Glossary
From Audacity Manual
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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.
General Terms
| Term | Description | |
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| 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. |
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| Algorithm: | A set of steps or a procedure that will produce a desired result. |
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| 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. | |
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| 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. |
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| Bit: | A measure of quantity of data. A bit is one binary digit, a 0 or a 1. |
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| Bit Rate: | The number of computer bits that are conveyed or processed per unit of time. Normally expressed in kilobits per second (kbps). |
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| CBR: | Constant Bitrate - In this format, the rate at which audio uses its data does not vary. Silence uses as much 'space' as audible sound. |
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| 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. |
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| 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. |
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| 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. |
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| 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. |
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| 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. |
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| 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. |
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| 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. |
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| 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. |
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| 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. |
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| dB: | Decibels. A logarithmic unit (typically of sound pressure) describing the ratio of that unit to a reference level. |
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| 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. |
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| FFT: | Fast Fourier Transform. A method for performing Fourier transforms quickly. |
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| 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. |
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| Filter: | A sound effect that lets some frequencies through and suppresses others. |
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| Fourier Transform: | A method for converting a waveform to a spectrum, and back. |
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| Frequency: | Audio frequency determines the pitch of a sound. Measured in Hz, higher frequencies have higher pitch. |
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| 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. |
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| 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. |
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| Headroom: | The difference between the peak level of an audio track and the 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. |
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| High Pass Filter: | A filter that lets high frequencies through |
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| Hz: | Hertz. Measures a frequency event in number of cycles per second. See Frequency and Sample Rate, both of which are measured in Hz. |
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| 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. |
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| LAME: | A software library that converts audio to MP3 format. |
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| 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. |
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| Linear: | A simple, straight-forward, directly proportional, one-to-one relationship. This term is used to contrast with logarithmic, or other complex relationships. |
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| Logarithmic: | A non-linear relationship where one item is proportional to the logarithm of the other item. Some measures, such as dB, are logarithmic by definition. |
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| 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. |
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| 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. |
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| Low Pass Filter: | A filter that lets low (bass) frequencies through. |
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| 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. |
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| 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. |
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| 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. |
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| 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. |
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| Red Book: | The most widely used standard for representing audio on CD, requiring stereo, 16-bit, 44100 Hz. |
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| RMS: | Root-mean-square. A method of calculating a numerical value for the average sound level of a waveform. |
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| 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. |
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| Sample Rate: | Measured in Hz like frequency, this represents the number of digital samples captured per second in order to represent the waveform. |
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| 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. |
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| 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. | |
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| 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. |
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| Waveform: | A visual representation of an audio signal. |
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| 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.
| Term | Description | |
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| 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. |
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| AIFF: | A container format, almost always used for lossless, uncompressed, PCM audio. The format is in Apple's Big-endian byte order. |
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| AU: | A container format, used by Audacity for storage of lossless, uncompressed, PCM audio data |
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| FLAC: | An Open Source lossless, size-compressed audio format |
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| 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. |
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| MP2: | A lossy, size-compressed audio format mainly used by the broadcast media |
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| MP3: | A lossy, size-compressed audio format which is the main format for transmitting audio over the internet |
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| Ogg Vorbis: | An Open Source lossy, size-compressed audio format |
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| WAV: | A container format, almost always used for lossless, uncompressed, PCM audio. The format is in Microsoft's Little-Endian byte order. |
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| 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. |
