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

  • ToDo Gale: Add a glossary definition for rolloff, and link to it in these pages
  • ToDo In Opera, columns are drawn, not what we wanted... - Gale

JC: Actually I don't think that looks bad. Maybe not what we intended but not an offense to the eye.

General Terms

Term Description
Wikipedia1.png 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.
Wikipedia1.png Algorithm: A set of steps or a procedure that will produce a desired result.
Wikipedia1.png ALSA: A Linux kernel component for providing device drivers for sound cards. Known as an audio host in Audacity.
Wikipedia1.png Amplitude: The level or magnitude of a signal. Audio signals with a higher amplitude will sound louder.
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.
Wikipedia1.png Audio CDs: CDs containing PCM audio data in accordance with the Red Book standard. They can be played on any standalone CD player as well as on computers.
Wikipedia1.png Batch Processing: The execution of a series of programs ("jobs") on a computer, it can save time and energy by automating repetitive tasks. Jobs are set up so they can be run to completion without manual intervention, so all input data are preselected through scripts, command-line parameters, or job control language. The term batch job originated in the days when punched cards contained the directions for a computer to follow when running one or more programs; multiple card decks representing multiple jobs would often be stacked in the hopper of a card reader and be run in batches. Batch processing in Audacity is implemented by using Chains.
Wikipedia1.png Bit: A measure of quantity of data. A bit is one binary digit, a 0 or a 1.
Wikipedia1.png Bit Rate: The number of computer bits conveyed or processed per unit of time. Normally expressed in kilobits per second (kbps). For an uncompressed, PCM file, kbps bit rate is sample rate multiplied by sample format mutiplied by number of channels, divided by 1000, giving 1411 kbps for Red Book WAV or AIFF. Rates are much lower for compressed or lossy formats and are independent of sample rate for MP3.
Wikipedia1.png CBR: Constant Bit Rate - In this format, the rate at which audio uses its data does not vary. Silence uses as much 'space' as audible sound.
Wikipedia1.png Cepstrum: The cepstrum of an audio signal is related to the spectrum, but presents the rate of change in the different spectrum bands. It's particularly useful for properties of vocal tracks and is used, for example, in software to identify speakers by their voice characteristics.
Wikipedia1.png Clipping: Distortion to sound, usually due to the audio being too loud. Unless the original audio is 32-bit sample format, waveforms louder than 0 dB will have their tops lopped off (flattened) at 0 dB, rather than showing smooth curves. Clipping can also be an intentional distortion effect that lops off part of the waveform, reducing its amplitude and changing its frequency content.
Wikipedia1.png Codec: A computer program capable of encoding and/or decoding a digital data stream. The term is a portmanteau (a blending of two or more words) of coder and decoder.
Wikipedia1.png Companding: Refers to the process of compressing the dynamic range of an audio signal before storage or transmission, then expanding the signal on retrieval or reception. The term is a portmanteau (a blending of two or more words) of compressing and expanding.
Wikipedia1.png Compressed Audio Format: Any format that will reduce the space required in storing or representing an audio signal. Space savings can be made for example by discarding certain frequency components which may be inaudible. MP3 takes this approach. Other formats such as FLAC compress without audio loss, but achieve lower compression rates.
Wikipedia1.png Compression: A process that tends to even out the overall volume level by increasing the level of softer passages and decreasing the level of louder passages. See also Compressed Audio Format.
Wikipedia1.png Cycle: An audio tone consists of an oscillating sound pressure on the ear. One cycle is one full transition of positive pressure through to negative pressure, back to positive pressure again.
Wikipedia1.png 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.
Wikipedia1.png Data CDs: Data CDs contain data intended to be read directly by a computer. The data may include audio and any other types of file such as images and documents. Most standalone CD players will not play data CDs, but some DVD players will. Including compressed audio files on a data CD can greatly increase the playing time compared to audio CDs.
Wikipedia1.png dB: Decibels. A logarithmic unit (typically of sound pressure) describing the ratio of that unit to a reference level.
Wikipedia1.png DC Offset: An offsetting of a signal from zero. A signal with DC Offset would appear in the Audacity Default Waveform view to be not centered on the 0.0 horizontal line. DC Offset results in reduced headroom and can cause clicks at the start and end or distortion after running effects. It can be corrected in Audacity by running Normalize.
Wikipedia1.png 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.
Wikipedia1.png Exponential: A non-linear relationship where a change in value is proportional to the current level. If you double the value in a time period, it doubles again in the next period; if you halve the level in a time period, it halves again in the next period. For an exponential fade in, the curve becomes "steeper" with time; an exponential fade out becomes "flatter" with time. See also Logarithmic.
Wikipedia1.png FFT: Fast Fourier Transform. A method for performing Fourier transforms quickly.
Wikipedia1.png File name extension: A suffix of three or four characters added to a file name which defines the format of its contents. The suffix is separated from the file name by a dot (period), as in "song.mp3". The extension of common formats is often hidden on Windows, but can be turned on in the system's Folder Options.
Wikipedia1.png Filter: A sound effect that lets some frequencies through and suppresses others.
Wikipedia1.png Fourier Transform: A method for converting a waveform to a spectrum, and back.
Wikipedia1.png Frequency: Audio frequency determines the pitch of a sound. Measured in Hz, higher frequencies have higher pitch. See this Wikipedia article.
Wikipedia1.png Gain: A measure of how much a signal is amplified. Usually expressed in dB, positive gain increases the amplitude of a signal, while negative gain reduces it.
Wikipedia1.png 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.
Wikipedia1.png Headroom: The difference between the peak level of an audio track and the maximum level that can be achieved without clipping. Recording at -6 dB below maximum level is a good compromise between getting far enough above the noise floor while having sufficient headroom to make edits that increase loudness.
Wikipedia1.png High Pass Filter: A filter that lets high frequencies through
Wikipedia1.png Hz: Hertz. Measures a frequency event in number of cycles per second. See Frequency and Sample Rate, both of which are measured in Hz.
Wikipedia1.png 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.
Wikipedia1.png kHz: One kilohertz (kHz) is 1000 Hz. For example, the common audio sample rate of 44100 Hz can also be expressed as 44.1 kHz.
Wikipedia1.png LAME: A software library that converts audio to MP3 format.
Wikipedia1.png Latency: A short delay between an audio signal being sent and received. In computer audio this is due to analog-to-digital and digital-to-analog conversion. Most commonly refers to the delay between recording a sound and a) hearing its playthrough or b) laying it down on disk.
Wikipedia1.png Linear: A simple, directly proportional, one-to-one, "straight-line" relationship. This term is used to contrast with exponential, logarithmic, or other complex relationships.
Wikipedia1.png Logarithmic: A non-linear relationship where one item is proportional to the logarithm of the other item. So for a logarithmic fade in, the curve becomes "flatter" with time; a logarithmic fade-out becomes "steeper" with time. Some measures, such as dB, are logarithmic by definition. See also Exponential.
Wikipedia1.png Lossless: A format that does not lose any information. It may be either a size-compressing format like FLAC where the quality is exactly as good as before compression, or an uncompressed format like WAV.
Wikipedia1.png 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.
Wikipedia1.png Low Pass Filter: A filter that lets low (bass) frequencies through.
Wikipedia1.png MME: Multimedia Extensions to Windows 3 appeared in Autumn 1991 as the first standardized Windows interface to support sound cards. It is one of the "audio hosts" selectable in Device Toolbar. MME was superseded in 1995 by Windows DirectSound.
Wikipedia1.png MP3 CDs: A specific type of data CD containing only MP3 audio files. All computers can play them as can some DVD and portable MP3 players.
Wikipedia1.png Noise Floor: A level or amplitude representing the amount of near-continuous background noise present in the signal. A background hiss would raise the noise floor, and could prevent a faint signal (one below the noise floor) being heard at all. Unwanted sporadic noise such as a member of the audience coughing is noise, but it doesn't contribute to the noise floor.
Wikipedia1.png Pan: Panning is the spread of a sound signal (either monaural or stereophonic pairs) into a new stereo or multi-channel sound field.
Wikipedia1.png 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.
Wikipedia1.png Pitch: Generally synonymous with the fundamental frequency of a note, but in music, often also taken to imply a perceived measurement that can be affected by overtones above the fundamental.
Wikipedia1.png Red Book: The most widely used standard for representing audio on CD, requiring stereo, 16-bit, 44100 Hz.
Wikipedia1.png Resampling: Converting a sampled signal from one sample rate to another without changing the length of the audio (hence without changing the playback speed or pitch). This necessarily changes the number of samples that the audio contains. Resampling can also mean converting from one sample format to another which changes the precision of each sample but not the number of samples.
Wikipedia1.png RMS: Root-mean-square. A method of calculating a numerical value for the average sound level of a waveform. The RMS level (colored lighter blue in Audacity) equates very approximately to how loud the audio sounds.
Wikipedia1.png Sample: A discrete value at a point in a waveform representing the audio at that point. Also the act of taking a sequence of such values. All digital audio must be sampled at discrete points. By contrast, analog audio (such as the sound from a loudspeaker) is always a continuous signal.
Wikipedia1.png Sample Rate: Measured in Hz like frequency, this represents the number of digital samples captured per second in order to represent the waveform.
Wikipedia1.png 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.
Wikipedia1.png 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.
Wikipedia1.png 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.
Wikipedia1.png Waveform: A visual representation of an audio signal.
Wikipedia1.png Windows DirectSound: A Windows interface between applications (such as Audacity) and the sound card driver. It is one of the "audio hosts" selectable in Device Toolbar. DirectSound was released in 1995 as a replacement for the older MME.
Wikipedia1.png Zero Crossing: The point where a line joining the audio samples crosses the zero horizontal line.

Audio File Formats

There are numerous audio file formats for storing audio on a computer.

  • WAV format is widely used on Windows and is needed for creating an audio CD.
  • AIFF is widely used on Apple's operating systems.
  • Compressed formats are used on portable music players.
  • Description of FFmpeg formats (AC3, AMR NB)
  • As we have now decided not to have separate pages for the different formats (only separate pages for format export options), we need to rethink where these Glossary entries link to - best left until we decide if we have an appendix containing details of "audio file formats" - Gale
  • Ed 13 Nov 2009 consider adding MP4 (aka M4A) formats which are also supported (see [1] the wikipedia article)
Term Description
Wikipedia1.png AAC: A lossy, size-compressed audio format. AAC files usually have M4A extension, with variants such as M4P (protected) and M4R (ringtones). Usually gives better quality for the same bit rate than the older MP3 format. Is default audio format for iTunes®, iPod® and iPhone®, and Sony PlayStation 3.
Wikipedia1.png AIFF: A container format, almost always used for lossless, uncompressed, PCM audio. The format is in Apple's Big-endian byte order.
Wikipedia1.png AU: A container format, used by Audacity for storage of lossless, uncompressed, PCM audio data. Not be confused with Sun/NeXT AU files, which are usually U-Law encoded PCM files but may be headerless.
Wikipedia1.png FLAC: An Open Source lossless, size-compressed audio format
Wikipedia1.png 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.
Wikipedia1.png MP2: A lossy, size-compressed audio format mainly used by the broadcast media
Wikipedia1.png MP3: A lossy, size-compressed audio format which is the main format for transmitting audio over the internet
Wikipedia1.png Ogg Vorbis: An Open Source lossy, size-compressed audio format
Wikipedia1.png WAV: A container format, almost always used for lossless, uncompressed, PCM audio. The format is in Microsoft's Little-Endian byte order.
Wikipedia1.png WMA: A container format. Windows Media Audio is a lossy, size-compressed audio format developed by Microsoft. It is a proprietary technology that forms part of the Windows Media framework. WMA consists of four distinct codecs. The original WMA codec, known simply as WMA, was conceived as a competitor to the popular MP3 and RealAudio codecs.
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