Download Music Technology Terms

Music Technology Terms
These definitions are not exhaustive by any stretch, but they will give
you the most commonly used meaning of these words for music
educators. If you know that something I have stated here is not
necessarily 100% correct, no offence, but this glossary is not
intended for your level of technical knowledge.
Amplifier
A hardware box which takes an audio signal, makes it louder, and let's you run
the signal through speakers. A guitar amp is one - in this case, the amplifier
and the speaker are combined in one box. Many mixers are said to be
powered, meaning they have a power amp built in. How can you tell? If the
back of the mixer has jacks where you can plug the speakers in, then it's
powered.
Audio Cable
The cables which connect the audio output of a synthesizer, guitar, etc. to the
input of an amplifier, PA system, mixer, etc.
Continuous Controller
Refers to Midi information other than notes. For example, volume, pitch
bending, modulation (vibrato). These parameters can change continuously
over time and allow electronically generated music to sound more expressive.
Controller
A device which let's you enter or change events into a computer or other
digital device. Examples: synthesizer, wind controller, mouse, computer
keyboard.
CPU
Acronym for Central Processing Unit, which is the main 'guts' of your
computer. It houses the memory, the internal hard drive, the disk drive, the
ports in the back to plug into, etc.
Digital Signal Processor
See Signal Processors. Most newer signal processors are digital. These usually
have computer memory built in which allows the instant recall of all the
parameter settings of the device. This allows you to quickly change the
devices function and sound without tweaking with the controls.
Drum Controller
A drum-like device which has a flat striking area which is usually separated
into different areas, each corresponding to a different sound. Some drum
controllers have sounds built into them, others are just triggering devices
which need to be hooked to a synthesizer or tone module.
Drum Machine
A smallish box with buttons on the front which correspond to drum and
percussion sounds stored in it's computer memory. Drum machines have small
sequencers in them which allow you to record musical patterns into it's
memory.
Hard Drive
A memory storage device now common to all computers. It's memory is stated
in megabytes (Mb). The analogy I like to use for people confused about
memory is this: Picturing an office scenario, your computer's main memory is
like the size of the top of your desk, i.e. how many projects you can have
going at once is directly proportional to how big your desk top is. Your hard
drive is like the file cabinet next to your desk - you can store a lot of
information there, but when you want to work on a project, you have to pull it
out of your file cabinet and put it on top of your desk.
Hardware Sequencer
A sequencer (see sequencer) which is built into a dedicated box for only this
purpose. It does not serve other functions (compare to a computer-based, or
software sequencer - the computer can become a completely different device
with new software). Some synthesizers have small hardware sequencers built
in. Hardware sequencers have the advantage of being more portable.
Junction Strip
Common box which plugs into an AC outlet and provides 4-6 extension
outlets. Also called a power strip. power strip - common box which plugs into
an AC outlet and provides 4-6 extension outlets. Also called a junction strip.
Keyboard
Usually refers to the synthesizer's piano keyboard, though it can also be the
computer keyboard.
MIDI
An acronym which stands for Musical Instrument Digital Interface. It is
merely a standard language that let's instruments, devices, software and
hardware from different manufacturers communicate fluently.
MIDI Cable
The cables which connect Midi devices together. Recognized by looking into
the end and having the 5 pins form a smile.
Mixer
A hardware box which combines different audio signals together and outputs
them in mono or stereo. Mixers come in many sizes and are referred to by the
number of channels (different audio inputs) they have. Most software
sequencers also have a mixer onboard which lets you control the volume
levels of the individual parts of your song.
Modulation
A continuous controller which can be applied to synthesized note(s), usually
from a joystick to the left of the lowest keyboard note. The sound is usually a
vibrato which gets applied more as you move the joystick farther forward.
Module
Same as tone module or sound module or tone generator. See tone module.
Monitor
The computer's screen.
Pan
Refers to moving an audio signal left or right in the stereo spectrum. Also
called the balance control. All stereo audio mixers have panning, and most
software sequencers allow you to set and change panning.
Patch
A synthesizer sound which is stored in it's computer memory. Usually refers to
a sound which can be altered, i.e. it's stored in RAM memory. Sometimes also
called preset, program, or sound.
Percussion Controller
Same as a drum controller, except that there are many percussion controllers
which are configured like mallet instruments and thus are very adept at
playing pitched parts.
Pitch Bend
A continuous controller which can be applied to synthesized note(s), usually
from a joystick to the left of the lowest keyboard note. The sound is a raising
or lowering of the pitch and changes as you move the joystick left and right.
Preset
A synthesizer sound which is stored in it's computer memory. Usually refers to
a sound which can not be altered, i.e. it's stored in ROM memory. Sometimes
also called patch, program, or sound.
Quantize
A process used in sequencing which 'fixes' rhythmic inaccuracies in a musical
track. Quantizing 'rounds off' musical notes to the nearest eighth or sixteenth
note (choose any note value) with respect to the sequencer's metronome.
RAM
Acronym for Random Access Memory. This is computer memory which can
be used over and over. A floppy disk is RAM memory because you can erase
and reuse it. This also refers to the amount of main memory your computer
has and is always stated in megabytes (Mb).
ROM
Acronym for Read Only Memory. This is computer memory which can't be
changed or erased. It is 'burned' into the computer or device. Most synthesizers
have some sounds which are in ROM memory and can't be altered. A sign of a
more expensive synthesizer is having sounds in RAM memory, implying that
you can alter the sounds and save variations as your own.
Sampler
Also called a digital sampler. A type of synthesizer which derives it's sounds
from recording actual sounds (instruments or non musical sounds) and then
storing them in computer memory, either floppy discs, hard drive, or recorded
onto CD-ROM. They are used extensively for generating sound effects.
SCSI
Acronym for Small Computer Serial Interface, which is a connection on the
back of your computer which allows connection to other hardware devices
such as external CD-ROM drives, external hard drives, some printers,
scanners, etc.
Signal Processor
An electronic device which audio signals can be routed through to affect the
sound of that signal. Examples: echo and reverb units, distortion devices, etc.
Most electric guitarists run their instruments through 'pedals' which are small
floor units process signals at the press of a foot pedal.
Sequencer
A piece of hardware or software which allows you to record multiple tracks of
music one on top of the other while listening to what was recorded previously.
Software Sequencer
a sequencing software package designed to be loaded into a computer.
Software sequencers usually have more features and have the advantage of
showing you a lot more information at once because of it's computer screen.
Sound Module
Same as, a tone module, module, or tone generator. See tone module.
Synthesizer
A piano keyboard with computer memory onboard which stores a bank of
different sounds, many of which will be traditional instrument sounds.
Tone Generator
Same as sound module, tone module, or module. See tone module.
Tone Module
A synthesizer without a piano keyboard. Since Midi allows one keyboard to
literally play another, there is little reason to acquire more piano keyboards
when wanting to expand your palette of sound choices. Buying tone modules
is usually a bit cheaper than the keyboard version, and saves valuable space.
Track
Sequencers borrowed this term from multi-track recording studios, referring to
tape tracks. A track is one of a number of locations where a musical part can
be recorded and played back. A typical software sequencer has 16-128 tracks.
Wind Controller
A controller 'instrument' which is woodwind-like or brass-like in it's fingering.
They are blown into and the air stream triggers sounds from a synthesizer or
tone module. Many do not have sounds of their own and must be connected
(through Midi) to a synthesizer or tone module. They will play whatever
sound is called up on the connected synthesizer.
Wind Synthesizer
Same as wind controller.
Quantization (signal processing)
From Wikipedia, the free encyclopedia
Jump to: navigation, search
Quantized signal
Digital signal
In digital signal processing, quantization is the process of approximating a
continuous range of values (or a very large set of possible discrete values) by
a relatively-small set of discrete symbols or integer values. More specifically,
a signal can be multi-dimensional and quantization need not be applied to all
dimensions. A discrete signal need not necessarily be quantized (a pedantic
point, but true nonetheless and can be a point of confusion). See ideal
sampler.
A common use of quantization is in the conversion of a discrete signal (a
sampled continuous signal) into a digital signal by quantizing. Both of these
steps (sampling and quantizing) are performed in analog-to-digital converters
with the quantization level specified in bits. A specific example would be
compact disc (CD) audio which is sampled at 44,100 Hz and quantized with
16 bits (2 bytes) which can be one of 65,536 (216) possible values per sample.
Dither
From Wikipedia, the free encyclopedia
Jump to: navigation, search
This article refers to dither and dithering as a technical term in digital signal
processing, and the fields audio; video; and digital imaging. In this context,
dither is only distantly related to the common dictionary definition of "dither"
as "a state of indecision" or "to be nervously irresolute in acting or doing."
Dither is a form of noise, or 'erroneous' signal or data which is added to
sample data for the purpose of minimizing quantization error. Dither is
routinely used in processing of both digital audio and digital video data. Jitter
is not synonymous with dither.
Dither Tricks
By Bob Masta
Interstellar Research
As we saw last time, dither can increase analog-to-digital converter (ADC)
resolution for repetitive inputs. It does this by adding a small offset to the input
signal, which is varied over a range at least equal to the least significant bit
(LSB) of the converter. The LSB will toggle on or off in proportion to the
amount of time the sum is above or below its threshold. By synchronously
averaging many repetitions, the value of the input signal may be determined
with a resolution proportional to the number of possible dither values and/or
number of repetitions averaged.
Normally, the dither comes from a broadband noise source, although some
input signals may have enough inherent noise to provide self-dither. Either
way, the resultant input is noisy. Although synchronous waveform averaging
will reduce that noise at the same time it increases the resolution, the noise
doesn't go down as fast as the resolution goes up: Doubling the number of
repetitions averaged will double the resolution, but it only reduces the noise
by 3 dB. So if noise is added to increase resolution, it will also increase the
noise in the final result.
But that assumes that the dither noise is uncorrelated with the averaging
process. Under some circumstances it's possible to beat the odds and
increase resolution without any noise penalty. The simplest conceptual
approach, which is rarely used, is DC dither.
This method requires that the DC level be changed between repetitions of the
input wave. The easiest way to achieve this is to have the averager increment
a digital-to-analog converter ( DAC ) value after each input repetition, before it
begins waiting for the next repetition sync. Before the DAC output is added to
the input, it must be attenuated such that its total stepped range will traverse
only +/- 1/2 LSB of the ADC during the desired number of repetitions. Figure 1
shows a typical setup:
Fig. 1: DC Dither Setup
Digital Anti-Aliasing Filter
According to the Sampling Theorem, any signal can be accurately
reconstructed from values sampled at uniform intervals as long as it is
sampled at a rate at least twice the highest frequency present in the signal.
Failure to satisfy this requirement will result in aliasing of higher-frequency
components, meaning that these components will appear to have frequencies
lower than their true values.
One way of avoiding the problem of aliasing is to apply a low-pass filter to the
signal, prior to the sampling stage, to remove any frequency components
above the "folding" or Nyquist frequency (half the sampling frequency). Such
anti-aliasing filters are commonly built into the analog interface chips and
codecs which convert analog input signals into digital form for processing by a
digital signal processor (DSP). In many cases, these anti-aliasing filters are
implemented using conventional analog circuitry. An alternative method is to
use a digital anti-aliasing filter. This avoids the noise and drift problems
inherent in analog filter circuits, and is the natural approach when the signal is
going to be processed digitally anyway.
How can a digital filter be used to remove frequency components above the
folding frequency, which is the highest frequency that can be handled by a
sampled data system? The answer lies in the application of a technique called
multirate digital signal processing, where different sampling rates are used at
different stages in the processing of a signal. The anti-aliasing filter used in
the FFT spectrum analyser applet (Java 1.1 version) works by "oversampling"
the input signal at a rate of 48 kHz - six times the 8000 Hz sampling rate used
by the spectrum analyser itself. Operating at the higher sampling rate, the
anti-aliasing filter is a low-pass filter with a passband from 0 to 4000 Hz (the
folding frequency). The output from the filter is "downsampled" by a factor of
six, simply by throwing away five samples in every six, giving a stream of
sampled values effectively at the lower sampling rate of 8000 Hz.
Filter specification
The 204-tap FIR anti-aliasing filter was designed using MATLAB, based on
the Parks-McClellan method.
The filter specification is as follows:





48 kHz sampling rate
0 - 4 kHz passband
0.1 dB passband ripple
40 dB stopband attenuation
500 Hz transition bandwidth
The calculated frequency response of the filter is shown in the diagram.
Basics of Digital Recording
CONVERTING SOUND INTO NUMBERS
In a digital recording system, sound is stored and manipulated as a stream of
discrete numbers, each number representing the air pressure at a particular
time. The numbers are generated by a microphone connected to a circuit
called an ANALOG TO DIGITAL CONVERTER, or ADC. Each number is
called a SAMPLE, and the number of samples taken per second is the
SAMPLE RATE. Ultimately, the numbers will be converted back into sound by
a DIGITAL TO ANALOG CONVERTER or DAC, connected to a loudspeaker.
Fig. 1 The digital signal chain
Figure 1 shows the components of a digital system. Notice that the output of
the ADC and the input of the DAC consists of a bundle of wires. These wires
carry the numbers that are the result of the analog to digital conversion. The
numbers are in the binary number system in which only two characters are
used, 1 and 0. (The circuitry is actually built around switches which are either
on or off.) The value of a character depends on its place in the number, just
as in the familiar decimal system. Here are a few equivalents:
BINARY DECIMAL
0=0
1=1
10=2
11=3
100=4
1111=15
1111111111111111=65535
Each digit in a number is called a BIT, so that last number is sixteen bits long
in its binary form. If we wrote the second number as 0000000000000001, it
would be sixteen bits long and have a value of 1.
Word Size
The number of bits in the number has a direct bearing on the fidelity of the
signal. Figure 2 illustrates how this works. The number of possible voltage
levels at the output is simply the number of values that may be represented by
the largest possible number (no "in between" values are allowed). If there
were only one bit in the number, the ultimate output would be a pulse wave
with a fixed amplitude and more or less the frequency of the input signal. If
there are more bits in the number the waveform is more accurately traced,
because each added bit doubles the number of possible values. The distortion
is roughly the percentage that the least significant bit represents out of the
average value. Distortion in digital systems increases as signal levels
decrease, which is the opposite of the behavior of analog systems.
Fig. 2 Effect of word size
The number of bits in the number also determines the dynamic range. Moving
a binary number one space to the left multiplies the value by two (just as
moving a decimal number one space to the left multiplies the value by ten), so
each bit doubles the voltage that may be represented. Doubling the voltage
increases the power available by 6 dB, so we can see the dynamic range
available is about the number of bits times 6 dB.
Sample Rate
The rate at which the numbers are generated is even more important than the
number of bits used. Figure 3. illustrates this. If the sampling rate is lower than
the frequency we are trying to capture, entire cycles will be missed, and the
decoded result would be too low in frequency and might not resemble the
proper waveform at all. This kind of mistake is called aliasing. If the sampling
rate were exactly the frequency of the input, the result would be a straight line,
because the same spot on the waveform would be measured each time. This
can happen even if the sampling rate is twice the frequency of the input if the
input is a sine or similar waveform. The sampling rate must be greater than
twice the frequency measured for accurate results. (The mathematical
statement of this is the Nyquist Theorem.) This implies that if we are dealing
with sound, we should sample at least 40,000 times per second.
Fig. 3 Effects of low sample rates
The Nyquist rate (twice the frequency of interest) is the lowest allowable
sampling rate. For best results, sampling rates twice or four times this should
be used. Figure 4 shows how the waveform improves as the sampling rate is
increased.
Fig. 4 Effect of increasing sample rate
Even at high sample rates, the output of the system is a series of steps. A
Fourier analysis of this would show that everything belonging in the signal
would be there along with a healthy dose of the sampling rate and its
harmonics. The extra junk must be removed with a low pass filter that cuts off
a little higher than the highest desired frequency. (An identical filter should be
placed before the ADC to prevent aliasing of any unsuspected ultrasonic
content, such as radio frequency interference.)
If the sampling rate is only twice the frequency of interest, the filters must
have a very steep characteristic to allow proper frequency response and
satisfactorily reject the sampling clock. Such filters are difficult and expensive
to build. Many systems now use a very high sample rate at the output in order
to simplify the filters. The extra samples needed to produce a super high rate
are interpolated from the recorded samples.
By the way, the circuits that generate the sample rate must be exceedingly
accurate. Any difference between the sample rate used for recording and the
rate used at playback will change the pitch of the music, just like an off speed
analog tape. Also, any unsteadiness or jitter in the sample clock will distort the
signal as it is being converted from or to analog form.
Recording Digital Data
Once the waveform is faithfully transformed into bits, it is not easy to record.
The major problem is finding a scheme that will record the bits fast enough. If
we sample at 44,100 hz, with a sixteen bit word size, in stereo, we have to
accommodate 1,411,200 bits per second. This seems like a lot, but it is within
the capabilities of techniques developed for video recording. (In fact, the first
digital audio systems were built around VCRs. 44.1 khz was chosen as a
sample rate because it worked well with them.)
To record on tape, a very high speed is required to keep the wavelength of a
bit at manageable dimensions. This is accomplished by moving the head as
well as the tape, resulting in a series of short tracks across the tape at a
diagonal.
On a Compact Disc, the bits are microscopic pits burned into the plastic by a
laser.The stream of pits spirals just like the groove on a record, but is played
from the inside out.To read the data, light from a gentler laser is reflected off
the surface of the plastic (from the back: the plastic is clear.) into a light
detector. The pits disrupt this reflection and yield up the data.
In either case, the process is helped by avoiding numbers that are hard to
detect, like 00001000. That example is difficult because it will give just a
single very short electrical spike. If some numbers are unusable, a larger
maximum (more bits) must be available to allow recording the entire set. On
tape, twenty bits are used to record each sixteen bit sample, on CDs, twentyeight bits are used.
Error Correction
Even with these techniques, the bits are going to be physically very small, and
it must be assumed that some will be lost in the process. A single bit can be
very important (suppose it represents the sign of a large number!), so there
has to be a way of recovering lost data. Error correction is really two
problems; how to detect an error, and what to do about it.
Fig. 5 Effects of data errors
The most common error detection method is parity computation. An extra bit
is added to each number which indicates whether the number is even or odd.
When the data is read off the tape, if the parity bit is inappropriate, something
has gone wrong. This works well enough for telephone conversations and the
like, but does not detect serious errors very well.
In digital recording, large chunks of data are often wiped out by a tape dropout
or a scratch on the disk. Catching these problems with parity would be a
matter of luck. To help deal with large scale data loss, some mathematical
computation is run on the numbers, and the result is merged with the data
from time to time. This is known as a Cyclical Redundancy Check Code or
CRCC. If a mistake turns up in this number, an error has occurred since the
last correct CRCC was received.
Once an error is detected, the system must deal gracefully with the problem.
To make this possible, the data is recorded in a complex order. Instead of
word two following word one, as you might expect, the data is interleaved,
following a pattern like:
words 1,5,9,13,17,21,25,29,2,6,10,14,18,22,26,30,3,7,15,19,27 etc.
With this scheme, you could lose eight words, but they would represent
several isolated parts of the data stream, rather than a large continuous chunk
of waveform. When a CRC indicates a problem, the signal can be fixed. For
minor errors, the CRCC can be used to replace the missing numbers exactly.
If the problem is more extensive, the system can use the previous and
following words to reconstruct a passable imitation of the missing one. One of
the factors that makes up the price difference in various digital systems is the
sophistication available to reconstruct missing data.
The Benefits of Being Digital
You may be wondering about the point of all of this, if it turns out that a digital
system is more complex than the equivalent analog circuit. Digital circuits are
complex, but very few of the components must be precise; most of the
circuitry merely responds to the presence or absence of current. Improving
performance is usually only a matter of increasing the word size or the sample
rate, which is achieved by duplicating elements of the circuit. It is possible to
build analog circuits that match digital performance levels, but they are very
expensive and require constant maintenance. The bottom line is that good
digital systems are cheaper than good analog systems.
Digital devices usually require less maintenance than analog equipment. The
electrical characteristics of most circuit elements change with time and
temperature, and minor changes slowly degrade the performance of analog
circuits. Digital components either work or don't, and it is much easier to find a
chip that has failed entirely than one that is merely 10% off spec. Many analog
systems are mechanical in nature, and simple wear can soon cause
problems. Digital systems have few moving parts, and such parts are usually
designed so that a little vibration or speed variation is not important.
In addition, digitally encoded information is more durable than analog
information, again because circuits are responding only to the presence or
absence of something rather than to the precise characteristics of anything.
As you have seen, it is possible to design digital systems so that they can
actually reconstruct missing or incorrect data. You can hear every little
imperfection on an LP, but minor damage is not audible with a CD.
The aspect of digital sound that is most exciting to the electronic musician is
that any numbers can be converted into sound, whether they originated at a
microphone or not. This opens up the possibility of creating sounds that have
never existed before, and of controlling those sounds with a precision that is
simply not possible with any other technique.
For further study, I recommend Principles of Digital Audio by Ken C
Pohlmann, published by McGraw-Hill inc ISBN number0-07-050469-5.
Peter Elsea 1996
Information on Quantisation
Quantisation is the process of converting a continuous variable signal (an
analogue signal) into discrete samples (a digital number). Only a limited
number of amplitudes are allowed in the discrete samples for example, if
three bits are used, only eight amplitudes can be represented. The difference
between the sample value and the original waveform value is called
quantisation error, or quantisation noise.
Books Related to Quantisation
S. Gutt, J. Rawnsley, D. Sternheimer and N. J. Hitchin, "Poisson Geometry,
Deformation Quantisation and Group Representations (London Mathematical
Society Lecture Note Series)".
—More information on Poisson Geometry, Deformation Quantisation and Group
Representations (London Mathematical Society Lecture Note Series)
Johan Stefan Erkelens, "Autoregressive Modelling for Speech Coding:
Estimation, Interpolation and Quantisation : Autoregressieve Modellering Voor
Spraakcodering : Schatten, Interpoleren En Kwantiseren".
—More information on Autoregressive Modelling for Speech Coding: Estimation,
Interpolation and Quantisation : Autoregressieve Modellering Voor Spraakcodering :
Schatten, Interpoleren En Kwantiseren
Other topics in our resources on Communications Systems related to
Quantisation include:
 analogue-to-digital conversion (ADC)
Information on Analogue-to-digital conversion (ADC)
Analogue-to-digital conversion (ADC) involves the conversion of the analogue
voltage levels of an analogue signal into ones and zeros of a digital word. It is
normally conducted by taking regular samples of the analogue signal and
converting the voltage level into a binary value.
Pulse-code modulation (PCM) is similar to pulse-position modulation (PPM) in
that the received information is not determined by the shape of the pulse, but
has the additional advantage that the precise location of the pulse is not
important either. The analogue signals are first sampled using pulse
amplitude modulation. The pulse amplitude modulation pulses are then
encoded into a binary code that is transmitted as a digital stream. At the
receiver these pulse code modulation (PCM) codes are decoded into pulse
amplitude modulation (PAM) pulses that are then used to reconstruct the
analogue waveform. In adaptive differential pulse-code modulation (ADPCM),
the binary number that is transmitted is not the number that corresponds to
the actual voltage level, but rather the difference between the current sample
value and the last. The difference values have a smaller dynamic range and
compression (of the order of one-half) results).