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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 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 required to reproduce identical sound waves within 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 can 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 into identical sound wavesto 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 2¹⁶ or 65,536 distinct values. Commercial music production typically uses 24-bits, representing 2²⁴ or 16,777,216 values. The number of samples per unit of time is called the '''sample rate''', 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 '''Nyquist frequency''', 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 depth/rate values are 16-bit/44.1 kHz (used on typical consumer CD's) and 24-bit/96 kHz (for commercial recording). Modern commercial hardware even supports up to 192 kHz, and some devices may operate at least theoreticallymuch higher rates in order to reproduce physical processes whose operation may affect lower-pitched, audible audio frequencies.

The amount of sound information a digital audio signal can represent is determined by the bit depth and the sample rate. <u>Bit depth</u> is the number of bits per sample. Each sample represents frequency above the amplitude of Nyquist frequency will be digitized identically to another one below the sound wave at that point in timeNyquist frequency. The higher the bit-depthSimilarly, when the more distinct values are possible for each sample, allowing more and more accuracy in recording. The number of samples per unit of time digital signal is called the <u>sample rate</u>converted to analog, typically each frequency represented in kHz, which is digitally will produce additional frequencies above the number of thousand samples per second. The sample rate determines the highest Nyquist frequency that can be accurately represented. The <uThese "phantom" frequencies that are produced by analog-to-digital (A>Nyquist Frequency</u> is 1/2 the sample rate D) and digital-to-analog conversion are called '''aliases''' and their creation is the highest-pitched frequency that can be accurately representedcalled '''aliasing'''. ''A sample rate of 44.1 kHZ means the highest>D and D>A converters use low-representable frequency is 22.05 kHZ, which is roughly pass filters to remove frequencies above the maximum Nyquist frequency that humans are capable of hearingto prevent aliasing. Typical values are 16-bit/44.1 kHZ (used on typical consumer CD's) and 24-bit/96 kHZ (for commercial recording). '

A device called an <u>analog-to-While digital converter</u> (ADC) turns audio is theoretically capable of perfectly representing analog electrical waves into digital signals. Converselysound, in practice, a <u>digital-todevices can inaccurately measure amplitude or mis-analog converter</u> (DAC) turns digital audio into analog electrical signals. ADC's and DAC's typically feature time a low-pass filter to remove high-frequencies that cannot be accurately represented by their maximum sample rate. This The quality of such devices is at the Nyquist frequencyparamount to accurate recording and reproduction of sound. Without filtering out such frequencies during Commercial audio production uses bit depths and sample rates higher than the analog-final mastered product to-digital conversion, certain frequencies may be incorrectly represented as lower frequencies, which ensure any error is called <u>aliasing</u>minimized. Similarly, the digital-to-analog conversion will produce frequencies above the Nyquist frequency which are aliases of lower While there is no need for frequenciesbetween 22. They should be filtered out for accuracy 05 kHz and 96 kHz (howeveras no human can hear them), it shouldn't matter if the Nyquist frequency exceeds the range of human hearing)higher sample rate prevents signal degradation as digital audio streams are manipulated and mixed together.

While 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 theoretically capable its ability to be copied infinitely without any loss of perfectly representing analog soundquality. Recorded takes can be copied, sliced up, rearranged, manipulated, in practicemixed-down, devices etc. without worry of any loss of quality. Record companies can inaccurately measure amplitude or mis-time distribute the master copy of an album to consumers without any degradation as it is transferred to a samplephysical medium. It has also reduced costs. The quality price of such devices transferring a compressed audio file from a server to a home computer is paramount miniscule unlike transferring data to accurate recording and reproduction of soundfrom magnetic tape reels or creating plates to press vinyl records. Commercial audio production uses bit depths and sample rates higher Similarly, rather than the final mastered product having to ensure any error 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 minimizedvirtually cost-less. While there is There's also no need for frequencies between 22time costs in terms of rewinding, erasing, etc.05 kHZ and 48 kHZ (as no human can hear them), the higher sample rate prevents signal degradation as - computers manipulate digital audio streams are manipulated and mixed togetherdata almost instantaneously.

Compact Discs or CD's stored '''LPCM''' 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 <u>'''containers</u>''', 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 <u>'''codec</u> ''' is what is actually used to en<u>'''co</u>'''de or <u>'''dec</u>'''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. Containers and codecs can be '''open''' or '''proprietary'''. Authors of open formats disclose the technical details of the containers/codecs and freely-license it to anyone else, allowing other containers/codecs and software to incorporate it. Others are proprietary, withholding technical details from the public and only offering restrictive licensing, if any. As digital audio became a popular form of consumer music, Apple and Microsoft created rivaling proprietary formats seeking to capture markets for distributing and playing digital audio, preventing each other's PC operating systems and digital audio players from creating or playing each other's file formats. '''Compression''' 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 '''lossy''' 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>Compression</u> uses various techniques to reduce the file size of digital audio. The main concern with '''Lossless''' compression is whether it noticeably degrades , on the audio other hand, results in no signal - many schemes result degradation, and the file can be converted back and forth from compressed and uncompressed formats without any change in a loss of quality, known as <u>lossy</u> compressionthe sample data. Such Compared to lossy compression is based upon human perception - reducing detail in places that , it is least likely be noticed. Of course, different implementations may be more noticeable than others; and many feature different settings, allowing trade-offs between the typically features a much smaller compression ratio , and quality of audiothus results in larger file sizes.

<u>Lossless<'''Digital Rights Management''' (DRM) refers to various methods that attempt to restrict decoding media files to authorized devices/u> compressionusers, on the other hand, results in no signal degradation, and the preventing unlicensed file can be converted back sharing and forth from compressed 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 uncompressed formats without any change in the sample datasoftware. Compared No standardized, cross-brand DRM scheme was ever adapted, leading to lossy compressionsignificant consumer frustration. At the same time, it typically features a much smaller compression ratiomost schemes were easily defeated. Thus, and thus results in larger file sizesmost of the music industry abandoned its use around 2009.

<u>Digital Rights Management<=== File Extensions/u> 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 type is .flac. Windows media audio (DRM.wma) refers supports both lossy and lossless encoding; however, it is generally assumed to various methods that attempt to restrict decoding media use the lossy codec. .m4p and .wma are the primary formats of DRM-enabled audio files - iTunes switched from .m4p to authorized devices/users. ITunes used m4a files when it stopped using DRM for a while before abandoning its use. Many web stores still use forms Almost all of these file types are capable of DRMadditional audio formats; however, although it is now less these are their most common for musicusages, and you can assume the type of audio contained in the file from the file type.

The main formats for storing uncompressed audio are {| class="wikitable"! Format !! Extension !! Compression !! Lossless !! Platform !! DRM-support|-| Wave || .wav (for || no || yes|| MS Windows), || no|-| Audio Interchange File Format || .aiff (for || no || yes|| Apple Mac), and Macintosh || no|-| Audio Unit || .au (|| no || yes|| Java, Unix(Linux, FreeBSD)|| no|-| MPEG-2 Audio Layer III || . The primary lossy compression formats are mp3 || yes || no|| any || no|-| MPEG-4 Audio Part 14 || .mp3mp4, .m4a, .oggm4p || yes || no|| any (iTunes DRM-free) || yes|-| Advanced Audio Coding || .aac, .aacm4p, and .wmam4a, . A popular lossless file types is 3gp, .flacmp4 || yes || rarely || any (including many non-PC devices, iTunes DRM) || yes|-| Vorbis || . ogg || yes || no || any (open standard) || usually not|-| Windows media audio (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 DRMasf || yes || optional || MS Windows || yes|-enabled audio files| Apple Lossless (ALAC) || . Almost all of these file types are capable of additional audio formats; howeverm4a || yes || yes || formerly Apple Mac only, these are their most common usagesnow any || yes|-| Free Lossless Audio Codec || .flac || yes || yes || any || no|}