Course:LFS400/Glossary
This page, information courtesy of the Audacity Reference Manual, 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 an audio interface which records an analog, real world sound like a voice or guitar and converts it to a <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Binary_numeral_system">numerical</a> representation of the audio that a computer can manipulate. |
| Algorithm: | A set of steps or a procedure that will produce a desired result. |
| Aliasing: | Aliasing is an effect that causes different audio signals to become indistinguishable (or aliases of one another) when sampled. It also refers to the distortion or artifact that results when the signal reconstructed from samples is different from the original continuous signal. |
| ALSA: | A Linux kernel component for providing device drivers for audio interfaces. Known as an audio host in Audacity. |
| Amplitude: | The level or magnitude of a signal. Audio signals with a higher amplitude will sound louder. |
| Artifact: | Sonic material that is accidental or unwanted, resulting from the editing of another sound. |
| <a href="asio_audio_interface.html" title="ASIO Audio Interface">ASIO</a>: | Audio Stream Input/Output (<a href="asio_audio_interface.html" title="ASIO Audio Interface">ASIO</a>) is a computer audio interface driver protocol for digital audio on Windows, created by Steinberg. It provides a low-latency, multi-channel interface between a software application and the audio interface. |
| <a href="audacity_projects.html" title="Audacity Projects">Audacity Project Format (.aup)</a>: | The format in which Audacity formerly stored its projects. This consists of a reference file with the extension .aup and a large number of small audio files with extension <a href="#au">.AU</a>. This structure makes it quicker for Audacity to move audio around - ideal for cutting and pasting audio in a project. The project format for Audacity 2.4.2 and earlier. |
| <a href="audacity_projects.html" title="Audacity Projects">Audacity Unitary Project format (.aup3)</a>: | The format in which Audacity previously stored its projects. This consists of a SQLite database file with the extension .aup3. The project format for Audacity 3.0.0 and later. |
| Audio CDs: | CDs containing <a href="#pcm">PCM</a> audio data in accordance with the <a href="#red_book">Red Book</a> standard. They can be played on any standalone CD player as well as on computers. |
| Band-pass filter: | A band-pass filter is a <a href="#filter">filter</a> that passes <a href="#frequency">frequencies</a> within a certain range and rejects (attenuates) frequencies outside that range. A band-boost filter is similar to a band-pass filter except that it amplifies frequencies within a certain range and passes frequencies outside that range untouched. |
| Band-stop filter: | A band-stop filter or band-rejection filter is a <a href="#filter">filter</a> that passes most <a href="#frequency">frequencies</a> unaltered, but attenuates those in a specific range to very low levels. It is the opposite of a band-pass filter. A band-cut filter is a band-stop filter that attenuates the frequencies in a given frequency band by a specified amount. A notch filter is a band-stop filter with a narrow stopband (high Q factor). |
| Batch Processing: | Automation of a series of repetitive tasks on a computer so that the tasks run without manual intervention. In the early days of computers this was done by processing stacks of punch cards. In Audacity, repetitive tasks are handled by creating a <a href="macros.html" title="Macros">Macro</a>. The Macro can apply a predetermined sequence of effects to the current project, or can be run unattended to apply effects and/or format conversions to a batch of external audio files. |
| Bit: | A measure of quantity of data. A bit is one binary digit, a 0 or a 1. |
| Bit Rate: | The number of computer <a href="#bit">bits</a> conveyed or processed per unit of time. Normally expressed in kilobits per second (kbps). For an <a href="#uncompressed_format">uncompressed</a>, <a href="#pcm">PCM</a> file, kbps bit rate is <a href="#sample_rate">sample rate</a> multiplied by <a href="#sample_format">sample format</a> multiplied by number of channels, divided by 1000, giving 1411 kbps for <a href="#red_book">Red Book</a> <a href="#wav">WAV</a> or <a href="#aiff">AIFF</a>. Rates are much lower for <a href="#compressed_format">compressed</a> or <a href="#lossy">lossy</a> formats like <a href="#mp3">MP3</a>. For MP3 at <a href="#cbr">constant bit rate</a>, reducing sample rate does not reduce the bit rate and hence does not make the MP3 smaller, except for 11,025 Hz and below. |
| CBR: | Constant Bit Rate - In this format, the rate at which audio uses its data <a href="#bit">bits</a> does not vary. Silence uses as much disk space as audible sound. |
| Cepstrum: | The cepstrum of an audio signal is related to the <a href="#spectrum">spectrum</a>, 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. |
| Clipping: | Distortion to sound, usually due to the audio being too loud. Unless the original audio is 32-bit <a href="#sample_format">sample format</a>, <a href="#waveform">waveforms</a> louder than 0 <a href="#decibel">dB</a> 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 <a href="#amplitude">amplitude</a> and changing its <a href="#frequency">frequency</a> content. |
| 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. |
| Companding: | Refers to the process of compressing the <a href="#dynamic_range">dynamic range</a> 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. |
| 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 <a href="#frequency">frequency</a> components which may be inaudible. <a href="#mp3">MP3</a> takes this approach. Other formats such as <a href="#flac">FLAC</a> compress without audio loss, but achieve lower compression rates. |
| <a href="compressor.html" title="Compressor">Compression</a>: | 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 <a href="#compressed_format">Compressed Audio Format</a>. |
| 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. |
| DAC: | Digital to analog converter. The part of an audio interface which plays back a <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Binary_numeral_system">numerical</a> representation of audio as an analog, real world sound like a voice or guitar. |
| 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 <a href="#compressed_format">compressed audio files</a> on a data CD can greatly increase the playing time compared to <a href="#audio_cd">audio CDs</a>. |
| dB: | Decibels. A <a href="#log">logarithmic</a> unit (typically of sound pressure) describing the ratio of that unit to a reference level. |
| <a href="dc_offset.html" title="DC offset">DC Offset</a>: | An offsetting of a signal from zero. A signal with DC Offset would appear in the Audacity <a href="audacity_waveform.html" title="Audacity Waveform">Default Waveform view</a> to be not centered on the 0.0 horizontal line. DC Offset results in reduced <a href="#headroom">headroom</a> and can cause clicks at the start and end or distortion after running effects. It can be corrected in Audacity by running <a href="normalize.html" title="Normalize">Normalize</a>. |
| <a href="dither.html" title="Dither">Dither</a>: | Intentional noise which is added so as to randomize the <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Quantization_error">quantization errors</a> (rounding errors) that occur when downsampling the <a href="sample_format_bit_depth.html" title="Sample Format - Bit Depth">Bit Depth</a> of an audio stream to a lower resolution than the current format. |
| Dynamic Range: | The difference between the loudest and softest part in an audio recording, the maximum possible being determined by its <a href="#sample_format">sample format</a>. For a device, the difference between its maximum possible undistorted signal and its <a href="#noise_floor">Noise Floor</a>. |
| <a href="recording.html#dropouts" title="Recording">Dropout</a>: | A dropout is a momentary loss of signal in your recording. Dropouts may be caused by a disk drive that cannot keep up with the recording. This can happen, for example, with a slow USB or network drive, or if antivirus software is slowing writing to disk, or if other activity on the computer is slowing the computer down. |
| 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 <a href="fades.html" title="Fades">fade in</a>, the curve becomes "steeper" with time; an exponential fade out becomes "flatter" with time. See also <a href="#log">Logarithmic</a>. |
| FFT : | Fast Fourier Transform. A method for performing <a href="#ft">Fourier transforms</a> quickly. |
| File name extension: | A suffix of three or four characters added to a file name which defines the <a href="#Audio_File_Formats">format</a> 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. |
| Filter: | A sound effect that lets some <a href="#frequency">frequencies</a> through and suppresses others. |
| Fourier Transform: | A method for converting a <a href="#waveform">waveform</a> to a <a href="#spectrum">spectrum</a>, and back. |
| Frequency: | Audio frequency determines the <a href="#pitch">pitch</a> of a sound. Measured in <a href="#hz">Hz</a>, higher frequencies have higher pitch. See <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Audio_frequency">this</a> Wikipedia article. |
| Gain: | A measure of how much a signal is amplified. Usually expressed in <a href="#decibel">dB</a>, positive gain increases the <a href="#amplitude">amplitude</a> of a signal, while negative gain reduces it. |
| Harmonics: | Most sounds are made up of a mix of different <a href="#frequency">frequencies</a>. In musical sounds, the component frequencies are simple multiples of each other, for example 100 <a href="#hz">Hz</a>, 200 Hz, 300 Hz. These are called harmonics of the lowest frequency sound. |
| Headroom: | The difference between the peak level of an audio track and the maximum level that can be achieved without <a href="#clipping">clipping</a>. Recording at -6 <a href="#decibel">dB</a> below maximum level is a good compromise between getting far enough above the <a href="#noise_floor">noise floor</a> while having sufficient headroom to make edits that increase loudness. |
| High Pass Filter: | A <a href="#filter">filter</a> that lets high <a href="#frequency">frequencies</a> through |
| Hz: | Hertz. Measures a <a href="#frequency">frequency</a> event in number of <a href="#cycle">cycles</a> per second. See Frequency and <a href="#sample_rate">Sample Rate</a>, both of which are measured in Hz. |
| ID3: | ID3 is a <a href="#metadata">metadata</a> container most often used in conjunction with the <a href="#mp3">MP3</a> audio file format. It allows information such as the title, artist, album, track number, and other information about the file to be stored in the file itself. |
| Interpolation: | Completing <a href="#waveform">waveform</a> data by estimating missing values. The values are estimated as being between other known values. To convert a waveform recorded at 22,000 <a href="#hz">Hz</a> or <a href="#sample_rate">samples per second</a> to one at a higher rate such as 44000 samples per second requires interpolation. |
| IVR: | Interactive Voice Response is a technology that allows a computer to interact with humans through the use of voice and DTMF tones input via keypad. |
| kHz: | One kilohertz (kHz) is 1000 <a href="#hz">Hz</a>. For example, the common audio <a href="#sample_rate">sample rate</a> of 44,100 Hz can also be expressed as 44.1 kHz. |
| <a href="faq_installation_and_plug_ins.html#lame" title="FAQ:Installation and Plug-Ins">LAME</a>: | A software library that converts audio to <a href="#mp3">MP3</a> format. |
| Latency: | A short delay between an audio signal being sent and received. In computer audio this is due to <a href="#adc">analog-to-digital</a> and <a href="#dac">digital-to-analog</a> conversion. Most commonly refers to the delay between recording a sound and a) hearing its <a href="recording_preferences.html#Playthrough" title="Recording Preferences">playthrough</a> or b) laying it down on disk. |
| Linear: | A simple, directly proportional, one-to-one, "straight-line" relationship. This term is used to contrast with <a href="#exponential">exponential</a>, <a href="#log">logarithmic</a>, or other complex relationships. |
| Logarithmic: | A non-linear relationship where one item is proportional to the logarithm of the other item. So for a logarithmic <a href="fades.html" title="Fades">fade in</a>, the curve becomes "flatter" with time; a logarithmic fade-out becomes "steeper" with time. Some measures, such as <a href="#decibel">dB</a>, are logarithmic by definition. See also <a href="#exponential">Exponential</a>. |
| Lossless: | A format that does not lose any information. It may be either a <a href="#compressed_format">size-compressing</a> format like <a href="#flac">FLAC</a> where the quality is exactly as good as before compression, or an <a href="#uncompressed_format">uncompressed</a> format like <a href="#wav">WAV</a>. |
| Lossy: | A format for <a href="#compressed_format">size-compressing</a> audio that may sacrifice a small amount of quality in order to reduce the file size more than <a href="#lossless">lossless</a> compression. Examples are <a href="#mp3">MP3</a> and <a href="#ogg">OGG</a>. |
| Low Pass Filter: | A <a href="#filter">filter</a> that lets low (bass) <a href="#frequency">frequencies</a> through. |
| Metadata: | Metadata tags - digital audio files can be labeled with more information than can be contained in just the file name, that descriptive information is called the audio tag or audio metadata. The metadata for compressed and uncompressed digital music is often encoded in the <a href="#id3">ID3</a> tag. |
| MME: | Multimedia Extensions to Windows 3 appeared in Autumn 1991 as the first standardized Windows interface to support audio interfaces. It is one of the "audio hosts" selectable in <a href="device_toolbar.html" title="Device Toolbar">Device Toolbar</a>. MME was superseded in 1995 by <a href="#wds">Windows DirectSound</a>. |
| MP3 CDs: | A specific type of <a href="#data_cd">data CD</a> containing only <a href="#mp3">MP3</a> audio files. All computers can play them as can some DVD and portable MP3 players. |
| Noise Floor: | A level or <a href="#amplitude">amplitude</a> 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 does not contribute to the noise floor. |
| Notch filter: | A notch filter is a <a href="#band_stop">band-stop filter</a> with a narrow stopband (high Q factor). |
| Oversampling: | Oversampling is the process of sampling a signal with a sampling frequency significantly higher than the Nyquist rate. Oversampling improves resolution, reduces noise and helps avoid <a href="#aliasing">aliasing</a> and phase distortion by relaxing anti-aliasing filter performance requirements. |
| Pan: | Panning is the spread of a sound signal (either monaural or stereophonic pairs) into a new stereo or multi-channel sound field. |
| PCM: | Pulse code modulation. A method of converting audio into <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Binary_numeral_system">binary numbers</a> to represent it digitally, then back to audio. The <a href="#waveform">waveform</a> is measured at evenly spaced intervals and the <a href="#amplitude">amplitude</a> of the waveform noted for each measurement. |
| Pitch: | Generally synonymous with the fundamental <a href="#frequency">frequency</a> of a note, but in music, often also taken to imply a perceived measurement that can be affected by overtones above the fundamental. |
| Red Book: | The most widely used standard for representing audio on <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/CD">CD</a>, requiring stereo, 16-<a href="#bit">bit</a>, 44,100 <a href="#hz">Hz</a>. |
| Resampling: | Converting a sampled signal from one <a href="#sample_rate">sample rate</a> to another without changing the length of the audio (hence without changing the playback speed or <a href="#pitch">pitch</a>). This necessarily changes the number of <a href="#sample">samples</a> that the audio contains. Resampling can also mean converting from one <a href="#sample_format">sample format</a> to another which changes the precision of each sample but not the number of samples. |
| RMS: | Root Mean Square, sometimes also abbreviated in technical literature as "rms". A method of calculating a numerical value for the average sound level of a <a href="#waveform">waveform</a>. The RMS level (colored lighter blue in Audacity) equates very approximately to how loud the audio sounds. |
| Roll-off: | A gradually reduced response at the upper or lower ends of the working frequency range. |
| Sample: | A discrete value at a point in a <a href="#waveform">waveform</a> representing the audio at that point. Also the act of taking a sequence of such values. All <a href="digital_audio.html" title="Digital Audio">digital audio</a> must be sampled at discrete points. By contrast, analog audio (such as the sound from a loudspeaker) is always a continuous signal. |
| <a href="digital_audio.html#Sample_rates" title="Digital Audio">Sample Rate</a>: | Measured in <a href="#hz">Hz</a> like <a href="#frequency">frequency</a>, this represents the number of digital <a href="#sample">samples</a> captured per second in order to represent the <a href="#waveform">waveform</a>. See <a href="sample_rates.html" title="Sample Rates">Sample Rates</a> for more details. |
| <a href="digital_audio.html#Sample_formats" title="Digital Audio">Sample Format</a>: | Also known as Bit Depth or Word Size. The number of computer <a href="#bit">bits</a> present in each audio <a href="#sample">sample</a>. Determines the <a href="#dynamic_range">dynamic range</a> of the audio. See <a href="sample_format_bit_depth.html" title="Sample Format - Bit Depth">Sample Format - Bit Depth</a> for more details. |
| Snapshot: | A read-only copy of the project database frozen at a point in time enabling <a href="recovery.html" title="Recovery">Recovery</a>, following a crash, to recover to the last edit you made. |
| Spectrum: | Presentation of a sound in terms of its component <a href="#frequency">frequencies</a>. |
| Uncompressed Audio Format: | An audio format in which every <a href="#sample">sample</a> of sound is represented by a <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Binary_numeral_system">binary number</a>. Examples are <a href="#wav">WAV</a> or <a href="#aiff">AIFF</a>. |
| VBR: | Variable Bit Rate. A method for compressing audio which does not always use the same number of <a href="#bit">bits</a> to record the same duration of sound. |
| <a href="audacity_waveform.html" title="Audacity Waveform">Waveform</a>: | A visual representation of an audio signal. |
| Windows DirectSound: | A Windows interface between applications (such as Audacity) and the audio interface driver. It is one of the "audio hosts" selectable in <a href="device_toolbar.html" title="Device Toolbar">Device Toolbar</a>. DirectSound was released in 1995 as a replacement for the older <a href="#mme">MME</a> and has an option to bypass the kernel mixer and so reduce <a href="#latency" title="Glossary">latency</a>. |
| Windows WASAPI: | The most recent Windows interface between applications (such as Audacity) and the audio interface driver. It is one of the "audio hosts" selectable in <a href="device_toolbar.html" title="Device Toolbar">Device Toolbar</a>. WASAPI was first officially released in 2007. |
| <a href="select_menu_at_zero_crossings.html" title="Select Menu: At Zero Crossings">Zero Crossing</a>: | The point where a line joining the audio <a href="#sample">samples</a> crosses the zero horizontal line. |
Audio File Formats
There are numerous <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Audio_file_formats">audio file formats</a> 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 (like MP3 and AAC) are used on portable music players.
| Term | Description |
|---|---|
| AAC: | A <a href="#lossy">lossy</a>, <a href="#compressed_format">size-compressed</a> audio codec and its reference audio codec implementation. AAC files usually have M4A <a href="#extension">extension</a>, with variants such as M4P (protected) and M4R (ringtones). Usually gives better quality for the same <a href="#bit_rate">bit rate</a> than the older <a href="#mp3">MP3</a> format. It is the default audio format for Apple Music/iTunes\xc2\xae, iPhone\xc2\xae, iPad\xc2\xae and iPod\xc2\xae and Sony PlayStation 3. |
| AIFF: | A <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Container_format_(digital)">container</a> format, almost always used for <a href="#lossless">lossless</a>, <a href="#uncompressed_format">uncompressed</a>, <a href="#pcm">PCM</a> audio with similar file size to <a href="#wav">WAV</a>. Although the classic AIFF format is in Apple's earlier <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Endianness#Big-endian">Big-endian</a> byte order, Mac OS X /macOS has always written "AIFF-C/sowt" files. These have the same AIFF <a href="#extension">extension</a> as classic AIFF and are identical to it except for being <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Endianness#Little-endian">Little-endian</a> like WAV format. Rarely, files with AIFC extension can contain <a href="#compressed_format">compressed</a> formats. |
| Allegro: | A text-based language for music representation. In common with <a href="#midi">MIDI</a> it represents notes, tempo, and other commands that may instruct a synthesizer or sampler what to play. In Audacity, Allegro (.gro) files may be imported as <a href="note_tracks.html" title="Note Tracks">Note tracks</a> or exported from Note tracks |
| Apple Lossless: | Also known as Apple Lossless Audio Codec (ALAC) or Apple Lossless Encoder (ALE), this is a <a href="#lossless">lossless</a>, <a href="#compressed_format">size-compressed</a> codec usually stored within an MP4 <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Digital_container_format">container format</a> with M4A <a href="#extension">extension</a>. ALAC is Apple's equivalent of <a href="#flac">FLAC</a> (which is not officially supported by Apple). |
| AU: | A <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Container_format_(digital)">container</a> format, formerly used by Audacity (2.4.2 and earlier) for storage of <a href="#lossless">lossless</a>, <a href="#uncompressed_format">uncompressed</a>, <a href="#pcm">PCM</a> audio data. Not be confused with <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Au_file_format">Sun/NeXT AU</a> files, which are usually U-Law encoded PCM files but may be headerless. |
| CAF: | A <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Container_format_(digital)">container</a> format for storing audio, developed by Apple Inc. It is designed to overcome limitations of older digital audio formats. Unlike WAV and AIFF its size is virtually unlimited and can theoretically save hundreds of years of recorded audio due to its use of 64-bit file offsets. |
| FLAC: | An Open Source <a href="#lossless">lossless</a>, <a href="#compressed_format">size-compressed</a> audio format |
| GSM 6.10: | Global System for Mobile communications is a standard developed by the European Telecommunications Standards Institute (ETSI) to describe protocols for second-generation (2G) digital cellular networks used by mobile phones. As of 2014 it has become the default global standard for mobile communications - with over 90% market share, operating in over 219 countries and territories. |
| 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 <a href="#wav">WAV</a> that uses thousands of <a href="#sample">samples</a> to record the full sound of the notes actually being played. |
| MP2: | A <a href="#lossy">lossy</a>, <a href="#compressed_format">size-compressed</a> audio format mainly used by the broadcast media |
| MP3: | A <a href="#lossy">lossy</a>, <a href="#compressed_format">size-compressed</a> audio format which is the main format for transmitting audio over the Internet |
| Opus: | An Open Source <a href="#compressed_format" title="Glossary">size-compressed</a> and <a href="#lossy" title="Glossary">lossy</a> audio format developed for Internet streaming. It uses both SILK (used by Skype) and CELT (from Xiph.Org) codecs and supports variable bit rates from 6 kbps to 510 kbps. |
| Ogg Vorbis: | An Open Source <a href="#lossy">lossy</a>, <a href="#compressed_format">size-compressed</a> audio format, strictly speaking the <a rel="nofollow" class="external text" href="https://en.wikipedia.org/wiki/Vorbis">Vorbis</a> format in a <a rel="nofollow" class="external text" href="https://en.wikipedia.org/wiki/Container_format_%28digital%29">container</a> having OGG <a href="#extension">extension</a>. |
| RAW: | RAW Audio format is an audio file format for storing uncompressed audio in raw form. Comparable to WAV or AIFF in size, RAW Audio file does not include any header information (sampling rate, bit depth, endian, or number of channels). |
| RF64: | A <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Container_format_(digital)">container</a> format, based on the Microsoft RIFF/WAVE format and Wave (WAV) Format. It allows for more than 4 GB file sizes when needed (the maximum filesize is now approximately 16 exabytes, which is effectively unlimited. |
| WAV: | A <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Container_format_(digital)">container</a> format, almost always used for <a href="#lossless">lossless</a>, <a href="#uncompressed_format">uncompressed</a>, <a href="#pcm">PCM</a> audio. The format is in Microsoft's <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Endianness#Little-endian">Little-Endian</a> byte order. |
| WMA: | A <a rel="nofollow" class="external text" href="http://en.wikipedia.org/wiki/Container_format_(digital)">container</a> format. Windows Media Audio is a <a href="#lossy">lossy</a>, <a href="#compressed_format">size-compressed</a> 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 <a href="#mp3">MP3</a> and RealAudio codecs. |