Glossary

From Audacity Manual

This page gives very brief explanations of technical terms related to digital audio, with some links to Wikipedia for much more comprehensive explanations.

General Terms

	Term	Description
	ADC:	Analog to digital converter. The part of a sound card which records an analog, real world sound like a voice or guitar and converts it to a numerical representation of the audio that a computer can manipulate.
	Algorithm:
	Audacity Project Format (.aup):	The format in which Audacity stores its projects. This consists of a reference file with the extension .aup and a large number of small audio files with extension .AU. This structure makes it quicker for Audacity to move audio around - ideal for cutting and pasting audio in a project.
	Audio CDs:
	Amplitude:
	Bit	A measure of quantity of data. A bit is one binary digit, a 0 or a 1.
	Bit Rate:	The number of computer bits that are conveyed or processed per unit of time. Normally expressed in kilobits per second (kbps).
	CBR:	Constant Bitrate - The rate at which audio in this format uses its data does not vary. Silence uses as much 'space' as audible sound.
	Cepstrum:	Some description here. It's like a logarithm of a Fourier transform. We need a better description [1]
	Clipping:	Distortion to sound that happens when the audio is too loud. When a waveform shows 'flat tops' rather than smooth curves it is usually an indication of clipping.
	Compressed Audio Format:
	Compression:
	Cycle:
	DAC:	Digital to analog converter. The part of a sound card which plays back a numerical representation of audio as an analog, real world sound like a voice or guitar.
	Data CDs:
	dB:	Decibels. A logarithmic unit (typically of sound pressure) describing the ratio of that unit to a reference level.
	Dynamic Range:	The difference between the loudest and softest part in an audio recording, the maximum possible being determined by its sample format. For a device, the difference between its maximum possible undistorted signal and its Noise Floor.
	FFT:	Fast Fourier Transform. A method for performing Fourier transforms quickly.
	Filter:	A sound effect that lets some frequencies through and suppresses others.
	Fourier Transform:	A method for converting a waveform to a spectrum, and back.
	Frequency:	Audio frequency determines the pitch of a sound. Measured in Hz (see below), higher frequencies have higher pitch.
	Gain:	How much to amplify the sound by.
	Interpolation:	Completing waveform data by estimating missing values. The values are estimated as being between other known values. To convert a waveform recorded at 22000 Hz or samples per second to one at a higher rate such as 44000 samples per second requires interpolation.
	Harmonics:	Most sounds are made up of a mix of different frequencies. In musical sounds, the component frequencies are simple multiples of each other, for example 100 Hz, 200 Hz, 300 Hz. These are called harmonics of the lowest frequency sound.
	High Pass Filter:	A filter that lets high frequencies through
	Hz:	Hertz. Measures a frequency event in number of cycles per second. See Frequency and Sample Rate, both of which are measured in Hz.
	LAME:	A software library that converts audio to MP3 format.
	Linear:
	Logarithmic:
	Lossless:	A format for size-compressing audio that does not lose any information. The quality is exactly as good as before compression. An example is FLAC.
	Lossy:	A format for size-compressing audio that may sacrifice a small amount of quality in order to reduce the file size more than lossless compression. Examples are MP3 and OGG.
	Low Pass Filter:	A filter that lets low (bass) frequencies through.
	MP3 CDs:
	Noise Floor:
	PCM:	Pulse code modulation. A method of converting audio into binary numbers to represent it digitally, then back to audio. The waveform is measured at evenly spaced intervals and the amplitude of the waveform noted for each measurement.
	Pitch:
	Red Book:	The most widely used standard for representing audio on CD, requiring stereo, 16-bit, 44100 Hz.
	RMS:	Root-mean-square. A method of calculating a numerical value for the average sound level of a waveform.
	Sample Rate:	Measured in Hz like frequency, this represents the number of digital samples captured per second in order to represent the waveform.
	Sample Format:	Also known as Bit Depth or Word Size. The number of computer bits present in each audio sample. Determines the dynamic range of the audio.
	Spectrum:	Presentation of a sound in terms of its component frequencies.
	Uncompressed Audio Format:	An audio format in which every sample of sound is represented by a binary number. Examples are WAV or AIFF.
	VBR:	Variable bit rate. A method for compressing audio which does not always use the same number of bits to record the same duration of sound.
	Waveform:
	Wavelength:

Sound File Formats

	Term	Description
	AIFF	A container format, almost always used for lossless, uncompressed, PCM audio. The format is in Apple's Big-endian byte order.
	AU	A container format, used by Audacity for storage of lossless, uncompressed, PCM audio data
	FLAC:	An Open Source lossless, size-compressed audio format
	MIDI:	MIDI is a small-sized file format which stores how to play notes, widely used for keyboard instruments. It is not an audio file format like WAV that uses thousands of samples to record the full sound of the notes actually being played.
	MP2:	A lossy, size-compressed audio format mainly used by the broadcast media
	MP3:	A lossy, size-compressed audio format which is the main format for transmitting audio over the internet
	Ogg Vorbis:	An Open Source lossy, size-compressed audio format
	WAV:	A container format, almost always used for lossless, uncompressed, PCM audio. The format is in Microsoft's Little-Endian byte order.