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<u>{{Top-Level Crumbs|6}}{{Recording Crumbs|1}} '''Digital Audio</u> ''' is the representation of sound waves digitally, such as on consumer audio CD's and downloadable .mp3's, as well as audio events inside DAW tracks and samples for digital samplers.
The number of samples per unit of time is called the <u>sample rate</u>, typically represented in kHz, which is the number of thousand samples per second. The sample rate determines the highest frequency that can be accurately represented, which is called the <u>Nyquist Frequency</u>, occurring at half the sample rate. With a 44.1 kHZ rate, the highest-representable frequency is 22.05 kHZ, which is roughly the maximum frequency that humans are capable of hearing. Typical values are 16-bit/44.1 kHZ (used on typical consumer CD's) and 24-bit/96 kHZ (for commercial recording). Each frequency above the Nyquist frequency is will be digitized identically to another one below the Nyquist frequency. Similarly, when the digital signal is converted to analog, each frequency represented digitally will produce additional frequencies above the Nyquist frequency. These "phantom" frequencies that are produced by analog-to-digital (A>D) and digital-to-analog conversion are called <u>'''aliases</u> ''' and their creation is called <u>'''aliasing</u>'''. ''A>D and D>A converters use low-pass filters to remove frequencies above the Nyquist frequency to prevent aliasing.''
ItContainers and codecs can be '''open''' or '''proprietary'''s important to understand that file . Authors of open formats do not necessarily indicate disclose the exact codec used to encode technical details of the audio they contain. They are best thought of as <u>containers</u>, with a defined header format. This allows various software to know how codecs and freely-license it to extract the header information, where to find the audio data in the fileanyone else, allowing other containers/codecs and how software to decode incorporate it. A <u>codec</u> is what is actually used to en<u>co</u>de or <u>dec</u>ode Others are proprietary, withholding technical details from the audiopublic and only offering restrictive licensing, if any. Most As digital audio player software has its own codecsbecame a popular form of consumer music, while some can refer to external software. Most file Apple and Microsoft created rivaling proprietary formats are limited, however, seeking to support only a few codecs. Thuscapture markets for distributing and playing digital audio, it is common to associate compression/encoding schemes with preventing each other's PC operating systems and digital audio players from creating or playing each other's file formats.
<u>'''Compression</u> ''' uses various techniques to reduce the file size of digital audio. The main concern with compression is whether it noticeably degrades the audio signal - many schemes result in a loss of quality, known as <u>'''lossy</u> ''' compression. Such compression is based upon human perception - reducing detail in places that it is least likely be noticed (ex. replacing multiple cymbal hits with high-frequency noise). Of course, different implementations may be more noticeable than others; and many feature different settings, allowing trade-offs between the compression ratio and quality of audio. Different types of compression are not necessarily firm, either. For example, the common "mp3" compression is subject to interpretation, with some implementations considered better (less evident in audio playback for the same compression ratio) than others.
<u>'''Lossless</u> ''' compression, on the other hand, results in no signal degradation, and the file can be converted back and forth from compressed and uncompressed formats without any change in the sample data. Compared to lossy compression, it typically features a much smaller compression ratio, and thus results in larger file sizes.
<u>'''Digital Rights Management</u> ''' (DRM) refers to various methods that attempt to restrict decoding media files to authorized devices/users, preventing unlicensed file sharing and piracy. Many web stores still use forms of DRM, although it is now less common for music. DRM typically only allows playback through approved devices and software. No standardized, cross-brand DRM scheme was ever adapted, leading to significant consumer frustration. At the same time, most schemes were easily defeated. Thus, most of the music industry abandoned its use around 2009.
→File Extensions/Specific Formats
== Basic Principles ==
The most basic way to think of digital audio is like animation or video. By playing , which switches a number of individual pictures called "frames" quickly enough, it gives the appearance to create an illusion of motion. In the case of audio, the frames are called <u>'''samples</u>. Unlike video, '''; however, digital audio is not an illusion - the digital representation of the sound is capable of storing all the information contained in and can be converted back into required to reproduce identical sound waveswithin a given frequency range, at least theoretically. The amount of information a digital audio signal can represent is determined by the ''bit depth'' and the ''sample rate''. '''Bit depth''' is the number of bits per sample. Each sample represents the ''amplitude'' (the relative air pressure above or below a neutral level) of the sound wave at that point in time. The higher the bit-depth, the more distinct values are possible for each sample, allowing more and more accuracy in recording. If the bit depth is significantly low, the values for each sample will be quite "off" from the analog wave. When such digital audio is converted back to analog, the resulting waveform will have a significant difference from the original. This is called '''quantization distortion''', and it can result in increased noise and added frequencies in addition to simply loss of fidelity. Sometimes this effect is used purposefully (usually in electronic music), referred to as '''bitcrushing'''. Consumer audio typically uses 16-bits of precision per sample (audio CD), representing 2<sup>16</sup> or 65,536 distinct values. Commercial music production typically uses 24-bits, representing 2<sup>24</sup> or 16,777,216 values.
The amount number of samples per unit of information a digital audio signal can represent time is determined by the bit depth and called the '''sample rate. <u>Bit depth</u> ''', typically represented in kHz, which is the number of bits thousand samples per samplesecond. Each The sample represents rate determines the amplitude of highest frequency that can be accurately represented, which is called the sound wave '''Nyquist frequency''', occurring at that point in timehalf the sample rate. The higher With a 44.1 kHz rate, the bithighest-depthrepresentable frequency is 22.05 kHz, which is roughly the more distinct values maximum frequency that humans are possible for each sample, allowing more and more accuracy in recordingcapable of hearing. If the bit Typical depth is significantly low, the /rate values are 16-bit/44.1 kHz (used on typical consumer CD's) and 24-bit/96 kHz (for each sample will be quite "off" from the analog wavecommercial recording). When such digital audio is converted back Modern commercial hardware even supports up to analog, the resulting waveform will have a significant difference from the original. This is called <u>quantization distortion</u>192 kHz, and it can result some devices may operate at much higher rates in increased noise and added frequencies in addition order to simply loss of fidelity. Sometimes this effect is used purposefully (usually in electronic music)reproduce physical processes whose operation may affect lower-pitched, referred to as bitcrushingaudible audio frequencies.
While digital audio is theoretically capable of perfectly representing analog sound, in practice, devices can inaccurately measure amplitude or mis-time a sample. The quality of such devices is paramount to accurate recording and reproduction of sound. Commercial audio production uses bit depths and sample rates higher than the final mastered product to ensure any error is minimized. While there is no need for frequencies between 22.05 kHZ kHz and 48 kHZ 96 kHz (as no human can hear them), the higher sample rate prevents signal degradation as digital audio streams are manipulated and mixed together.
What has driven digital audio to almost completely replace analog formats in both the production and consumer side of the music industry? The best answer is its ability to be copied infinitely without any loss of quality. Recorded takes can be copied, sliced up, rearranged, manipulated, mixed-down, etc. without worry of any loss of quality. Record companies can distribute the master copy of an album to consumers without any degradation as it is transferred to a physical medium. It has also reduced costs. The price of transferring a compressed audio file from a server to a home computer is miniscule while unlike transferring data to and from magnetic tape reels or creating plates to press vinyl records is not. Similarly, rather than having to record each take in a recording studio on expensive high-quality magnetic tape that would degrade in quality with erase and rewrite operations, recording audio digitally onto a hard drive is virtually cost-less. There's also no time costs in terms of rewinding, erasing, etc. - computers manipulate digital data almost instantaneously.
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Digital audio can be represented by a number of different file formats, which mainly pertain to the compression scheme they use (or at least support), as well as additional features such as Digital Rights Management (DRM).
Compact Discs or CD's stored <u>'''LPCM</u> ''' audio at a set format. This means every CD player expected uncompressed audio with a bit depth of 16 bits and 44.1 kHz sample rate. Digital audio eventually expanded far beyond this medium, and players and software became smarter and more varied as far as what kinds of audio they could play. It's important to understand that file formats do not necessarily indicate the exact codec used to encode the audio they contain. They are best thought of as '''containers''', with a defined '''header''' format. This allows various software to know how to extract the header information, where to find the audio data in the file, and how to decode it. A '''codec''' is what is actually used to en'''co'''de or '''dec'''ode the audio. Most audio player software has its own codecs, while some can refer to external software. Most file formats are limited, however, to support only a few codecs. Thus, it is common to associate compression/encoding schemes with file formats.
=== File Extensions/Specific Formats ===The main formats for storing uncompressed audio are .wav (for MS Windows), .aiff (for Apple Mac), and .au (Java, Unix). The primary lossy compression formats are .mp3, .m4a, .ogg, .aac, and .wma. A popular lossless file types type is .flac. Windows media audio (.wma) supports both lossy and lossless encoding; however, it is generally assumed to use the lossy codec. .m4p and .wma are the primary formats of DRM-enabled audio files - iTunes switched from .m4p to .m4a files when it stopped using DRM. Almost all of these file types are capable of additional audio formats; however, these are their most common usages, and you can assume the type of audio contained in the file from the file type. {| class="wikitable"! Format !! Extension !! Compression !! Lossless !! Platform !! DRM-support|-| Wave || .wav || no || yes|| MS Windows || no|-| Audio Interchange File Format || .aiff || no || yes|| Apple Macintosh || no|-| Audio Unit || .au || no || yes|| Java, Unix (Linux, FreeBSD) || no|-| MPEG-2 Audio Layer III || .mp3 || yes || no|| any || no|-| MPEG-4 Audio Part 14 || .mp4, .m4a, .m4p || yes || no|| any (iTunes DRM-free) || yes|-| Advanced Audio Coding || .aac, .m4p, .m4a, .3gp, .mp4 || yes || rarely || any (including many non-PC devices, iTunes DRM) || yes|-| Vorbis || .ogg || yes || no || any (open standard) || usually not|-| Windows Media Audio || .wma, .asf || yes || optional || MS Windows || yes|-| Apple Lossless (ALAC) || .m4a || yes || yes || formerly Apple Mac only, now any || yes|-| Free Lossless Audio Codec || .flac || yes || yes || any || no|}
DAW softare typically works exclusively with uncompressed audio files. Each DAW may have encoders to create various file types with various encoding during rendering or bouncing, and you may be able to import other formats into a project; but internally the DAW will prefer to work with raw audio data. Imported audio is converted to uncompressed formats and stored as such. This prevents it from devoting resources to encoding or decoding audio unnecessarily.
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[[Category:DAW]]
[[Category:Sound]]