a system.


A Triode is an electron tube (or valve) containing three elements.  They consist of an anode, cathode, and a control grid.  Small changes in grid voltage produce large changes in values of current in the plate circuit (the ratio of delta plate current to delta grid voltage is its gain in transconductance or mu.)  They are most commonly used in audio pre-amplifier, and other low-level applications.  Typical triodes found in audio applications include the 12AX7 and 6SL7 high mu (gain), and the 12AU7 and 6SN7 medium mu devices.  All of the devices listed are “dual” (two in one envelope).


See Electron Tube.

Turnover Frequency

The frequency in a phonograph equalization curve below which the master was recorded with the cutting head operating in constant displacement mode rather than in constant velocity mode.  This is used to limit the excursions of the cutting stylus so that bass notes do not cause the cutting stylus to break through to the adjoining groove wall.

Here is a listing of the most common turnover frequencies utilized by brand and vintage:

 200 Hz:                   Columbia (1925 - 1937)

                                Victor (1925 - 1937)

 250 Hz:   Decca (1935 - 1949)


                                English Columbia

 300 Hz:   Columbia (1938 - End)

 400 Hz:                   Capitol


500 Hz:   Brunswick

                                Decca (1925 - 1929)

                                Edison Laterals (1929)



                                Victor (1938 - 1952)


The British term for an electron tube.  It arises out of the valve like effect that a grid has on the flow of electrons between the devices cathode and anode.  Also, refer to Electron Tube.

Vector Quantity

Any physical quantity, like the displacement of a record stylus, whose specification involves both magnitude and direction and which obeys the parallelogram law of addition.

Vertical Cut

(Hill and Dale)

A record recording technique in which the groove modulation (undulations) occur in an up-and-down direction as opposed to side-to-side.  This technique was used by Thomas Edison in his original invention of the phonograph, and was maintained as the recording method used by his companies cylinders and Diamond Discs.

Volt (V)


The unit of measurement of electrical potential difference (or electromotive force) equal to the difference in potential which occurs in a conductor which is carrying 1 ampere, and the power being dissipated in the conductor is 1 watt, with the resistance of the conductor being 1 ohm.


A measurement unit of electrical power equal to the ability to do work at the rate of 1 Joule per second.

P = V x I  wherein P = power in watts, V =  voltage in volts, and I = current in amperes.

Wave file

The sound file format that Diamond Cut supports.  This (.wav) is the standard Windows file format.


The signal output of a special effects generator (such as the Reverb) which contains the modified (processed) signal. “Wet” refers to the effects signal alone.  The non-processed signal from such a generator is referred to as “dry.” As with most special effect generators, the Reverb has an output mix control which allows you to transfer a signal from the effects generator, which ranges from completely dry to completely wet (no source signal), or to some mixture in between.

White Noise

White Noise is random noise that is characterized as containing equal energy per unit frequency (Hertz).  White Noise is sometimes referred to as Johnson, shot, or thermal noise.  White noise derives its name from the analogous definition of white light.  Audio white noise can be created using the Make Waves function.

Window Weighting

Window weighting is a concept which pertains to systems which involve fast Fourier transforms (FFT’s.)  Signals, which are observed for finite intervals of time, may contain distorted spectral data in the transform due to the ringing of the Sin(f)/f spectral peaks of a rectangular window.  This distortion is minimized by the use of a window-weighting function, which is applied before the DFT is performed.  The window weighting functions used in the FFT based Diamond Cut algorithms is proprietary.

Wire Table

Standard Annealed Copper

Gauge in AWG

Diameter in Mils  

Resistance per Foot *

(@ 20 degrees C)


























































* Ohms

 Note 1: The temperature coefficient of resistance for copper wire  =  + 0.4% / degree C

Note 2: The resistance of a 2 conductor cable will be have to be doubled to account for the round trip.


A slow periodic change in the pitch or low frequency flutter which may be present on phonograph, tape, or soundtrack recordings due to a non uniform velocity of the recording medium.  Wow is generally a frequency modulating effect that occurs at a deviation rate between 0.5 to 6 Hz.  The Wow could have been introduced in the recording process, the playback process, or a combination of both.  Wow found on record recordings is usually caused by a non-concentric spindle hole.  Wow found on tape recordings is generally caused by warped take-up or supply reels.  Diamond Cut is not capable of correcting audio problems of this nature at this point in time. 

Wow and Flutter

Wow and flutter is the combined FM effect of both mentioned parameters.  The frequency spectrum in which this rate of frequency deviation is made is in the spectrum that exists between 0.5 to 250 Hz.


This is the horizontal axis of a graph.  In Diamond Cut, it contains the time information for your wave file that is divided up into ten equally spaced grids.


This is the vertical axis of a graph.  In Diamond Cut, it contains the amplitude information for your wave file that is divided up into four equally spaced grids.


Popular Products

Personal Forensics Investigation

$99 Evaluation

Maybe you have a cell phone recording, maybe it was a digital hear muffled voices in the background. What are they saying? Not knowing has been eating at you and you need answers. You've seen advanced forensics tools on television, but you know nothing about audio and even less about clarifying a spoken voice on a recording.



This new flagship product offers features that any restoration enthusiast would love with no fat.  We don't try to be all things to all people, but if you want the world's best audio restoration...stop lookin.

DC 8 Software Training DVD


Our resident software guru comes into your home via our DVD and walks you through tips and tricks for maximizing the quality and speed of your restoration results.  You'll be amazed at what you can learn in 2 hours.

Better Audio Restorations Training DVD

This detailed introduction to the Zen of Audio Restoration features a beginner's guide to restoration software.  Make the most of your restoration software


DC LIVE/Forensics

There is simply no better Audio Forensics product on the market.  You can pay thousands more, but if you're clarifying noisy recordings or even running live audio surveillance, DC LIVE/Forensics is the best in the business.  $1499

Advanced Forensics Concepts Using DC LIVE/Forensics DVD

The in depth walkthrough covers all of the great new features of DC/LIVE/Forensics 7.5.  $59

Audio Mentor

New Price $29!

This intelligent wizard helps any user instantly achieve spectacular results.  Mentor takes you by the hand and gets you from the Record or Tape to CD in several painless steps. 

Learning The CNF DVD

Go "under the hood" of the continuous noise filter.  This in depth training course will help any level user achieve better results with this powerful filter.



Computer Transfer Preamps

New thinking for your vinyl restoration.  These preamps give you the most accurate reproduction off of your record that is a price you won't believe


Automatic Forensics Adaptive Filter VST Plug-In

Now you can get one of DC LIVE/Forensics most powerful automatic adaptive filters and plug it into the software of your choice.  This cool new VST plug automatically removes noise so that you can find a voice buried beneath the muck.  Only $399

Virtual Valve Tube Simulation VST Plug-In

Add real tube sound to any music file an instantly add the warmth and depth that only real tubes can provide.  This new VST Plug In is useful in so many ways that you'll instantly find it your favorite plug-in.  Only $99

DC 8 Software Training DVD


Our resident software guru comes into your home via our DVD and walks you through tips and tricks for maximizing the quality and speed of your restoration results.  You'll be amazed at what you can learn in 2 hours.

Better Audio Restorations Training DVD

This detailed introduction to the Zen of Audio Restoration features a beginner's guide to restoration software.  Make the most of your restoration software


DC LIVE/Forensics

There is simply no better Audio Forensics product on the market.  You can pay thousands more, but if you're clarifying noisy recordings or even running live audio surveillance, DC LIVE/Forensics is the best in the business.  $1499

Advanced Forensics Concepts Using DC LIVE/Forensics DVD

The in depth walkthrough covers all of the great new features of DC/LIVE/Forensics 7.5.  $59

Audio Mentor

New Price $29!

This intelligent wizard helps any user instantly achieve spectacular results.  Mentor takes you by the hand and gets you from the Record or Tape to CD in several painless steps. Only $59

Learning The CNF DVD

Go "under the hood" of the continuous noise filter.  This in depth training course will help any level user achieve better results with this powerful filter.

Diamond Cut Millennium

Powerful Audio Restoration and enhancement product.  Only $59



Computer Transfer Preamps

New thinking for your vinyl restoration.  These preamps give you the most accurate reproduction off of your record that is a price you won't believe

Automatic Forensics Adaptive Filter VST Plug-In

Now you can get one of DC LIVE/Forensics most powerful automatic adaptive filters and plug it into the software of your choice.  This cool new VST plug automatically removes noise so that you can find a voice buried beneath the muck.  Only $399

Virtual Valve Tube Simulation VST Plug-In

Add real tube sound to any music file an instantly add the warmth and depth that only real tubes can provide.  This new VST Plug In is useful in so many ways that you'll instantly find it your favorite plug-in.  Only $99

Call Us Toll Free At 866 260 6376 3600 Board Rd.  York, PA  17402