Digital Audio

Constant Bit Rate

CBR; Codecs encodes data at a constant rate regardless of density of the audio file

Coaxial

Cable used to transmit data; Inner cable is surrounded by a plastic insulator, which is surrounded by a wire mesh conductor that insulates the internal signal wire from external interference and an outer casing that functions as a ground

CPU Buffering

Channels are processed one at a time and the results are stored on multiple CPU buffers that alternately send data as DAW requests the data for playback; First few seconds are relayed to the buffer prior to processing and playback and it continuously buffers after that; Effective streaming technique but is very susceptible to latency

Decoder

Circuit that interprets the meaning of the symbols as they were chosen and arranged by the encode

Sample-and-Hold

Circuit that seizes voltage values with each tick of an A/D device's internal clock

AoE Formats

CobraNet; EtherSound; Dante; AVB (currently under development)

Codec

Computer program or device capable of encoding and/or decoding a digital data stream with the end result being a reduced file size

Sawtooth Wave

Contains all even and odd harmonics associated with a fundamental tone, making it a rich source for modeling other sounds; Amplitude of each overtone decreases exponentially as a ratio of the harmonic's frequency to that of the fundamental

Conversion Buffering

DAW's and software must buffer when converting or bouncing and this latency can add up if not monitored

Optical Cable

Data is transmitted over fiber optic lines; Uses a TOSLINK connecter instead of an RCA type; Can transmit multi-channel audio; Not susceptible to ground hum and loops; Able to support far higher rates of data transfer over greater distances than coaxial cable

Lossless

Data reduction technique that does not effect quality of original audio; No effect on original quality; Typically around 50% reduction; Exact reconstruction of digital code for the audio signal

Lossy

Data reduction technique that selectively removes original information in order to significantly reduce the file size; Some data is lost; Files can be reduced up to 99% in size (90% with no perceived sound quality loss); Bit rate effects the perceived quality of reduced audio file

Audio over Ethernet (AoE)

Data transmission protocol over which computer network traffic travels; Poorly suited to real-time transmission but numerous attempts have been made to harness the technology because of its flexible routing; Uses simple Cat5 cabling; Resists interference and attenuation

dBFS

Decibels Full Scale

Sample Rate

Defines the number of sample per second taken from a continuous signal to make a discrete signal; Governs the frequency response of digital audio

Logical Format

Describes acceptable data, performances both offered and essential for a disc player, and the complete user experience

Physical Disc Format

Describes various optical disc characteristics including the size and shape of the disc, the size of pits, the speed at which the disc spins, and a multitude of aspects regarding the specifications of the player itself

Jitter

Deviation from a normal, steady pulse or tick of a clock that contributes to misrepresentation of a signal; Result of small timing irregularities that become magnified during the transmission of digital signals as the signals are passed from one device to another

Index of Reflectivity

Difference in brightness between land and pit on a CD Physical Format

Pulse Width Modulation

Digital (binary) measurements of how long each pulse is either on or off; Width of increasing voltage or decreasing voltage is assigned a 1 or 0 respectively

D/A Conversion Signal Flow

Digital Word -> Series of Resistors (each with assigned charges) -> Sample-and-Hold Circuit -> Anti-Imaging Filter (Smoothing Filter) -> Reconstructed Sample

Delta-Sigma Modulation

Digital and analog processing capability is combined on a single microchip allowing for 1-bit resolution at high sample rates

Quantization Intervals

Discrete incremental distinctions made between the value of one sample and the next; Breaks down bit depth into a series of evenly spaced intervals

European Broadcasting Union

EBU

Edit Decision List

EDL; Final list of samples used in the audio editing process; Identified by time code

6 dB

Each bit in the bit depth is equal to a _____ increase in dynamic range

Dolby 7.1

Eight channel digital surround sound system by Dolby

Photoreceptor

Electromagnetic receptor that detects the radiation known as visible light

Anti-Aliasing Filter

Eliminates frequencies above the Nyquist limit from becoming samples; Occurs prior to quantization

Claude Shannon

Father of modern information theory; Solidified the Nyquist Theory by adding the concept that bits per second (binary representation of audio signals) must be at equal intervals to accurately represent data

Harry Nyquist

Found that aliasing was always a problem no matter how fast you sample; Less data recorded but more accurate; 2 samples per wave length.

Significand

Fractional part of a floating-point number; Also called the mantissa; Defines precision

Joseph Fourier

French mathematician that noted that any complex sound can be broken down into a series of component pure tones

dB/FS

Full Scale; Type of metering that measures level in digital recording system; Recording and Mixing levels should NEVER exceed 0dB FS in digital audio or clipping will occur

Nyquist Frequency

Governs the frequency response of a digital system; The highest-frequency component that can be captured with a sampling rate; always 1/2 of sampling rate; Also called the limiting frequency

DVD-Audio

HD Audio format; Lossless Compression; 24-bit/96 kHz; 5.1 Surround or 24-bit / 192 kHz stereo sound

Square Wave

Have odd numbered harmonics

Sinusoidal

Having a repeated succession of waves or curves as in a sound waveform

Compression

High Pressure, Part of a longitudinal wave where the particles of the medium are close together

Multichannel Audio Digital Interface (MADI)

High channel count; 64 channels on one cable; Coaxial cable with BNC connector or fiber optic with ST1 connector

2 Dimensions of Sound

How Loud (Y-Axis) & How Fast (X-Axis)

Sampling Theorem

If a signal is sampled at a rate higher than twice the highest significant signal frequency and at evenly spaced intervals, then the samples contain all the information of the original signal

Acoustics

Pertaining to hearing or sound; Combination of the intensity of air pressure molecules with amplitude

Buffering Locations

Playback; I/O Connections; CPU (Streaming); Conversion from DAW or Software

Additive Synthesis

Process of building a complex tone by starting with the fundamental frequency and adding pure tone harmonics

Spectrum Multiplication

Process that begins with a fast FFT analysis of the spectra of two input signals, then the multiplication of like frequencies, and IFFT to finalize the process

Perceptual Coding

Psychoacoustic model of data reduction used for general audio compression that aims to transmit only features perceptible to the human ear; Useful for mastering streaming media

Buffering

RAM holds in memory audio data before it is transferred to the memory controller; Certain amount of data is processed before large amounts of data are streamed to prevent latency

Voltage

Rate at which energy is drawn from a source that produces a flow of electricity in a circuit; Expressed in volts

Signal-to-Noise Ratio

Ratio of magnitude of the analytical signal to the magnitude of the background noise signal

Decimation Filter

Reduces A/D sample rate from the oversampled rate to nominal rate by turning series of one-bit samples into a series of multi-bit PCM samples; (ex. => 2.8MHz sample rate converted to 44.1kHz and simultaneously converts 1-Bit samples to multi-bit); Low Pass Filter finishes the process

Dithering

Reduces the percieved distortion due to quantization error; Low level white noise source is introduced to make the profile of the quantization "noise" more irregular; Useful when reducing the number of bits per word in a signal (i.e. when converting 20 bit to 16 bit resolution); Introduced at 1/2 of quantization interval when bouncing at shorter word length

Aliasing

Improper (false) sampling of high frequencies that cause much lower frequencies to be produced when the audio is reconstructed; Foldover happens at higher frequencies as harmonics reach levels that exceed the Nyquist limit.

Playback Buffering

In order to play multiple channels at one time data is buffered as read to disk; Latency happens between play command & beginning of playback

Interpolation Filter

Increases D/A sample rate from nominal rate to oversampling rate by turning series multi-bit PCM samples into 1-bit samples; (ex. => 44.1kHz sample rate converted to 2.8MHz and simultaneously converts multi-bit samples to 1-bit); Low Pass Filter finishes the process

Normalizing

Increases or decreases the digital signal so that the loudest sample is brought up to 0dBfs; Uses all bits from dynamic range and makes it even from track to track

Pulse Density Modulation

Reference voltage determined by summing the voltage values of a predetermined number of previous samples; Numbers of 1 in row = waveform peak; Numbers of 0 in row = waveform trough; Basis of Sony's Direct Stream Digital (DSD)

Recording Levels

Industry Standards: -6 dB Peak = -20 RMS Meter

Stapedes Reflex

Inner ear component that attaches to the stapes and helps to decrease the amplitude of vibrations; Causes the masking phenomenon

Anti-Imaging Filter

Removes high frequency images and noise and smoothes the stair case output coming from of the sample and hold circuit; Also called a SMOOTHING FILTER

Intensity Stereo

Joint-Stereo Technique; Since the human brain is unable to localize sounds at high frequencies well sounds above 9 kHz threshold are encoded in mono

Pad Head & Tail

Leaving space at beginning and ending of song for data crunching during mastering; Last step in mastering process

Threshold of Pain

Level above which audible sounds are painful (125 - 130 db)

Blu-Ray

Lossless Format; Can hold up to 25GB on a single-layer disc and 50GB on a dual-layer disc

Rarefaction

Low Pressure; Part in a longitudinal wave where the particles are spread apart

Motion Pictures Experts Group

MPEG; Standardizing body of audio coding

Quantization

Represents the amplitude component of the digital sampling process; Technique of incrementing a continuous analog event into a discrete set of binary digits (bits)

RMS

Root Mean Square; Refers to taking the square root of all instantaneous amplitudes; Takes the average of those squares; (-6 Peak Level is approximately equal to -20 RMS)

Speed of Sound

Roughly around 1,130 ft/s

Equal Loudness Contour

Measure of sound pressure over the frequency spectrum, for which a listener perceives a constant loudness when presented with pure steady tones

Intensity

Measure of the amplitude of a longitudinal wave

Sampling (Samples)

Measurement at regular intervals of the amplitude of a varying waveform (in order to convert it to digital form); There must be a minimum of 2 samples for each cycle in a waveform

Peak Level

Measures the highest levels of a signal being recorded or mixed; Monitors for clipping, which occurs at 0dBFS); Does not always reflect perceived volume of signal

Quantizer

Measuring equipment in A/D conversion that processes voltage and provides a value for that voltage

Digital

Method of representing an acoustic quantity with a series of binary numbers; Can have only specific individually distinct values

Oversampling

Method of sampling data at a higher resolution (higher sample rate) as a means of reducing harmonic content during D/A conversion; (x2) oversampling gets rid of all odd harmonic content

Pulse Code Modulation

Method used in digital recording and reproduction in which a signal is sampled at various points and the resulting value is translated into binary numbers

Interleaved

Mixing data and control characters in a single operation

Amplitude Accuracy

More accuracy in low amplitudes and less in higher amplitudes

Joint Stereo

More aggressive lossy data reduction techniques that require further manipulation of the stereo field; Examples are "Intensity" & "M-S"

Entropy Coding

Most significant lossless coding technique in current use; Measure of disorder in which long strings of data are represented by short symbols and uses the shortest symbols to represent the most common repetitive audio data maximizing data reduction

PCM

Multi-Bit Words; (Pulse Code Modulation)

Bit Rate

Number of bits per second processed when sampling sound; (Sampling Rate x Bit Depth) = Resolution

Bit Depth

Number of bits used to represent the smallest unit of information in an audio file; Greater bit depth = better quality audio

Exponent

Number or variable that represents the number of times the base of a power is used as a factor; Defines magnitude

I/O Connection Buffering

Occurs as data is assembled into meaningful bits or information and as left & right channels are separated

Base 2 System

Only 2 digits used; The value of each place (ones, hundreds, etc.) are as follows from greatest to least: 128, 64, 32, 16, 8, 4, 2, 1

Harmonic Content

Overtones that contribute to the timbre of a sound and make up a complex waveform's physical characteristics

Masking Analysis

Perceptual coding technique that uses louder sounds of a similar frequency to decide what information is to be saved during data reduction

A/D Conversion

"Capturing" part of digital audio; Never captures a signal perfectly

D/A Conversion

"Reconstructing" part of digital audio

AES3

(AES/EBU); 110Ω, 2-channel balanced digital audio cable with an XLR connection; NOT a mic cable!!

Y-Axis Terminology

(Amplitude Based) Amplitude: Voltage; Quantization; Bit Depth; Quantization Intervals; Quantization Noise; [Signal:Quantization Noise Ratio]; Dither; Dynamic Range

X-Axis Terminology

(Time Based) Frequency: Aliasing; Anti-Aliasing Filter; Sample Rate; Nyquist Limit

Storage Conversion Steps

1.) Bit Rate x Sample Rate (you'll get b/sec) 2.) Multiply by 60 if converting seconds to minutes 3.) Divide by 8 to convert bits to Bytes and get B/min 4.) Divide by 1,024 to get KB/min and keep doing it until you get desired bit rate specification (result should be under 1000) 5.) Multiply number of stereo channels 6.) Multiply length of time

Requirements for A/D Conversion

1.) Taking a series of evenly-spaced measurements 2.) Signal contains no frequency components higher than half the sample rate

DVD-14

12.33 GB; DS/ML

Red Book

12cm plastic disc; 1.2mm thick; One-sided; Red Laser; 1.6 microns between tracks; 125 nanometer pits

DVD-18

15.9 GB; DS/DL

Requirements for CD Audio

16-Bit; 44.1 kHz; PCM; Stereo

CobraNet

1st commercially successful AoE format for the transmission of digital audio, video, and control signals over 64-channel 100Mbps Ethernet networks

DVD-5

4.38 GB; SS/SL

DVD-9

7.95 GB; SS/DL

Tascam Digital Interface Format (TDIF)

8-in/8-out on one cable; 25-pin D-sub connector

DVD-10

8.75 GB; DS/SL

Impulse Response

A digital filter's time domain output sequence when the input is a single sample is input

Blu-Ray

A drive that can read and write on optical media that hold up to 50 GB on two layers; 24-bit/96 kHz for 8-Channel; 24-bit/192 kHz for 6-Channel

Cutoff Frequency

A frequency specified for a filter (digital or electronic) the marks the point at which the frequency content of a signal is altered +/- 3dB

Floating Point

A method of representing real numbers using a mantissa and an exponent

Ethernet

A network communications protocol that specifies how machines will exchange data; Uses a broadcast system in which one machine transmits its message on the communication medium and the other machines listen for messages directed to them

Convolution

A sample-by-sample operation on two signals

Overflow

A situation where a calculated value cannot fit into the number of digits reserved for it

Word Clock

A time regulator that makes all samples and bits to align when working with interconnected digital devices; Basically a signal that all of the digital devices refer to when operating.

Lossy Formats

AAC (Advanced Audio Coding); MP3; RA; WMA; OGG Vorbis; Dolby Digital/AC-3; DTS; ADPCM

Average Bit Rate

ABR; Codecs that encode data by determining how dense or sparse areas of the audio are while also keeping bit rate within specified limits to avoid rebuffering

Lightpipe

ADAT Optcal; 8-in/8-out on two cables; Fiber-optic, TOSLINK connector

Adaptive Pulse Code Modulation

ADPCM; Pulse code modulation that produces a digital signal with a lower bit rate than standard PCM; Records only the difference between samples

Audio Engineering Society

AES

Fidelity

Accuracy with which an electronic system reproduces the sound or image of its input signal

M-S Stereo

Algorithm uses matrix of a mid/side microphone pair to determine a side signal & that signal is reduced then distributed as code in stereo

Footroom

Allowance of noise floor below that which is required for the final product

Oversampling

Allows for an internal sample rate at multiples of the input and output rates; Alleviates the need for steep "brickwall" filters; Often combined with internal "1-bit" processing; Increases smoothing effect

Fletcher-Munson Curve

Also known as equal loudness curves; Graph that indicates the average ear sensitivity to different frequencies at different SPL levels (as volume increases, these curves flatten out)

RMS Meter

Amplitude meter that takes the square root of all instantaneous amplitudes and averages them to find a mean and squares that value; Useful with particularly complex waveforms

Redither

Anytime bit depth is reduced the gap gets bigger so more dithering is required

Lossless Formats

Apple Lossless; Windows Media Lossless; DTS HD Master Audio; Dolby True HD; FLAC

Sample Rate Effect on Anti-Aliasing

As sample rate is increased more room is created for a smoother slope of the attenuation band because Nyquist limit extends well beyond range of hearing with each increase

Effective Bit Depth

Based on Full Scale (dB/FS); -6dB represents a loss of one bit so account for this when calculating

Subbands

Based on psychoacoustics, these are the basis of frequency analysis for a perceptual codec;

Psychoacoustics

Branch of psychology concerned with the subjective perception of sound

Headroom Bits

Built into DAWs; Bits are added when signals are mixed together to avoid clipping

Foldover

Same as "aliasing"

Non-Compressed Audio Data Rate Formula

Sample Rate x Bit Depth x # of Channels

Noise Shaping

Samples are duplicated and the playback sampling rate correspondingly increased; Significantly raises the Nyquist limit to a range well beyond human hearing; Processing "pushes" the distortion resulting from quantization error into these higher frequency ranges, rendering it less "visible" to our ears

SCMS

Serial Copy Management System; main difference between AES3 & S/PIDF

Morse Code

Series of dots and dashes representing the letters of the alphabet; Most common letters are represented by the shortest dots and dashes; Example of entropy coding

Direct Monitoring

Signal conversions are mixed with playback tracks resulting in near-zero latency

Analog

Signal that uses variable voltage to create continuous waves resulting in an inexact transmission

Successive Approximation

Signal voltage is relayed to a register from sample-and -hold circuit; Holds reference frequencies in binary form that decrease in value; Finds approximated value & assigns binary number accordingly

Sony-Philips Digital Interface Format (S/PDIF)

Single-pin RCA cable or fiber-optic TOSLINK connector used for digital transfer; 75Ω coaxial, 2-channel unbalanced; "Consumer" format of AES3

Dolby 5.1

Six channel (five speakers and one subwoofer for bass) digital surround sound system by Dolby

Resolution

Smallest interval measurable by a scientific instrument; Defined by bit rate (sample rate x bit depth)

Frames

Snippets of time in which frequency analysis takes place in a perceptual codec

Threshold of Hearing

Softest sound that can be heard by the average human ear (0 dB)

SACD

Sony and Philips optical disc format; Utilizes sigma delta DSD to offer higher resolution; 1-bit; 2.8224 MHz; 6-Channel

Direct Stream Digital

Sony sigma-delta modulation based technology that bypasses the decimation and interpolation steps found in PCM converters

Algorithm

Specific set of instructions for carrying out a data reduction technique that determines how to "save" binary data information efficiently

Zero-Latency Monitoring

Splits the input signal and mixes it with an analog copy so that no latency is present

Inter-Channel Redundancy

Stores only one copy of a stereo signal and assigns it to both channels in order to save 50% of original bandwidth

Decimal-to-Binary Conversion

Subtract place values from the decimal number and place ones or zeros in the correct places

Fourier Series

Sum of all harmonics; Sum of sine and cosine waves which have frequencies f, 2f, 3f, 4f...

MONO

Take up half as many bits (50%); Algorithm can keep same quality by lossing the stereo track

Internal Resolution

The ability of a digital system to perform complex DSP without running into problems with overflow or loss of resolution

Oscillation

The act of a frequency swinging back and forth with a steady, uninterrupted rhythm

Spectra

The amount of energy at each wavelength

Gain Staging

The art of deciding where to place a processor in signal flow based on how that processor will be influenced by the other processors in the path

Attenuation

The continuous loss of signal strengths as a signal travels through a medium

Quantization Error

The difference between the analog value and the approximated digital value due to the "rounding" that occurs while converting the analog signal to digital

Dynamic Range

The difference in volume between the loudest and quietest sounds of a source

Latency

The elapsed time it takes for a packet of data to arrive at its destination; Lagging or pause of an audio signal as digital processing occurs; Can be managed utilizing several forms of "audio monitoring"

Cutoff Frequency

The frequency above or below which attenuation begins in a filter circuit

Pass Band

The frequency range that is allowed through a filter

0 dB FS

The loudest point of a Full Scale system

Digital Signal Processing

The mathematics, algorithms, and the techniques used to manipulate signals after they have been converted to digital form

Bit Depth Effect on Dynamic Range

The more bits allocated during quantization, the more accurate the measurement

Frequency

The number of compressions or rarefactions in one second; The higher the frequency the more compressions & rarefactions per second; Measured in Hertz

Compression

The process of reducing the space required to store data by efficiently encoding the content.

Transfer Protocol

The set of rules that computers use to move files from one computer to another on an internet

TOSLINK

Toshiba developed digital audio interface utilizes fiber optics as a transmission medium.

Sampling Rule

Twice as many samples as the highest frequency at minimum; Sampling rate totally controls frequency response

EtherSound

Ultra low-latency, 512-channel (on a gigabit network), less flexible AoE format; Routed like audio cables...not network cables

Nanometer

Unit of measurement that is equal to one billionth of a meter

Micron

Unit of measurement that is equal to one millionth of a meter

Glass Master

Used as the main disc from which other discs are made; Composed of ground glass with a very fine photoresistor layer; An imaging laser burns pit and land patterns in preparation for duplication

dB/SPL

Used when the reference pressure of a sound is 20 microPa (0.00002); Sound Pressure Level; Measure of amplitude

Data Packing

Uses entropy coding as the basis; Computer data compression algorithm that packages files such as .ZIP & .RAR

Variable Bit Rate

VBR; Most common & best data reduction technique; Codecs that encode data by determining how dense or sparse areas of the audio are; Can result in buffering issues

Class-D Amplifier

Very quiet digital amplifier that produces a series of output pulses with the audio signal coded the same as the width of the output pulses; Pulses are used to represent wave forms and are either on or off; Intense signals have long pulses with short gaps; Cheaper, cleaner and used widely

Low-Latency Monitoring

Very selective method of lowering buffer levels by halting different levels of audio processing

Sonogram

Visual graph that shows how loud a sound is at different frequencies

A/D Conversion Signal Flow

Voltage -> Dither -> Anti-Aliasing (Low Pass Filter) -> Sample & Hold Circuit -> Successive Approximation/Quantizer --) 100111010 (PCM Audio File)

Sine Wave

Waveform of a pure tone showing simple harmonic motion

Buffer Size

When recording you want the smallest buffer available; When mixing you want the largest buffer available