153 terms

Digital Audio Technology Final Exam

the rate at which sound oscillates, measured in hertz.
the time it takes for one full cycle to occur
cycles per second
measurement of air pressure for sound that humans can healthily.
the amount of change in air pressure
speed of sound travels at 1,130 ft/sec divided by frequency.
Simple Waveforms
only generate energy at a single frequency,
Complex waveforms
sounds that contain more than just a single pitch (harmonics)
sound is represented electrically, (compression is positive going voltage, rarefraction is negative-going voltage)
carries the sound and is the electrical equivalent of the air particle motion of the sound in acoustical form.
Ohm's Law
Voltage equals Current multiplied Voltage. V=IR
Represents the level of a signal above or below reference point. The lowest is 0 db.
A doubling of power
3 db increase
A doubling of voltage is
6 db increase
measuring units used for power
Inverse Square Law
Doubling the distance decreases the sound Pressure by 6 db. Halving the distance increases it by 6 db.
In Phase
When two waves of the same frequency, with their compressions and rarefractions, coinciding exactly
Out of Phase
when the half cycle of one compressions aligns with the rarefraction of the othe wave, in the same time and space
180 degrees out of phase
When this happens, the waves cancels each others out, resulting in no input
deals with time alignment of two or more waves of the same frequency
deals with the relationship of positive and negative voltages.
When two waves with the same phase and polarity
the amplitude is doubled
Sound Pressure Level
used to measure how loud a sound is based upon a reference of 20 micropascuals.
SPL Weighting Filters
these are meters that have built in curves as options to mimic how humans percieve loudness
A Weighted SPL Filter
boosts low and high frequencies and is reported as "dBA"
C weighted
rolls off the very extremes, and is very accurate.
Noise Floor
when the inherent noise of a system becomes audible, this is the noise floor.
Dynamic Range
The area between the noise floor and the onset of distortion.
Dynamic Range Point
0 VU, masks the noise floor, and is still far away from the onset of distortion.
Signal to Noise Ratio
Distance between the noise floor and 0 VU
Head Room
Distance between 0 vu and the onset of distortion.
Digital System
on/off two states, binary code
Binary Counting
There are coloumns for ones, twos, fours, eights, sixteens, thirty-twos, sixty-fours, one hundred 28, etc.
Made up of 8 bits. (B)
One Digital Binary Digit
a device that converts one type of energy into another
Sound is recorded
by converting continuous variations in sound pressure to continuous variations in electric voltage.
magnetic recording, a length of plastic covered with a magnetic coating. The tape machine head is basically a big coil. When the electrical current, flows through a coil, a magnetic field is created.
Digital Recording
Continuous variations in sound pressure are stored as binary numbers, which represent the amplitude of a signal.
Digital Replay
can differentiate between the what was originally captured and what is replayed, unwanted noice becomes discarded.
3 Components of DAW
Audio Sources, Signal Flow/bussing, processing/topology, the hardware
Audio System
governs the way the computer records and processes Audio Signals
Native Audio System
uses the computer CPU to process Audio, sometimes introduces latency which can cause an audible delay, greater potential of of audible delay when recording.
Digital Signal Processing
systems use dedicated hardware to process audio signals, usually conncected via a PCI Slot, lower latency,
Audio Interfaces
allows audio to get into the computer, and let it leave computer,
When the AD converter meausres the amplitude of the incoming signal at specific points in time.
The system assigns the closest set value.
Quantizing Error
The difference between the actual location of the sampled signal and where it ends up after being quantized
Signal to Noise Ratio
the level of wanted signal to the amount of of unwanted noise introduced by the recording system, dependent on bit depth resolution.
The greater the SNR (sound to Noise Ratio)
the cleaner the system
To get the SNR
Multiply the number of bits by 6. (Example 8 bit word = SNR 48, 16 bit word =SNR 96 db 24 bit word = SNR 144 dB)
If too few samples are taken per second
The audio is misrepresented, (Example: 7khz sine wave,3 khz wave, 13hz wave below)
If too few samples are taken per cycle, the initial signal may be misrepresented and reproduce a different audio signal upon output.
How do you avoid Aliasing?
The sampling rate must be equal to atleast 2x the highest audio frequency te system has to reproduce. Because you need at a minimum, one sample for the negative, and positive amplitude.
Keep Aliasing Inaudible
The highest frequency of the original audio signal must not extend past the Nyquist frequency (1/2 sampling rate). Or put a low pass filter on the SR.
Plug In
A software that runs inside another software, in audio applications in real time and non real time.
A device that reduces or attenuates the volume level of a range of frequencies.
Active Filter
A filter combined with an amplifier, allowing frequencies to be boosted.
Passive Filter
Filter that can attenuate frequencies
Pass Band
The portion of the frequency spectrum that are unaffected by the filter.
Stop Band the
range of frequencies Affected by the filter
Band Reject Filter
attenuates a band of frequencies allowing frequencies low
Low Pass high Cut Filter
A filter that attenuates the frequencies above a cutoff frequency and passes those below through unaffected.
Cutoff Frequency
The frequency where the output level of the filter is attenuated or boosted 3 db compared to the input level.
A circuit that contains one resistor and one capacitor. A one pole circuit functions a low pass or high pass filter with 6 dB of attenuation per octave. (2 pole filter is 12 db/octave, 3 pole is 18 db/octave etc)
A reference that corresponds to the number of poles in a filter, referencing 6db per octave. (A second order filter attenuates at 12 db/per octave.)
Hz on Eq
A parameter on EQ that determines center frequency of the signal to be affected, at the cut off frequency the points are attenuated +/- db.
Gain on Eq
Parameter that determines the amount of increase or decrease implemented into the audio signal.
Determines the frequency range
Cutoff Frequencies
the points ath are +/- 3 db above or below the center frequency
Low Filter
a filter that can boost or attenuate frequencies below a cutoff frequency and pass those above unaffected.
High Shelf Filter
a filter that can boost or attenuate the frequencies above a cutoff frequency and pass those below through unaffected.
Turnover Frequency
The 3 db up or down point (similar to that of a Cutoff Frequency)
(A device that acts as an automatic level controller) reduces the dynamic range of asignal. (Keeps the loudest of a portion of a program within a tolerable level difference from the softest portion of a program.)
an input signal exceeds a certain threshold, and then kicks in.
indicates how many decibels of level reduction once it exceeds threshold. (Example for every 3 db above ratio, the output will only be 1 db above threshold)
the amount of time between when the signal crosses the threshold and when the compression kicks in.
persistence of sound in a space after the original sound source has been removed or stopped.
RT60 Decay
How long the signal takes to decay 60 db, calculated using Sabines formula as 0.049 V/A
Sound Absorption
INdicates the proportion of sound which is absorbed by the surface compared to the proportion that is reflected back into the room. (open window absorption coefficient of 1, an acoustic mirror is very close to 0)
difference in time of arrival between direct sound and the subsequent first association reflection.
sets the degree to which initial echo density increases over time, high setting result in high initial build up of echo density
Peak Meters
tells you how far the voltage swings, show the signal level at an instantaneous point in time, how high or low the voltage is at that moment. Doesnt accurately display voltage, Sine waves would always average 0.
RMS Measuring
more precise measurements, squares the amplitude to be measured, finds the average value. For a sine wave RMS voltage is equal to .707 of the PEAK Voltage, For a square wave RMS Voltage is eual to Peak Voltage.
VU Meter
volume unitsaverages the absolute values of voltages, fairly accurate representation of percieved loudness changes.
decibel full scale, also called a digital peak meters, highest leel is 0dbFs the digital ceiling. If the level passes 0, the signal is distorted. All values are preceded by a minus sign.
0 dBu on a DbFS
equals 0.775 units.
Interfacing Analog and Digital
requires references for both to be indicated for consistency Between equipment. (-20 dbFS = 0 VU)
The first reverb effects created for recordings used a real phsycial space as a natural _ _
echo chamber
Plate Reverb
uses a clectromechanical transducer, to create vibration in a large plate of sheet metal. A pickup captures the vibrations as they bounce across the plate.
Spring Reverb
uses a transducer at one end of a spring and a pickup at the other, to create and capture vibrations within a metal spring.
Digital Reverb
use various signal processing, algorithms in order to create reverb effects. Use multiple feedback delay circuits to create a large decaying series of echoes.
Inpulse Response
Represents the reaction of a particular environment, to sound, can be generated with a provided isolated transient, frequency sweep.
Convolution Reverb
Used to artificially recreate the reverberation of a room through convolution between an impule and the response of the room generated by it.
AES/EBU Interface
Allows for 2 channels of digital audio to be transferred Serially over a balanced line, sampling rates up to 96 khz and bit resolutions up to 24 big. Sync info can be transmitted into connection, no seperate word clock needed. Nominal input impedance of 110 ohms. (Looks like XLR cable)
Allows for 2 channels of digital audio to be transferred serially over an unblanced line. Uses an RCA connector, capable of sampling rates up to 96 khz and bit resolutions of 24 bit. Nominal Imput Impedance of 75 ohms.
Optical Cable
Data is transmitted over fiber optic lines. Just like SPDIF, but uses a TOSLINK connecter instead of an RCA type. TOSLINK can transmit a 2 channel PCM audio stream. Can transmit multi channel audio such as Dolby Digital. Optical cables are not susceptible to ground hum and loops.
Mini Toslink
Usually found on labtops, wired to accomodate an analog headphone out or an optical stream.
ADAT Lightpipe
Optical standard that allows for full fidelity (non data reduced) audio transmission. Allows 8 channels of 24 bit, 48 khz audio to transmit across one optical cable.
Multichannel Audio Digital Interface, allows a high number of channels to be transmitted across a single line. either 28, 56, of 64 channesl across a single line. Used for connecting digital consoles or transmitting in live sound situations. Can transfer up to 100 meters.
Tascam Digital Interface, 8 channels of digitial audio in 2 directions using a 25 pin D-Sub Connector.
Also known as a container, which dictates the way data is stored on a file.
Pulse Code Modulation Files
the main uncompressed audio type, referred to as RAW files snce there is no header information. Often unreadable by audio programs. to read, (they're usually wrapped as WAV, AIF, BWF)
Uncompressed File Sizes
require the same amount of storage regardless of content.
developed by Microsoft, PCm format has a header and body.
Audio Interchange File Format, Apple, header and body.
Broadcast Wave Format
like the Wav, but has extended information in the header allowing it to contain Metadata, the new standard uncompressed audio file format.
data about data, provides information about the data portion of the file. (album song title genere)
Lossless Audio Compression
allows for the quallity of the uncompressed format but with a reduced file size. (needs time to analyze, encode, and decode information) (AIF-C, Apple Lossless, FLAC, WMA, WavPack)
Lossy Audio aka Data Reduced
audio format that analyzes the sonic make up of a piece of audio to determine which poritons of the frequency spectrum can be altered. Resulting in a smaller file size, reduced fidelity. (Doesn't sound as good)
Lossy Audio Formats
AAC (Advanced Audio Coding), M4p, RA, WMA,
codec, time ellement associated with achieving compressed audio, Higher quality encoders will need to scan data twice to determine where and what it can alter.
Constant Bit Rate
econdes the data at a constant rate regardless of density of the particular part of media.
Variable Bit Rate
encodes data by determining how dense or sparse areas of the audio are. (most common) and best results.
Id3 Tag
a form of metadata containing info such as song name or writers.
Bit Depth determines
The resolution of the audio.
What does Quantization do?
assigns a numeric value to the audio signal
Quantization Error
If a value is between one step or another, an inaccurate representation of the original signal.
Quantizing Distortion
sharp cut offs in sound or toggling between level and "no leve" (with low level signals)
white noise is introudced to the signal during quantizing, to remove quantizing distortion. Mathematically removes the harmonics, or other highly undesirable distortions entirely and that replaces it with a constant fixed noise level.
Quantizing a signal again, to change the bit depth, must always use dither to mask errors.
Word Clock
A time regulator, that makes all samples and bits to align when working with interconnected digital devices; a signal that all of the digital devices refer to when operating.
Internal Clock as a Master
Digital Console (internal) Master and Pro Tools, and Digital Reverb Unit (Word Clock in) SLAVE.
Word Clock Cable
BNC (funny silver cable) plugs in to Consoles, and Pro Tools.
AES Digital in Sync
Looks like XLR, plus into Digital Reverb unit.
Master Clock
external word generator
Exterior Generator
Master Word Clock and everything else is a slave.
the difference between when the ideal square wave and the actual square wave changes. The variation in timing of a periodic event (signal transition)
When should you use a clock?
When connecting 2 or more digital Devices, set one to master and the other to slaves.
Connection point for everything. (Processors RAM, Hard drives, communication port, expansion cards)
PCI type interfaces
Connects to the computer via PCI Bus, which connects directly to the motherboard. The Audio I/o Connections can be apart of the card or attached via cable like Pro Tools.
Random Access Memory, serves as a buffer between hard drive and the audio engine.
Digilink Cable
A PCI interface, a protools connect. (AVID HD I/O)
Analog Interfaces
The analog signal must be converted into digital signal, then when outputed, the digital signal must be converted back into analog signal. (at the mercy of the converters) (96i)
Digital Interfaces must
be conscience of clocking, otherwise you'll get unwanted artifacts.
PMCIA/PCI Express Interfaces
Personal Computer Memory Card International Assosciation, Labtop version of PCI connections, usually need an adapter since most audio interfaces are PCI based.
Firewire Interfaces
the faster better choice for larger I/O options.
Plug In Formats
Real Time (native & platform) and NOn Real Time
Real Time Processing
performs audio digital signal processing, usually from an effects box or outboard signal processing. (high pass filter, flanger, compressor) (takes up system usage)
can also be a musical instrument or software synthesizer.
Non Real Time Processing
Effect is rendered with the oriinal sound source to create a new file that is a mix of the original sound file and the effect. (saves CPU resources)
Disk Formatting
breaks up the space on a disk drive into smaller areas. (Formatted capacity of a disk is smaller than an unformated) Writes boundaries into sectors and tracks.
Low Level Formating
when the disk drive writes sector & track information to a disk.
Disk Reinitialization
the user tries to resotre the drive back to its factory configuration, and identify the bad blocks, and zero-filling.
High Level Format
sets up an empty file system, no new data is written, but the directory allocates all of your space as avialable.
Disk Fragmentation
reduces writing and reading performances, because of jumping holes, can cause pro tools error.
RAM buffering
A place where data is temporariy stored until needed. (A large buffer, small number of channels and the sampling rate determind how much of a RAM Buffer is needed)
Error -0693 "the operating system held off interupts for too long"
Change the buffer size, to keep from having too long of a delay.
Recording Analogue Sound
Sound is recorded by converting continuous variotions in sound pressure to continuous variations in electric voltage. The tape machine head is a big coil. When electrical current, flows through coil a magnetic field is created. The magnetic flux, changes this into an electrical version of the signal that was on the tape.
Audio Drivers
a translator between a hardware device and the applicatoins or operating systems that use it. (allows programmers to write software independent of a specific hardware device)
act as the glue between hardware devices and operating systems (Windows Mac) or are used to refer to system extions that allow applicatoins to pass audio to other applications. (Application Programming Interface)
The hardware Abstraction Layer, that allows programmers to write device independent software.
Aggregate Device
combines the inputs and outputs of multiple devices to appear as a single device.
Digital Signal Processing Systems
DSP dedicated hardware to process audio signal