Digital Basics Flashcards
Analog to digital converter
converts an analog audio signal to digital; analog signal –> anti-aliasing –> PAM (pulse amplitude modulation) –> sample and hold –> quantizing –> PCM (pulse code modulation) –> storage
Digital to analog converter
converts a digital audio signal to analog; storage –> reconstruction –> resampling –> anti-imaging filter –> playback
Codec
software application that encodes a digitized audio signal file into a different format, often for the purpose of reducing its size; also called compression
PCM
pulse code modulation; uses a modulated pulse wave to
capture amplitude information using numerical code, which in turn represent signal amplitude
values
DSD
direct stream digital; Sony’s 1 bit digital
audio coding system that uses a very high sampling frequency, used for audio representation on the Super Audio CD (SACD)
Latency
delay incurred in executing audio operations between input and output of a system; low latency is best
Redbook Audio
standard for CDs; max playtime is 79.8 minutes, min duration for a track is 4 seconds, max number of tracks is 99, max number of index points (subdivisions) in a track is 99 with no max time limit, International Standard Recording Code (ISRC) should be included
Sample rate
number of samples per second in an audio signal
Word length
size of the digital word used for the audio samples in PCM
audio (number of bits)
Clocking
high-frequency signal, usually generated by a crystal-controlled oscillator, which is used as a master timer to control the events in a digital system
Word clock
digital timing reference signal that is the same frequency
as the sampling rate being used
Over
overloading of peaks in a digital audio signal, producing clipping of the signal and, if severe, bursts of noise
BWF
digital audio file format introduced by the EBU in 1996 to facilitate professional editing and file exchange; special case of Microsoft’s wav format, but adds extra information deemed important for professional use such as titles, date, time, origination, editing data, etc.
WAV
Windows file extension that indicates Microsoft’s audio file format; uncompressed
AIFF
audio file format standard used for storing sound data for personal computers and other electronic audio devices; uncompressed PCM
Lossy
type of audio file compression that does not recover the original signal exactly
Lossless
no information is lost in the coding-decoding process
Uncompressed
audio file with all the original information of the recorded source
Headroom
difference in level between the highest level present in a given signal and the maximum level the device can handle without noticeable distortion
Nyquist
sampling rate of two times the maximum signal frequency is just sufficient to describe that highest frequency without ambiguity; ex. sample rate of 44.1kHz for 22kHz
Aliasing
if we allow frequencies higher than Nyquist frequency to enter system, violating frequencies will not be recorded, but will instead create new frequencies
Anti-aliasing
low-pass filter used during the A/D process that restricts the audio bandwidth to less than half the sampling frequency
Delta-sigma modulation
rather than capturing the actual momentary voltage value, each bit simply records whether the current amplitude of the moment is higher or lower than the previous one
Dither
pseudorandom noise signal with a level equal to half the least significant bit added to the input audio signal during A/D conversion; ensures that if any audio signal is present at the input it will broach the necessary threshold and be rounded up and successfully encoded; also randomizes the effects of quantization at low levels
Jitter
noise caused by timing inaccuracies in D/A or A/D conversion
Superclock
Pro Tools uses signal at a multiple of 256 times the sampling rate for slaving devices together with low sampling jitter
Upsampling
changing a session to a higher sample rate
Downsampling
changing a session to a lower sample rate
Elastic audio
track-based algorithm that allows you to stretch and compress audio to fit within beats and bars
Time compression-expansion
Pro Tools plugin that allows the user to make audio files longer or shorter, and to shift the pitch of audio files
Warp
allows creation of markers for elastic audio to stretch and compress audio
Floating point
some of the bits (the mantissa) represent a whole number, while other bits (the exponent) dictate how this number is multiplied or divided; able to have a very large dynamic range this way
Quantization error
difference between stairstep shape of waveform after sampling and the continuous waveform; results in wideband noise on a musical signal
PLL
phase locked loop; electronic circuit consisting of a voltage-controlled oscillator and a frequency discriminator
connected in such a way that the output of the discriminator controls the oscillator; acts as a stabilizer
PWM
pulse wave modulation; a method of encoding a signal
by using the length of a pulse as a measure of the height of a sample of the waveform; all analog
SACD
Sony and Phillips created; uses one-bit encoding of the audio signal at a sampling rate of 64 times 44.1 kHz or 2,822,400 samples per second, eliminating the use for an anti-aliasing filter at the input and anti-imaging filter at the output; eliminates phase distortion these filters produce
Claude Shannon
invented signal flow graphs
Two’s complement
way to show negative binary values; use a 1 to represent a negative number on the largest number in a byte; for example, -8 is 1000 (-8+0); -4 is 1100 (-8+4)
Flutter
in an analog tape recorder, if the tape speed varies above 5Hz or so
Wow
in an analog tape recorder, if the tape speed varies below several hertz
