|The A-weighting curve is a filter designed to emulate the way the human ear hears noise and is defined in IEC61672:2003. It is designed to mirror the 40 phon equal-loudness-level of the human ear to pure tones as set out by Fletcher and Munson in their 1933 paper. In other words, each frequency is weighted according to the ear's sensitivity at that frequency.
|Of an electronic connection, this refers to a connection that can only pass alternating current (AC) and blocks DC (Direct Current), usually by the usage of series capacitors or an audio transformer.|
|ActiveX describes a group of technologies (incorporating OLE, automation etc) by which different Windows applications can communicate with each other and use each others' capabilities.
|While ADAT originally and more correctly referred to the Alesis digital multi-track tape format (Alesis Digital Audio Tape), it is now more commonly used to refer to the ADAT 'Lightpipe' digital interface (officially known as the ADAT Optical Interface) which carries 8 tracks of audio on a single fibre optic cable. This interface is now in widespread use by manufacturers other than Alesis and used in many different applications other than ADAT recording. While it uses the same physical optical cable standard as TOSLINK, the two are entirely incompatible. Furthermore, although it still retains the advantage over S/PDIF, TOSLINK or AES3 of being able to carry 8 signals over one cable (24bit, 48kHz), that advantage has been significantly diminished by the arrival of AES10 (MADI) which can now carry up to 64 channels over a single co-axial cable.
|An AES3 carrier used as a Reference Sync rather than to carry an audio signal. Also known as `DARS` (Digital Audio Reference Signal).
|An AES Standard describing measurement methods for digital audio systems. For more information, see AES17 test applications.|
|?||AES17 low-pass filter|
|A low-pass filter specified by AES17 with an upper band-edge frequency of 20kHz and a stop-band attenuation of at least -60dB by 24kHz. This filter is useful in eliminating switching noise from class-D amplifier measurements and also out-of-band noise in measurements of noise-shaped systems.|
|A two-channel digital audio interface standard. Also known as AES/EBU this format is used in professional applications usually with balanced XLR connections. It carries audio wordlengths up to 24 bits, plus Valid bit, User bit, Channel Status and Parity bit per channel.
|Aliasing can take place when a frequency above the folding frequency of an EUT is applied to its analogue input (or where internal down-sampling occurs). Aliasing is manifest by the appearance of a spurious frequency component at the EUT's output, below the folding frequency by the same frequency as the input stimulus is above the folding frequency. Thus the stimulus has been 'folded' about the folding frequency.
|?||Alternating Current (AC)|
|Of an electrical signal: one that alternates between positive and negative polarity, and whose current flow therefore reverses in direction, usually at regular intervals.|
|Usually used synonymously with "level", but more specifically the level of a periodic function, such as the voltage of an audio signal or the pressure of an acoustic wave. There are several different ways of measuring it: instantaneous, peak, peak-to-peak, RMS, etc. so it should be clarified which is meant.|
|(Scripting) an array of variables is a series of variables that are all of the same type and size. Individual elements in the array are accessed using the index of the element (zero-based);|
|(Audio Stream Input/Output) A digital audio device driver protocol for Windows computers that bypasses the Windows audio processing, allowing audio to be passed unaltered and with low latency between different applications and devices.|
|Not Synchronised. In digital audio this will be true of different pieces of equipment that are not running from the same clock.
|The gradual loss of energy as a signal or waveform propagates through a circuit or medium.|
|?||Audio Engineering Society (AES)|
|Professional and Standards body devoted exclusively to audio technology. See also the AES web site|
|Able to automatically select an appropriate gain range depending on the signal level.
(Test and Measurement) dScope Series III provides gain ranging in 2dB steps on its analogue inputs in order to precisely match the incoming audio signals to the AD converters and make the most of their performance. This auto-ranging can be manually overridden and a fixed gain set, although dScope will still auto-range in the event that the signal is too large for the manually set gain range.
|(dScope) Automation is the process by which one application can control another using its ActiveX interface. More specifically, the dScope can be automated to perform high speed production tests, or to perform custom measurements.|
|A method of transmitting an analogue audio signal or digital audio carriers, where two wires are used each carrying a representation of the signal or carrier in opposite polarity. Receiving equipment extracts the signal by subtracting one `leg` from the other, thus rejecting any signals common to both wires. In this way, interference from mains, radio communications etc. is rejected, assuming the CMRR of the receiving equipment is adequate at the appropriate frequency. dScope`s Analogue Inputs and Outputs can work in balanced or unbalanced modes.
|?||Band pass filter (BP filter)|
|(dScope) A filter in the Continuous-Time Detector and FFT Detectors of the dScope`s Signal Analyzer which can be set to band pass (BP) to make frequency-selective measurements.|
|?||Band reject filter (BR filter)|
|(dScope) A filter in the Continuous-Time Detector and FFT Detectors of the dScope`s Signal Analyzer which can be set to band reject (BR) to make residual measurements (e.g. THD+N).|
|1) (Signal Processing) The range of frequencies that a device or filter passes. Numerically, this is difference between the frequency points where the signal is 3dB down from the pass-band level in Hz.
2) (Computing) The rate at which data can be transferred over a connection. Typically this is measured in bits/s (or kbits/s, Mbits/s etc.)
|(FFT analysis) A single point in the graphical result of an FFT, corresponding to a small range of frequencies. The amplitude of each bin corresponds to the spectral content of the input signal within that frequency range.
|(dScope) A bin centres test signal is one made up of many individual sine waves, each one set to a frequency corresponding to the centre of an FFT bin.|
|The number of bits that are passed through an interface or processed per unit time. Common units are kbit/s, Mbit/s.|
|A timing signal that provides synchronisation for digital audio having transitions that correspond to individual data bits.|
|?||Brick wall filter|
|A high-pass or low-pass filter with a very narrow transition band. dScope's FFT Detectors have idealised brick wall filters with perfectly flat pass bands, infinitely attenuative stop-bands, and transition bands of only a single bin width.|
|?||Bridge Tied Load (BTL)|
|A technique for using two amplifiers connected in "bridged" mode to drive a single load in order to achieve an increased power output from a given voltage power supply. This is achieved by connecting the load between the positive terminals of two amplifiers and feeding one amplifier with an inverted version of the signal fed to the other.
In theory this should give 4x the output power of one of the amplifiers on its own (2x the voltage implies double the current, and therefore 4x the power), but in practice, since the power output of the individual channels is usually limited by their current capability, the output power is usually only doubled.
|A sound weighting filter that is largely flat with -3dB points at 31.5Hz and 8kHz, defined in IEC61672:2003.
|Within audio, usually used to refer to an AES3 or S/PDIF carrier, the digital interface signal carrying a sequence of binary data bits. These carriers are degraded in the real world by factors such as cable losses, which may cause jitter or other problems. The dScope`s Digital Output Generator can simulate such degradations, and its Digital Input Analyzer can measure them.|
|(dScope) A graphical representation of a section of the Digital Input Carrier displayed by the dScope|
|(dScope) A predefined array of crosspoints made up of one or more dS-NET switchers, such as I/O Switchers. A Channel Array can be controlled as a single entity, removing the need for the user to control multiple switchers directly|
|(Test and Measurement) A special mode of operation of the dScope's Digital Output and Digital Input used to verify data integrity of audio samples. This is achieved using a PRBS (pseudo-random bit sequence), which can be generated at the Digital Output and verified at the Digital Input. Input and output need not be synchronised, and so may be separated in distance (e.g. satellite link) or time (e.g. digital recorder). The Prism Sound DSA-1 hand-held tester can generate and verify the same sequence.|
|?||Channel Status (CS)|
|Status information embedded in an AES3 or S/PDIF digital interface, one bit per channel per sample-period, which accumulate into a 192-bit frame for each channel every 192 sample-periods. The frame is arbitrarily split into many fields of various lengths, with diverse functions as described in the appropriate interface standard document. The definitions of the fields and their meanings are different for 'Consumer' Channel Status (where the first bit of the frame is 0, used in S/PDIF) and 'Professional' Channel Status (where the first bit is 1, used in AES3). Originally conceived to add functionality to digital equipment interconnects, the proliferation of outputs with sloppy Channel Status implementation and inputs which mute if any unexpected Channel Status is received has led, like the Babel Fish, to much entirely unnecessary conflict.|
|See Swept Sine|
|A sophisticated digital re-clocking system designed to remove jitter from any
reference sync source before it is used as a conversion timebase, so as to eliminate any
audible effects of sampling jitter.
CleverClox is a trademark of Prism Media Products Ltd.
|?||Common-mode rejection ratio (CMRR)|
|A measurement of the ability of a balanced input circuit to reject an undesired signal that is common to both input terminals. The dScope Analogue Outputs have a CMRR test mode where the output signal is presented as common-mode instead of differential.|
|The part of an analogue signal that is "common to" (i.e., the same on) both legs of a balanced connection. Strictly speaking, it is the sum of the signal on both inputs of a differential amplifier divided by two, which is not quite the same thing.|
|?||Continuous Time Analyzer (CTA)|
|dScope Series III instrument for making time domain amplitude measurements with filters. A software instrument for displaying the readings from the CTA is called a "Continuous Time Detector"
|?||Continuous Time Detector (CTD)|
|dScope Series III time domain software instrument for measuring amplitudes with filters
|The frequency at which the response of a filter is 3dB down from its pass-band level.|
|The ratio the peak of a waveform (crest of a wave) to its RMS value. For random signals this needs to be measured over a specified time interval. For repeating signals, this needs to be measured over one cycle. A sine wave has a crest factor of 1.414 or √2 since the peak is 1.414 × the RMS value. This can also be expressed as 3.01dB.|
|Undesirable leakage of a signal from one circuit or channel to another.
|Also known as the corner frequency or 3dB down point, this is the frequency at which the response of a filter or device has fallen by 3dB from its maximum value. -3dB is equivalent to a halving of power.|
|(dScope) A setting of the dScope`s Inputs and Outputs which relates the maximum analogue amplitude to the maximum digital amplitude. It is expressed as an analogue amplitude which corresponds to 0dBFS (full-scale digital). This feature of the dScope is especially useful in measuring EUTs with both analogue and digital ports (as is common in applying the AES17 standard) since it allows settings or measurements in one domain to be expressed in the units of the other. For example, an A/D converter can be stimulated with a signal 1dB below that which will produce full-scale output by setting the dScope`s analogue generator amplitude to -1dBFS, provided that the dScope has been set to the appropriate D/A line-up for the converter.|
|The ratio of the rated impedance of a loudspeaker to the source impedance the amplifier driving it. Only the resistive components of both impedances are considered. It is measured at the loudspeaker terminals so the source impedance includes any speaker cable used to connect the speaker.|
|A type of interface jitter. Data jitter is that part of the interface jitter which is caused by variations in the duty cycle of the AES3 or IEC60958 carrier acting with high-frequency losses in the transmission medium (e.g. cable capacitance) such that edge timing in the carrier is modulated by the activity of the data bits. This is distinct from fs jitter, which is inherent in the carrier source. dScope can measure data jitter and fs jitter independently, so that the cause of jitter problems can be identified. Also referred to as `inter-symbol interference`.|
|(Decibels Full Scale) A logarithmic unit used to express signal amplitude in digital and mixed-domain systems in terms of the maximum amplitude which can be accommodated by an EUT. A 0dBFS signal has the same RMS amplitude as a sine whose peaks just touch the maximum level of the system. A -6dBFS signal has half this amplitude, for example. Note that for non-sine signals such as square waves or DC it is possible for the system to accommodate amplitudes greater than 0dBFS. In the dScope it is possible to specify or measure analogue signal amplitudes in dBFS, in which case the D/A line-up is used as a reference.|
|decibels relative to an amplitude of 1.000 milliwatt. Because this is a power measurement, it requires knowledge of the impedance.|
|decibels relative to a reference amplitude. The reference amplitude must be specified for measurements in dBr to be meaningful. In the dScope, dBr measures with respect to the Reference Amplitude as specified on the Signal Generator or Signal Analyzer panels.|
|an acoustic amplitude unit ("decibels, sound pressure level"): decibels relative to a the threshold of hearing (0dBSPL or 20µPa). 94dBSPL is equivalent to a sound pressure level of 1 Pa (Pascal), and is a common amplitude at which to state the calibrated output voltage of a measurement microphone.|
|decibels relative to an amplitude of 1.000 Volts.|
|Of an electronic connection, this refers to a connection that can pass DC (Direct Current).|
|An analogue audio input or output is said to be DC-coupled if it does not remove DC content from transmitted signals. If not, it is said to be DC-blocking. The dScope Analogue Outputs are DC-coupled. The Analogue Inputs are DC-blocking by default, but can be DC-coupled if required.|
|A way of expressing one number relative to another as a ratio. A Decibel is one tenth of a Bel, although the Bel is almost never seen except in texts such as this. A Bel is a factor of 10 of power so 2 Bels is two factors of 10 or 100 × the power. This turns out to be a rather large unit in practice, so the decibel is used instead. Since the decibel is a power ratio, when working with power values, it is calculated as 10Log10(power1/power2). When working with voltages, since power is proportional to voltage squared, the formula becomes 10Log10(V12/V22) which can be written 20Log10(V1/V2).
It can be seen from this that a decibel is a dimensionless unit - it is just a means of expressing the ratio of two values in the same units. It is particularly useful for representing very large differences, for example, a power ratio of 1 million to 1 becomes 60dB.
For making absolute measurements, the dB can be used with a specified reference. In this way dBm is decibels relative to a level of 1milliwatt. Sound pressure level (dBSPL) is measured relative to a level of 20µPa (twenty micro Pascals) and is calculated as 20Log10(P/P0) where P0 = 20µPa
|(dScope specific) A rectifying voltmeter with a particular dynamic response, used to measure the amplitude of audio signals or residuals. dScope`s Continuous-Time Detector and FFT Detectors offer two alternative versatile types of Detector.|
|?||Digital Audio Reference Signal (DARS)|
|An AES3 carrier used as a Reference Sync rather than to carry an audio signal.
|?||Digital Audio Workstation (DAW)|
|A system designed to record, edit and play back digital audio, typically on a computer. The main characteristic of a DAW is its ability to freely edit and manage multiple channels of digital audio.|
|?||Digital Signal Processor (DSP)|
|A specialised microprocessor designed specifically for mathematically processing digital signal data.|
|?||Digital Theater Systems (DTS)|
|A family of multi-channel digital surround sound formats, used for a wide range of commercial and consumer audio technologies developed and licensed by DTS Inc.|
|?||Direct Current (DC)|
|Of an electrical signal: one which has a fixed polarity and therefore a current that flows in one direction only.|
|?||Direct Stream Digital (DSD)|
|A technology for digitally storing and recreating audio signals using pulse density modulation (PDM) of a single bit at 2.8224MHz. DSD is a trademark used by Sony and Philips and is the technology used for SACD.|
|?||Disc Description Protocol (DDP)|
|An industry standard protocol (ANSI) for describing all the files and parameters necessary for accurately replicating CDs or DVDs. As well as files that contain the contents of the disc, it includes additional files that describe the structure, subcode, table of contents, indexing etc. required to create a complete disc, making it possible for a mastering house to send a complete disc definition to a pressing plant as a single file. DDP is a registered trademark of DCA Inc.|
|?||Discrete Fourier Transform (DFT)|
|A Discrete Fourier Transform (DFT) calculates the spectrum of a sampled signal, i.e. it transforms the time domain signal sample data into the frequency domain.
|Anything that changes the shape of (distorts) an audio waveform other than by changing its amplitude. This includes noise and hum, as well as effects of non-linearity (i.e., where the output level is not directly proportional to the input level).
|low amplitude noise, added to a signal before quantization, or re-quantization, to linearize the loss of precision. In the dScope, dither is applied by default to Digital Outputs. Best linearization is achieved by TPDF dither.
|A Prism Sound proprietary open serial interface protocol used to connect peripherals such as I/O Switchers to the dScope.
|The proportion of time that a signal or device is active ("on duty") or non-zero and is given by the active time divided by the total time.|
|The ratio of the maximum signal level to the level of noise expressed in dB. This can apply to a signal or a device. In the former case, it is also used to mean the ratio of the loudest signal to the lowest signal in a recording or transmission, and in the latter it is used to refer to the ratio of the loudest signal that can be produced without distortion to the output noise floor.
|?||Dynamic Range Enhancement (DRE)|
|Prism Sound DRE is a process designed for increasing the
dynamic range of digital recordings when further post-processing is required. It requires an encode process on recording
and a decode process on playback. It enables 20-bit dynamic range on 16-bit tracks or
24-bit dynamic range on 20-bit tracks.
|Or "Ground"; a reference point in an electrical or electronic circuit from which other voltages are measured. This may or may not be physically connected to the Earth.
|?||Edit Decision List (EDL)|
|An EDL traditionally is a list on paper indicating how a film or video program is to be edited. It would contain a sequential list of reel and timecode markers to indicate where each clip comes from in order to compile the completed project.
On modern digital editors, the EDL is a computer file that is constructed by the editing software as the operator creates edits, and is used to automate the process of constructing a final edit from the source material.
|(scripting) an element is a single item in an array.|
|?||Equipment Under Test|
|(Test and Measurement) The equipment or system being tested.|
|(dScope Automation) A causal occurrence for the dScope Event Manager or an Event-driven VBScript. Examples might be breaching of a Limit, or a change in received Channel Status.|
|(Scripting) an event handler is a subroutine that is inserted into a script to handle a certain event. When that event is triggered, the event handler subroutine will be called to take the appropriate action.|
|A dScope feature which allows the user to set links between various causes and effects. Thus a range of interesting occurrences in the EUT can be pre-armed to trigger responses such as audible or visible warnings, entries in log files, or even running of VBScripts.|
|(Scripting) a script is said to be event-driven if its main body has finished running, and the only code that subsequently runs is triggered by a certain event happening (for example, a Limit Line being breached).|
|(Scripting) a combination of keywords, operators, variables, and constants that yield a value. This value may be a string, number, or object. An expression can perform a calculation, manipulate characters, or test data.|
|Also known as an eye diagram, this is an oscilloscope type display of a digital carrier waveform which has been accumulated over a number passes in order to show the variation of the signal over time.
|The AES3 and IEC60958 standards define acceptable carrier degradation in terms of amplitude and edge-timing using an eye-diagram, which shows the minimum acceptable differential carrier amplitude over a defined period within 1 UI of the carrier. This can be verified on the dScope using the Carrier Display feature.
|The dScope can measure the worst-case narrowing of the eye of an AES3 or IEC60958 carrier. This is essentially a measurement of data jitter, and can be referred to the eye-diagram in the AES3 or IEC60958 standards
|?||Fast Fourier Transform (FFT)|
|A Discrete Fourier Transform (DFT) calculates the spectrum of a sampled signal, i.e. it transforms the time domain signal (e.g. the dScope`s sample buffer, as shown by the Scope Trace) into the frequency domain. A Fast Fourier Transform (FFT) allows a DFT to be calculated more efficiently, i.e. faster, assuming that the length of the data set (sample buffer) is 2n samples.|
|?||FFT Analyzer (FFTA)|
|(dScope) The dScope's FFT Analyzer processes buffers of captured audio sample data, converting into the frequency domain using FFTs. It has a wide range of Detector functions which access this frequency domain information in order to make measurements using filters etc.
|?||FFT Detector (FFTD)|
|(dScope) The dScope's FFT Analyzer processes buffers of captured audio sample data, converting into the frequency domain using FFTs. FFT Detectors are software instruments for deriving audio measurements from the frequency domain data.
|Apple's brand name for the IEEE 1394 serial interface standard. Widely used for data transfer by computer equipment, it has also become common to be used as an interface between computers and digital audio equipment where its ability to sustain high transfer data rates with low latency has proved very useful for carrying multiple channels of high resolution digital audio.
The original 1995 version of the IEEE 1394 standard specified what is now known as FireWire 400 (based on its ability to transfer data at almost 400 Mbit/s). The 2002 update (IEEE 1394b-2002) introduced a new encoding scheme called "beta mode" which doubled this transfer rate and became known as FireWire 800. This standard uses a different connector, but is backwards compatible with FireWire 400 if a "bilingual" cable is used to connect older devices to the newer port.
|Half the sample rate of the EUT. Input frequencies above the folding frequency are subject to aliasing. Where an EUT applies internal down-sampling, the folding frequency is half of the lowest internal sample rate employed.|
|In imaging systems such as television and computer graphics, the rate at which images are presented, usually expressed in frames per second.
In digital audio transmission protocols such as S/PDIF and AES3, the audio is packaged into frames that contain the sample data for one sample point on both channels. Frame rate in this context is therefore equivalent to sample rate.
|?||Frequency Correction Window|
|(FFT Window functions / Test & Measurement) Within the dScope Series III, a special kind of window function is available which allows 'windowless' FFTs (i.e. a Rectangular window) in situations where the Generator and Analyzer sample frequencies are not the same, for example in cross-domain or SRC measurements. Selection of these 'frequency correction' window functions causes the Analyzer signal to be sample-rate converted to return frequency components which were generated at bin centres to be restored to bin centres despite the sample frequency difference.
|A type of interface jitter. fs jitter is that part of the interface jitter which is inherent in the equipment which is the source of the AES3 carrier. This is distinct from data jitter, which is caused by variations in the duty cycle of the AES3 carrier acting with high-frequency losses in the transmission medium (e.g. cable capacitance) such that edge timing in the carrier is modulated by the activity of the data bits. dScope can measure data jitter and fs jitter independently, so that the cause of jitter problems can be identified.|
|?||Full-scale amplitude (FS)|
|(AES17 standard) A signal whose amplitude is the maximum which can be accommodated by the EUT. In the case of a sine, this amplitude is 0dBFS. For systems where the output is accessible in the digital domain, full scale is defined as the RMS voltage of a 997Hz sine wave whose positive peak value reaches the positive digital full scale. For systems where the digital signal is not accessible, or where digital full scale cannot be reached, full scale is defined as the level 0.5dB below that where 1% THD+N or 0.3dB compression occurs (whichever comes first) at the EUT output for a 997Hz sine wave.|
|The lowest frequency of a harmonic series. |
(Test and Measurement) The original signal, particularly when measuring Total Harmonic Distortion etc.
|The increase in level of an audio signal. Measured in dB, an increase in level gives a positive gain. A negative gain is also referred to as attenuation.|
|Isolation of electrical systems so that electrical charge carrying particles cannot pass between them, however energy or information can still be transferred between them by other means such as capacitance (a capacitor coupled input), inductance (for example, via a transformer) or optically (for example, via an optical digital interface).|
|Or "Earth"; a reference point in an electrical or electronic circuit from which other voltages are measured. This may or may not be physically connected to the Earth.|
|The condition in an electrical circuit or system where there is more than one path to ground from a given point. Any difference in the two ground paths can cause current to flow which results in unwanted noise. The remedy will depend on the particular circuit, but under no circumstances should equipment safety grounds be removed.|
|?||Head and Torso Simulator (HATS)|
|A dummy head and torso with artificial ears and ear canals into which are fitted microphones for the purpose of simulating human responses to acoustic stimuli.|
|A unit of frequency equal to one cycle per second.|
|A convention for conveniently representing binary data such as digital audio samples. Each four-bit 'nibble' of the binary word is represented by a character indicating its value: 0..9,A..F. For example, the 24-bit binary value 000000010010110111101111 would be represented in hex as 012DEF. Hex is available as an amplitude unit throughout dScope in order to facilitate some digital measurements.|
|?||High-pass filter (HPF)|
|A filter which attenuates frequencies below a particular corner frequency and allows higher frequencies to pass.
|(Scripting) This is a system used by some programmers, in which the type of a variable is specified by inserting one or more letters at the beginning of the variable name.
|ICP is an acronym for "Integrated Circuit Piezoelectric", and is a registered trademark of PCB Group, Inc. ICP transducers incorporate built-in signal-conditioning electronics and are used to measure dynamic pressure, force, strain, and acceleration. The built-in electronics convert the high-impedance charge signal that is generated by the piezoelectric sensing element into a usable low-impedance voltage signal that can be transmitted over ordinary two-wire or coaxial cables over long distances with little degradation. ICP sensor circuitry can also include other signal conditioning features, such as gain, filtering, and self-test features.
The electronics within ICP accelerometers require a constant-current regulated, DC voltage source. This power source is sometimes built into test equipment to allow the direct connection of ICP devices and is sometimes referred to as ICP Power.
|A two-channel digital audio interface standard also known as S/PDIF. This format is used in consumer applications usually with an unbalanced phono/RCA copper or TOSLINK optical connection. It carries audio wordlengths up to 24 bits, plus Valid bit, User bit, Channel Status and Parity bit per channel.|
|Imaging can take place in an EUT with a D/A converter or which performs digital up-sampling. It is manifest by the appearance at the EUT`s output of a spurious frequency component above the folding frequency by the same frequency as the stimulus is below it. It is caused by insufficient stop-band attenuation in the EUT, and is most noticeable close to the folding frequency.
|?||IMD difference-tone measurement|
|An IMD measurement method (e.g. CCIR) wherein two tones, of equal amplitude, close together in frequency (e.g. 19kHz and 20kHz, or 14kHz and 15kHz for band-limited systems) are applied to the EUT. The amplitude of the distortion component at the difference frequency (e.g. 1kHz) is measured, usually relative to one of the original tones.|
|?||IMD side-band measurement|
|An IMD measurement method, typically using a low-frequency high-amplitude tone, and a high-frequency tone at 1/4 the amplitude (the SMPTE standard uses 60Hz and 7kHz). The intermodulation distortion appears as side-bands around the high frequency tone, although historically it has been measured after demodulation to the base-band.|
|A measure of opposition of a circuit to alternating current (AC). Impedance is the sum of resistance and reactance. Resistance is analogous to friction in mechanical systems and introduces no phase difference between the voltage and current flowing through it. Reactance is caused by the build up of electric or magnetic fields due to the current flow. These cause an opposition voltage that is either proportional to the rate of change of current or the time integral of current through the circuit and causes a phase shift between voltage and current. The equivalent in a mechanical system would be a mass on a spring in which case the reactance comes from the inertial mass (inductance) and spring constant (1/capacitance). A pure reactance would not dissipate any power.|
|?||Impulse response analysis|
|An analysis technique for measuring EUT parameters such as temporal dispersion and frequency response by comparing the EUT`s output with its input. Impulse response analysis is most often used in acoustic testing, to characterise rooms or loudspeakers; however it can also be useful in testing ordinary analogue or digital EUTs. The technique is general, and requires no special stimulus, although the stimulus should ideally cover the entire band of interest at reasonable amplitude. In the dScope, the method of comparison is such that the stimulus must repeat exactly over a 2^n sample period - for these reasons, `swept sine` and `bin centres` stimuli are most often used. The comparison of output and input yields the `impulse response` of the EUT, i.e. the output which it would produce if a perfect narrow impulse were applied to its input. The impulse response itself is indicative of such factors as delay and reverberance of the EUT. The FFT of the EUT reveals the EUT`s frequency response. By excluding all noise before and after the impulse prior to calculating the FFT, a loudspeaker can be measured as if in anechoic conditions.|
|Sound at frequencies too low to be heard by the human ear, typically 0.001 to 20Hz.|
|Jitter present on a digital audio carrier or reference sync. Interface jitter usually comprises fs jitter and data jitter components. The dScope can generate and measure interface jitter directly, and incoming interface jitter can also be demodulated for analysis by the Signal Analyzer. Interface jitter is often blamed for sonic degradation in A/D and D/A converters, but this is usually due to sampling jitter within the conversion equipment resulting from the equipment failing adequately to remove incoming interface jitter from the conversion clock. Where good quality converters are used, interface jitter is not usually problematic until it reaches very high levels, when data loss can result. The AES3 standard defines a jitter tolerance template (jitter vs frequency) for correct receipt of data.|
|?||Intermodulation Distortion (IMD)|
|Intermodulation Distortion. When a signal consists of more than one frequency, a non-linear device under test will produce the original frequencies plus an infinite number of IMD products, given by
(a * F1) + (b * F2) + (c * F3) + ...
where (a, b, c) etc. are all possible integer numbers, and (F1, F2, F3) etc. are the frequencies of the original tones.
|?||ITU-R 468 weighting|
|(originally CCIR 468) A weighting curve used for measuring noise in audio systems. Whereas A-weighting is based on human response to pure tones at low levels, ITU-R 468 was developed in conjunction with Q-peak response to give results that correlate with human response to noise. It was originally intended for use in assessing FM radio and analogue cassette noise etc.
|Variation in edge-timing of a clock signal. In audio systems, manifestations are interface jitter and sampling jitter.|
|The effect on an EUT of jitter present on its reference sync input (which may also be its data input). Such effects may include jitter passed to the EUT`s digital output or, if the EUT includes A/D, D/A or SR conversion, distortion of the audio signal owing to sampling jitter. In severe cases, loss of digital data at the EUT`s input can also occur.|
|?||Jitter Time Analyzer (JTA)|
|(dScope) An element of the dScope`s hardware which analyzes the incoming digital audio carrier. It performs time-domain analysis (jitter and amplitude measurement) of the carrier, and collects data for the Carrier Display.|
|1000Hz or one thousand cycles per second.|
|The time it takes for an audio signal to pass through a system, usually expressed in milliseconds (ms).
|?||Least Significant Bit (LSB)|
|(Digital Audio) When representing digital sample values in binary, the LSB is the bit which represents the smallest value or the value of 1 and is also sometimes known as the right-most bit.|
|(Test & Measurement) In pass/fail testing, a limit can be applied to a measurement such that if the reading exceeds the limit the EUT is failed. This can be set to cause an audible or visual warning, logging of the Event in a log file, or even the automatic running of a VBScript.
|(Test & Measurement) In pass/fail testing, a Limit Line can be applied to a Trace, so that an Event is triggered if the Limit Line is breached. This might cause an audible or visual warning, logging of the Event in a log file, or even the automatic running of a VBScript.
|A linear system is one where the output is directly proportional to the input. The output level will always be a multiple of the input, and therefore will only change the amplitude or phase of a signal without affecting its shape. Drawn on a graph with the input on one axis and the output on the other, the transfer function will be represented by a straight line, hence the name.|
|?||Linear Pulse Code Modulation (LPCM)|
|This is the most common and basic method of representing an analogue waveform in uncompressed digital form. Essentially the amplitude of the waveform is measured at uniform intervals and stored as a series of numbers. For example, audio for use on a CD is measured (sampled) 44100 times per second. At each point the amplitude is stored as a 16 bit binary number having 65536 different possible values where the number is a direct linear representation of the amplitude. Both channels are represented separately using the same technique. When stored as a computer WAV file, the resulting string of numbers is simply stored with a header defining the format used. Storage on CD requires further encoding to enable the mechanics of the CD to work properly, but the digital audio when recovered from the disk essentially remains encoded as a series of 16 bit numbers at the rate of 44100 per second.|
|(dScope) A Scope, FFT or Sweep Trace in the dScope`s Trace window. These Traces are subject to `Live update` as opposed to Copy Traces, Filter Traces etc.|
|See Swept Sine|
|?||Low-pass filter (LPF)|
|A filter which attenuates frequencies above a particular corner frequency and allows lower frequencies to pass.
|(Multi-channel Audio Digital Interface) Defined by AES10-2003, MADI is a standard protocol for sending multiple channels of digital audio over a single connection. It has many features in common with the AES3 standard. It uses serial transfer of data over either a co-axial cable or fibre-optic link and is capable of carrying up to 64 channels with sampling rates up to 96kHz and up to 24 bits per sample.
|?||Maximum Length Sequence (MLS)|
|A pseudo-random binary sequence which includes all the possible binary combinations of a digital word of length n except all zeros. The sequence itself is then 2n -1 bits long. Maximum Length Sequences have an inherently white (flat) frequency spectrum. They have been used for a number of years for deriving impulse responses by the use of the Fast Hadamard Transform (FHT). The FFT analysis of the resulting impulse response allows quasi-anechoic acoustic measurements. Their use has been largely superseded by Log Swept Sine techniques which have a number of practical advantages.
|Data about data - typically in a computer file or data stream, this is information about the data contained in the file or carried on the stream.|
|(Scripting) an object's method is a function that has been made available via its OLE interface, allowing the function to be called by another application.|
|?||Modular Digital Multitrack (MDM)|
|This now somewhat antiquated term was used to refer to machines such as the TASCAM DA-88 and Alesis ADAT digital recorders which recorded 8 tracks of digital audio to video tape. They were modular in the sense that they had to be used in multiple units synced together to work with more than 8 channels simultaneously.|
|?||Most Significant Bit (MSB)|
|(Digital Audio) When representing digital sample values in binary, the MSB is the bit which represents the greatest value and is also sometimes known as the left-most bit. The actual binary value of this bit will depend on the length of the digital word, for example, 16 bit or 24 bit.|
|Prism Sound MR-X, is a 'word-mapping' or 'bit splitting' system which allows tracks on a
multi-track medium to be sacrificed in order to make up the extra wordlength.
|(Scripting) if an application is multi-threaded, it means that is has two or more threads all running concurrently, and this enables more than one task to be performed at a time.
In fact the tasks do not run at exactly the same time, but the Windows operating system manages switching between them to make it look as if they are happening at the same time.
|A method of testing where a number of discrete tones are used to stimulate the EUT simultaneously. By capturing a single data set of the output of the EUT, many measurements can be calculated simultaneously. These can include scalar Results such as noise, distortion etc. as well as graphical plots against frequency, such as frequency response, distortion spectrum etc. This method is much faster than traditional methods, which would require stimuli to be changed for each scalar measurement and stepped through many frequencies for each plot. The dScope has a uniquely user-friendly way of setting up multi-tone tests.|
|?||Musical Instrument Digital Interface (MIDI)|
|A digital serial interface standard designed to transmit and receive musical note information between electronic musical instruments. Rather than transmit audio, it transmits data about note on and note off events which the receiving equipment is expected to interpret. Since its inception, its usage has expanded to a far wider range of applications from transferring system information and control information to being used for equipment software upgrades, show control and time code. Although the MIDI standard defines a wired interface with an opto-isolated 5 pin 180 degree DIN connector, variants of the MIDI protocol are increasingly being sent down other connections such as Ethernet, USB and FireWire.|
|(FFT Window Functions / Test & Measurement) Within the dScope Series III, a variant of the Rectangular window is included to improve impulse response testing with single or low-number shots of the stimulus where results can be compromised by low frequency drifting of the EUT output - this is especially common when the EUT is DC-blocked with a low corner-frequency and the 'settling' time of the blocking filter is long in comparison to the duration of the stimulus. This compromises the synchronous nature of the FFT (it must be continuous between its end and beginning, since it is notionally repeated over all time) and results in disturbances which look like sporadic periods of ringing on the impulse response. The effect is not noticeable when using a continuous stimulus, because there is no discontinuity when the DC-blocking system has settled.
Selecting the 'None (n-shot correction)' Window function instead of the normal Rectangular window function alleviates this. This window function is still rectangular in shape, but applies a 'tilt' to the sample buffer in order to remove the discontinuity between its two ends.
|A Sweep which is repeated for each state of a second varying `outer` source. A `two-dimensional` Sweep.|
|Unwanted sound or signals, usually unrelated to a wanted sound or signal.|
|The process by which quantization and dither noise is processed in order to make it less audible. This is mostly done as part of a bit reduction scheme, for example, creating a 16 bit CD from a 24 bit master. The dither noise and quantization noise from this process has a flat (white) spectrum which results in a noise similar to tape hiss. The ear is more sensitive to some frequencies than others, so by shaping the frequency content of the noise, we can shift it to frequencies where the ear is least sensitive so as to give an
improvement in subjective signal to noise ratio.
|(Electronics / Test & Measurement) A non-linear system is one where the output is not directly proportional to the input. This gives rise to distortion of a waveform passed through the system which results in frequency components being present on the output that are not present in the original input. Drawn on a graph with the input on one axis and the output on the other, the transfer function will be represented by a line that is not linear, hence the name.
(Editing) An editing system is described as non-linear if it is capable of handling audio in an order other than that in which it was originally recorded. A tape recording system is referred to as linear because the audio can only be played back in the same order in which it was recorded.
|In sampled systems, the Nyquist frequency (also known as the folding frequency or the cut-off frequency) is half the sampling frequency.|
|?||Object Linking and Embedding (OLE)|
|(Scripting) This is a term used for a number of ways in which Windows allows different applications to interact with each other.|
|The interval between two frequencies where the upper frequency is double the lower.|
|Of an electrical or electronic circuit, containing a gap over which electrons cannot flow.|
|The Prism Sound Overkiller is a progressive analogue peak-limiter fitted to various Prism Sound products which incorporate A/D conversion and available as a stand alone device.
The Overkiller allows analogue input signals far above the normal maximum handling level of the A/D converter to be accommodated without causing the converter to clip. This is done in a gentle and progressive manner so that distortion is as inaudible as possible.
|A bit added to a given set of bits to ensure that the sum of the bits is always even or odd. In the AES3 interface standard a parity bit is provided to permit the detection of an odd number of errors resulting from malfunctions in the interface. If set, it indicates an even parity.|
|The SI unit of pressure equal to one Newton per square metre.
In acoustics, the dBSPL scale is referenced to an acoustic RMS pressure of 20µPa which roughly corresponds to the human threshold of hearing at 1kHz.
|The band of frequencies that are passed by a filter. This is normally measured between the cutoff frequencies (3dB down points) of the system.|
|The highest instantaneous level of an audio signal, positive or negative.
(dScope) When measuring peak level using dScope, the Peak is distinct from the Peak-sample as it is interpolated using an oversampling filter.
|?||Peak Program Meter (PPM)|
|An audio meter designed to measure and display the peak level of an audio signal. Originally a BBC standard dating back to 1938, it has a peak-like response with a 1.7ms attack time and 650ms decay time. This meter is defined in IEC 60268-10. A version for monitoring digital signals is also defined in IEC 60268-18.|
|The highest instantaneous sample value of a digital audio signal.
|The difference between the maximum instantaneous positive and maximum instantaneous negative amplitude of an audio signal.
|The duration of one cycle of a repeating event or waveform. The reciprocal of frequency.|
|A DC voltage applied equally to both signal lines of a balanced audio connection. Its primary use is to power condenser microphones although it is also used to power DI boxes etc.
|(waves) For sinusoidal waveforms, the phase is the fraction of a complete cycle since a specified point in time. Phase is usually measured in degrees or radians since a complete cycle of a sinusoidal function takes place over 360 degrees or 2π radians.
|?||Phase Locked Loop (PLL)|
|A circuit for adjusting the frequency of an oscillator so that it is perfectly synchronised to another external oscillator. This is done by comparing the frequency and phase relationship between the two and adjusting the local oscillator until the phase difference between the two is fixed. The primary application in audio is for a digital receiver circuit that must lock to an external digital input carrier.|
|Random noise whose energy is constant per octave band. When viewed as an FFT spectrum on a logarithmic frequency axis, its level drops at 3dB per octave due to the linearly spaced analysis frequency intervals. When viewed on a real time analyzer using fractional octave bands, it gives a flat line.
|(Voltage) Whether a potential is positive or negative with respect to some reference or whether a parameter is proportional or inversely proportional to another.
|Encoding of subcode information to a digital master in preparation for CD manufacture. The PQ data includes such things as the TOC (Table of Contents), the track start points, the track sub-index points etc. Its name refers to the first two of the 8 subcode channels on an audio CD: P, Q, R, S, T, U, V, and W.
|(Scripting) an object's property is a setting that has been made available via its OLE interface, allowing the property to be set and/or read by another application.|
|?||Pulse Code Modulation (PCM)|
|This is the most common and basic method of representing an analogue waveform in uncompressed digital form. Essentially the amplitude of the waveform is measured at uniform intervals and stored as a series of numbers. For example, audio for use on a CD is measured (sampled) 44100 times per second. At each point the amplitude is stored as a 16 bit binary number having 65536 different possible values. Both channels are represented separately using the same technique. When stored as a computer WAV file, the resulting string of numbers is simply stored with a header defining the format used. Storage on CD requires further encoding to enable the mechanics of the CD to work properly, but the digital audio when recovered from the disk essentially remains encoded as a series of 16 bit numbers at the rate of 44100 per second.|
|'Quasi-peak' response. CCIR 468-2 specifies a fast-attack, slow-decay 'Q-Peak' detector which is intended to produce a measure of noise signals which corresponds to subjective level. It is usually used in conjunction with a special weighting filter also specified in CCIR 468-2.|
|The process of approximating a continuously variable parameter by a series of discrete values. When sampling audio for storage on CD, the signal is quantized to 16 bit resolution, meaning the level at any particular sample point has to be represented by a number that can be defined by a 16 bit binary word. This means it must be one of 65,536 distinct numbers (216 = 65,536). An 8 bit telephone signal has only 256 distinct different levels. The difference between the original signal level and the quantized level is what gives rise to quantization noise.|
|(Scripting) A property is said to be read-only when its value can be read, but not written.|
|(dScope) A dScope Result can be converted to a Reading (capital R) by dragging it off its home dialogue box. Readings have many additional functions over native Results; for example, they can be resized, user-coloured, and can have Limits and bar graphs attached to them.|
|Of a process that takes place at the same time and the same rate as real events. Taking a digital audio computational process as an example, a process that can be run on the incoming audio and completed fast enough that it can be output again continuously at the same rate it came in is said to be a real-time process. If the processing cannot keep up with the audio at its input in order to maintain an un-interrupted output, it cannot be said to be real-time and has to be performed "off-line".|
|?||Real Time Analyzer (RTA)|
|A device that displays the frequency content of an audio signal in real time. Originally this was achieved by filtering the signal into octave or fractional octave (often 1/3rd octave) bands and displaying the level in each band with a bar-graph LED meter. Now this is far more likely to be implemented in software and displayed on a computer screen.
Because the RTA divides the spectrum into equal fractional octave bands it will display a flat line with pink noise (pink noise having equal energy per octave) whereas an FFT spectrum analyzer (which divides the spectrum into bins of an equal number of Hz) will show a flat line with white noise which has equal energy per Hz.
|(FFT Window functions) A rectangular FFT window function is effectively a "windowless" FFT as it applies no weighting to the sample data prior to the FFT calculations.
|A signal passed between digital audio equipment for the purpose of defining the sampling clock. It is usually in AES11, Wordclock or video format. The dScope can accept a Reference Sync for its Digital Outputs in any of these formats|
|(Test & Measurement) A process by which a Result is brought to a target value by varying a defined parameter, usually on the signal generator.|
|A quantity left over at the end of a process. In audio test and measurement, this typically refers to the remaining signal when the original test tone has been removed with a notch filter such as when measuring THD+N.|
|(dScope) Any numerical output in a dScope dialogue box. Results can be converted to Readings by dragging them off the dialogue box in order to give them additional functionality.
|?||Root Mean Square (RMS)|
|The square root of the mean (average) of a series of numbers that have each been squared. A way of averaging a series of numbers, particularly that includes positive and negative values, in order to get a meaningful effective average.|
|?||Rub and Buzz|
|(Test and Measurement) a fairly broad term typically used to describe mechanical defects in manufacture, particularly of loudspeakers, that exhibit noises characterised by rubbing and buzzing, as well as rattle defects and noises caused by loose particles.|
|(Sony/Philips Digital Interface) A digital audio interface standard defined by IEC60958|
|?||Sample Rate Conversion (SRC)|
|The process of converting a digitally sampled signal from one sample rate to another (e.g., 96kHz to 44.1kHz), while introducing as little distortion and noise as possible.
|The rate at which a digital audio signal has been sampled. Standard sample rates include 32kHz, 44.1kHz, 48kHz, 88.2kHz, 96kHz, 176.4kHz and 192kHz|
|Caused by jitter present on the sampling clock of an A/D or D/A converter (or a sample-rate converter). Sampling jitter results in distortion of the converted audio, which is worse at higher frequencies. In practice, sampling jitter often occurs in conversion equipment which does not adequately remove interface jitter from its reference sync. Sampling jitter is usually measured by passing a high-frequency tone through the converter under test, and applying jitter of varying frequency to its reference sync. Sampling jitter is manifest by side-bands on the converted tone, and the variation in amplitude of these with varying jitter frequency enables the jitter rejection characteristic of the conversion equipment to be measured.|
|(dScope) An abbreviation for oscilloscope. Scope trace: the time domain waveform as displayed in dScope software.
(Scripting) the scope of a variable defines the context in which it can be seen by a script. A variable declared outside any functions or subroutines has 'global' scope, that is, it can be seen and used by anything in the script. A variable declared within a function or subroutine has 'local' scope, and can only be used from within that function or subroutine.
|(Scripting) A script engine is the entity that processes a script - that is, checks for syntax errors and runs the code in the script.|
|(dScope automation) a means of automating dScope test sequences or creating custom measurements by using small program sequences called scripts.|
|(Scripting) A scripting host is a program that contains a script engine, thus allowing it to run a script to control itself or other applications|
|(transducers) A measure of how responsive a transducer is to an input stimulus, or how effective it is at transforming one form of energy into another. It is usually defined as the number of output units that are measured for a given input unit and is not the same as efficiency.
|(Signal In Noise And Distortion) This is the reciprocal of THD+N and is expressed in dB. Whereas THD+N is the ratio of the noise and distortion to the signal (small is better), SINAD is the ratio of the signal to the noise and distortion (big is better). If the THD+N is -90dB, SINAD under the same conditions would be 90dB. SINAD is more common in communications and broadcast, whereas THD+N is more common in the audio industry.|
|In electronics, the slew rate represents the maximum rate of change of a signal, usually expressed in Volts per microsecond (V/µs). An electronic circuit whose slew rate is too low to be able to follow the input signal accurately will introduce distortion to the waveform.|
|?||Sound Pressure Level (SPL)|
|A measure of the RMS sound pressure relative to a reference pressure and expressed in dB. For sound in air, the reference is 20µPa RMS, thus the equation becomes
dBSPL = 20Log10(pRMS/pref)
|Split96 is a mode of digital interfacing (usually of AES3) whereby a two channel interface carries a single audio channel sampled at twice the frame rate of the interface. Each frame contains the data for two successive samples of the same channel, rather than a sample from the left channel and a sample from the right channel. Also known as 'two-wire' interfacing as two wires become necessary to transmit a stereo signal in this format.|
|A periodic waveform which alternates at regular intervals between two different levels.|
|?||Standard high-pass filter|
|(AES17 Testing) A high-pass filter with a stop-band beginning at the upper band-edge frequency of the EUT. This filter is used to measure `out of band` components.
|?||Standard low-pass filter|
|(AES17 Testing) A low-pass filter with a pass-band up to the upper band-edge frequency of the EUT. This filter is used to restrict measurements to 'in band' components.
|?||Standard notch filter|
|(AES17 Testing) A band-reject filter with a Q between 1 and 5, for example the dScope`s band-reject filter at the `1/3 octave` setting.|
|?||Standard weighting filter|
|(AES17 Testing) A weighting filter with a response according to CCIR468, but normalised to have unity gain at 2kHz, for example the dScope's 'CCIR468-2k' weighting filter.|
|Non-audio control data encoded with the audio on a CD that defines such things as track start and end points, running times, track numbers etc. The data in a CD are arranged in frames 33 bytes long, one byte of which is the subcode. Each bit of the subcode is given a letter from P to W and corresponds to different channels of subcode information. Only the first two, P and Q are defined and used in Red-Book standard audio CDs.
|?||Super Noise Shaping (SNS)|
|A family of noise shaping algorithms that can operate at sample rates between 44.1 kHz and 192 kHz and is used to reduce longer word lengths to standard 16 bits, while retaining a high degree of perceived dynamic efficiency and very low noise.
|A sequence of individual measurements made whilst varying a parameter of the stimulus. For example a frequency response Sweep would be made by measuring the gain of an EUT whilst varying the frequency of the stimulus. dScope provides a versatile sweeping capability, wherein many different Generator parameters can be varied whilst plotting up to four simultaneous Results. As well as being progressive, Sweeps can be table-based or sensed. In unusual circumstances which cannot be addressed within the sweep system, VBScripting allows automatic collection of sequential Results interspersed with any desired setting changes. Many tests which have traditionally been frequency-swept are now better performed using multi-tone techniques, which are much faster.|
|(Test & Measurement) a Signal Generator function often used as a stimulus in impulse response and acoustic testing (also known as a `chirp`) comprising a sine wave swept smoothly across the audio band. The rate of progress may be linear or logarithmic (a `log chirp`). The entire sweep is usually arranged to repeat exactly over the FFT Analyzer buffer acquisition period to allow synchronous (windowless) analysis and contiguous averaging, and often contains a period of silence between repetitions in order to prevent reverberance spilling between acquisitions.|
|A system which shares a common timebase from which to time events. Musicians playing "in time" could be considered to be "in sync". In this analogy, the drummer, a metronome or the conductor could be providing a "clock" to which the musicians are "locked". In digital audio, a system is said to be synchronous when different parts of the system share, and are locked to, the same clock.
|TEAC Digital Interface Format or TASCAM Digital Interface Format (TASCAM being a brand name of TEAC). A bidirectional multi-channel digital interface format first used on the TASCAM DA-88 digital multi-track recorder. It uses unbalanced connections on a 25 pin D-sub connector to transmit and/or receive up to 8 channels of digital audio between devices.
|(Scripting) a multi-threaded application has several threads running. Each thread performs a separate task or group of tasks, each one effectively running at the same time as the others.|
|An optical digital interconnect standard developed by Toshiba (TOSLINK = TOShiba Link) also going by the name "EIAJ Optical". It is most commonly used to carry an optical equivalent of an S/PDIF signal in consumer audio equipment such as CD and DVD players where it can also carry Dolby Digital or DTS encoded signals. While ADAT uses the same physical connectors and cable, it is incompatible.
TOSLINK is a registered trademark of Toshiba Corporation.
|?||Total Harmonic Distortion (THD)|
|Not to be confused with THD+N (Total Harmonic Distortion plus Noise), this is a specialised measurement of the level of the distortion harmonic components in a signal. Whereas the THD+N measurement filters out the fundamental and measures what is left, including noise, hum, etc, along with the distortion components, the THD measurement selectively band pass filters the distortion harmonics and measures them in isolation. The result is usually expressed in percent or dB relative to the fundamental and should state the number of harmonics included in the measurement.
|?||Total Harmonic Distortion Plus Noise (THD+N)|
|A measure of the distortion and noise in a signal achieved by measuring what is left after the wanted signal is filtered out. It is usually expressed as a percentage or dB ratio of the wanted signal.
|(dScope) One of the graphical plots displayed on the dScope`s Trace window. Traces can be of various types, e.g. Scope, FFT, Sweep, Limit Line, Filter, Window function etc. The user can select which Traces are to be displayed at any time, and in what colours etc. In two-channel mode, Traces for both Analyzer channels an be displayed simultaneously, either on the same or separate axes.|
|A device for converting energy between two different forms. A microphone converts acoustic energy to electrical energy, a loudspeaker converts electrical energy to acoustic energy.|
|?||Triangular Probability Density Function (TPDF)|
|A noise function whose a graph of probability vs amplitude is triangular. This type of noise is often used for dithering digital audio since it produces a precisely linear transfer function.|
|(dScope) The dScope`s FFT Analyzer is activated by a scope-like trigger. The trigger is sensitive to a user-defined threshold and transition polarity, and can be set to be iterative or single-shot. The trigger can also be over-ridden for continuous or manual operation if required. The Continuous-Time Analyzer runs continuously, independently of the FFT Analyzer's trigger.|
|A test signal consisting of two sinusoidal tones at defined frequencies and amplitudes, primarily used for measuring intermodulation distortion.
|The two's complement system is a way of representing negative numbers in binary.
|(Scripting) A Type library must be defined by any program that supports OLE Automation. The Type library is simply a definition of all the methods and properties that can be externally controlled|
|Sound at frequencies too high to be heard by the human ear, typically taken to mean >20kHz|
|A method of transmitting an analogue audio signal or digital audio carriers, where a single wire carries the signal with respect to a ground, or screen conductor. This method is more common in consumer equipment and is more prone to interference than the balanced method commonly used in studios. dScope`s Analogue Inputs and Outputs can work in balanced or unbalanced modes.|
|?||Unit Interval (UI)|
|A UI of an AES3 or IEC60958 carrier is 1/128 of the frame period, the duration of a single biphase-mark `cell`, or half a bit period.|
|?||Upper band-edge frequency|
|(Test & Measurement / AES17) The maximum frequency to be measured, which must be less than the folding frequency of the EUT in digital systems. In most cases, the upper band-edge frequency is 20kHz unless this is restricted by the sample rate of the EUT. The dScope implementations of the AES17 tests always set the upper band-edge frequency to 20kHz.
|(Digital Audio Interfaces) A per-channel, per-sample status bit in the AES3 interface. There are many different User bit implementations in use, some pseudo-standard (such as CD and DAT sub-codes and AES18 data transmission) and others which are entirely proprietary.|
|A per-channel flag bit carrier in the AES3 interface. The meaning of the Valid bit has changed slightly since the AES3 standard was originated, so unfortunately its implementation sometimes differs between equipment. In general, it indicates (when 0) that the channel is suitable for conversion to analogue, and most receiving equipment mutes if it detects the flag set to 1. However, in some instances it has been used to indicate that error correction or concealment has taken place, the effort of which may have been wasted if receiving equipment mutes as a result of seeing the flag. The dScope can set the Valid bits in its Digital Outputs and monitor them at its Digital Inputs.|
|(Scripting) Variables in VBScript are all stored internally as Variants. This means that the value can be of any type - e.g. integer, string or just an array of bytes. They assume a type when they are first assigned a value.|
|"Visual Basic Scripting Edition" is an active scripting language developed by Microsoft used extensively in server side internet scripting for ASP.
(dScope) A script or program in the VBScript language which customises an element of the dScope`s operation. This may be an Automation script, which allows the dScope to perform a pre-defined series of operations, or it may define various mathematical functions such as Weighting filters or Window functions.
|A dS-NET peripheral which adapts the dScope's AES3 outputs and inputs for connection to an EUT which has I2S-like serial audio multiplex interfaces. The multiplex is programmable to interface with a wide range of component-level devices.|
|A meter designed originally to measure and standardise the levels in telephone lines. It is defined in ANSI C16.5-1942, British Standard BS 6840, and IEC 60268-17.
VU meters typically have a range from −20 to +3dB where 0VU is +4dBu for a sine wave at 1kHz. The rise and fall times of the meter are both 300 milliseconds, meaning that if a constant sine wave of amplitude 0 VU is applied suddenly, the meter will take 300 milliseconds to reach the 0 on the scale. It behaves as a full-wave averaging instrument, and is not optimal for measuring peak levels. Its inability to measure peak levels accurately, critical in particular for digital recording, earned it the reputation as a bacronym for 'Virtually Useless'.
|A computer file format for storing digital audio waveforms. This is the default method of storing uncompressed digital audio on a Windows computer where the audio is typically encoded as linear PCM. The file format consists of a header which defines the data that follows (number of channels, sample rate, word length etc.) followed by the raw audio sample data.|
|(acoustics) the length of a cycle of an acoustic wave. It is given by λ = c / f where c is the propagation speed of the wave (approx 340ms-1 in air) and f is the frequency in Hz. This gives figures of wavelength of sound in air from about 17m at 20Hz to 17mm at 20kHz.|
|(dScope) A file containing a waveform that can be looped (played repeatedly in a loop) to create an arbitrary waveform.|
|(Test & Measurement / dScope) A filter which controls the emphasis of certain parts of the audio spectrum in the measurement. These filters are usually `standard responses` used to provide compliance with specific measurement standards, although FFT Detector Weighting filters can be user-defined with a VBScript.|
|Random noise with equal power per hertz. On an FFT derived spectrum, white noise will tend towards a flat line when averaged since each FFT bin has a fixed bandwidth and will contain equal power.|
|A weighting function applied to sample data before a FFT transform is applied in order to reduce the effects of discontinuities at the ends of the sample data.
|A Reference Sync signal in the form of an unbalanced square clock at the sample rate. It is nominally at TTL level with 75R impedance, on a BNC connector.|
|In computing and in digital audio, a 'word' is the natural unit of data storage. In a computer, this is typically 32 or 64 bits. In PCM digital audio, the sample data is stored in 'words', the length of which determines the resolution to which each sample can be defined. Common values for digital audio are 16 and 24bit.|
|The horizontal axis on a graph or trace|
|The vertical axis on a graph or trace|
|Zero weighting or flat response.
IEC 61672-1 defines sound level meters and mandates the A-weighting curve for any compliant sound level meter. In addition, it specifies a z-weighting which is a zero weighting or a flat frequency response. Essentially, sound level meters used to state "linear" or "flat" frequency response, and now they would state Z-weighted or FLAT. The implication is that a class-1 or class-2 sound level meter set on z-weighting will have a flat response within the tolerance implied by the class rating of the instrument.
|The point at which a waveform crosses the zero line, either positive going or negative going. When joining waveforms or editing waveforms that are intended to loop, cutting at a zero crossings (consistently either positive going or negative going, but not mixed) minimises distortion at the resulting join.|