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.
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.
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
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
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.
(Farad) and uF
(micro Farad or 1 X 10 ^ -6 Farad)
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):
2. High (IEC Type 2) Usually Chromium Oxide based):
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*:
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.
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.
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.
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.
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
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
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.
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.
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
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
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:
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 =
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.
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
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.
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.
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.
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
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:
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.
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.
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
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.
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.
The shape of the tip of certain phonograph record playing
styli which improves the high frequency response as compared to standard conical
The average amplitude of a Wave file as displayed in the
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.
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
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
It allows severely warped records to be transferred without skipping.
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.
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.
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.)
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:
= Vout / Vin)
--- (Voltage Gain in dB = 20 log Vout / Vin)
= Iout / Iin)
--- (Current Gain in dB = 20
log Iout / Iin)
= Pout / Pin)
--- (Power Gain in dB = 10
log Pout / Pin)
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
only future conversion back to the analog world would occur during the playback
process of the CD or the DAT through your audio system.
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:
Hz, 62 Hz, 125 Hz, 250 Hz, 500 Hz, 1 KHz, 2 KHz, 4 KHz, 8 KHz, 16 KHz.
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.
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 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.
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
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.
See Vertical Cut.
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.
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
Z = Impedance in ohms
R = Resistance
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 ohm - The basic unit of measurement for Electrical Resistance,
Impedance, or Reactance
2 ohms (sound re-enforcement systems), 3.2 ohms (antique audio),
4, 8, 16, and 32,
ohms - Standard Loudspeaker
(8 ohms is the most common in
1996 in the United States).
50 ohms - Standard Unbalanced Co-Axial impedance for RF signal
75 ohms - Standard Unbalanced Co-Axial impedance for Television and FM
300 ohms - Standard Balanced impedance for Television and FM signal
377 ohms - Impedance of Free Space
500 ohms - Standard Balanced Microphone impedance
600 ohms - Standard Telephone Exchange Audio line impedance
2,000 ohms - Antique Audio (headphones & 1920's vintage horn
20,000 ohms - Common single ended input impedance found on Professional
47,000 ohms - Common Magnetic Phono Cartridge Loading Impedance
50,000 ohms - Standard Unbalanced High Impedance Microphone Impedance.
100,000 ohms - Common Input Impedance on Audio Equipment
1 M. ohms - De-Facto Standard, Oscilloscope Input Impedance
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
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
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.
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
* This speed is also used by reel to reel analog data
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - -
Rotary Head Tape Recorder Speeds:
0.321 ips (8.15 mm / sec)
1.31 ips - SP (Standard Play)
0.66 ips - LP (Long Play)
0.44 ips - EP (Extended Play)
0.797 ips (2.0 cm / sec) - Beta
0.524 ips (1.33 cm / sec) - Beta
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.
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.
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.
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.
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
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.
Variable Reluctance (early magnetic cartridges)
Moving Magnet (the most commonly used)
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)
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 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.
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
The following table provides the frequencies of two Octaves
of the tempered scale (1/2 step between notes) rounded in integers:
(or D flat)
(or E flat)
(or G flat)
(or G flat)
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.
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.
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.
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
allow detail to be seen in a signal when the detail exists towards the top or
bottom of the signal workspace display area.
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.
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
A variable electronic filter in which the following three
parameters may be adjusted on each parametric channel:
Level (attenuation or amplification)
Parametric equalizers are usually equipped with several
parametric channels which can all be used simultaneously or each one can be
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 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
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 =
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
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.)
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
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
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
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
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.
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.
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
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.
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
Standard RMA (Radio Manufacturers Association) Color Code:
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
usually 16 bits, which is 1 part in 65,536.
However, with the appropriate sound card, Diamond Cut
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
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
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:
3180 uS (50 Hz)
318 uS (500 Hz) (turnover frequency)
75 uS (2120 Hz) (rolloff frequency)
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:
7960 uS (20 Hz)
3180 uS (50 Hz)
318 uS (500 Hz) (turnover
75 uS (2120 Hz) (rolloff
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
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
Early Brown Wax Cylinders: 125 to 144 RPM
Brown Wax Cylinders (1892 - 1899): 125 RPM
Brown Wax Cylinders ("New Process" - 1900): 144 RPM
Edison Concert Cylinders: 100 RPM
Edison Gold Molded Cylinders: 160 RPM
Pre 1900 Berliner Discs: 57 to 72 RPM
Speed (from 45 RPM) Change Speed ratio’s are as follows:
45 RPM to 78.2 RPM - Use +73.7 % speed change
45 RPM to 78.8 RPM - Use +75.1 % speed change
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.
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
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.
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
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
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.
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
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
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)
Total Acoustic Power
Weighted in Watts)
(dB re 10^-12 watts)
Very soft Voice
Auto on Highway
Small Aircraft Engine
75 Piece Orchestra
4 Propeller Airplane
Ram jet Engine
Sound Wave Velocity in air as a function of temperature is
given by the following:
c = 33,100 (1 + 0.00366t)^1/2
= 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.
The Wavelength of a sound wave is given by the following
L = c / f
= wavelength in centimeters
= Sound Wave Velocity
= frequency in Hz (cycles per second)
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.
A device for analyzing and displaying the Amplitude versus
Frequency characteristic of a portion of a spectrum. They fall into two general categories:
Bandpass Filter (a serial process of analysis)
Analyzer (a parallel process of analysis)
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: