Understanding Digital Audio - Revision history

Meambobbo: /* File Extensions/Specific Formats */

2016-06-11T03:20:11Z

‎File Extensions/Specific Formats

Meambobbo: /* File Formats */

2016-06-11T03:14:10Z

‎File Formats

Meambobbo: /* File Extensions/Specific Formats */

2016-06-11T02:48:45Z

‎File Extensions/Specific Formats

Meambobbo: /* File Extensions/Specific Formats */

2016-06-11T02:48:15Z

‎File Extensions/Specific Formats

Meambobbo: /* File Extensions/Specific Formats */

2016-06-11T02:28:10Z

‎File Extensions/Specific Formats

Meambobbo: /* File Extensions/Specific Formats */

2016-06-11T02:27:47Z

‎File Extensions/Specific Formats

Meambobbo: /* Basic Principles */

2015-10-13T01:20:57Z

‎Basic Principles

Meambobbo: /* Basic Principles */

2015-10-13T00:28:57Z

‎Basic Principles

Meambobbo at 00:16, 13 October 2015

2015-10-13T00:16:48Z

Meambobbo: /* Basic Principles */

2015-10-08T02:58:28Z

‎Basic Principles

@@ Line 40: / Line 40: @@
 {| class="wikitable"
-! Format !! Extension !! Compression !! Lossy !! Platform !! DRM-support
+! Format !! Extension !! Compression !! Lossless !! Platform !! DRM-support
 |-
-| Wave || .wav || no || no || MS Windows || no
+| Wave || .wav || no || yes|| MS Windows || no
 |-
-| Audio Interchange File Format || .aiff || no || no || Apple Macintosh || no
+| Audio Interchange File Format || .aiff || no || yes|| Apple Macintosh || no
 |-
-| Audio Unit || .au || no || no || Java, Unix (Linux, FreeBSD) || no
+| Audio Unit || .au || no || yes|| Java, Unix (Linux, FreeBSD) || no
 |-
-| MPEG-2 Audio Layer III || .mp3 || yes || yes || any || no
+| MPEG-2 Audio Layer III || .mp3 || yes || no|| any || no
 |-
-| MPEG-4 Audio Part 14 || .mp4, .m4a, .m4p || yes || yes || any (iTunes DRM-free) || yes
+| MPEG-4 Audio Part 14 || .mp4, .m4a, .m4p || yes || no|| any (iTunes DRM-free) || yes
 |-
-| Advanced Audio Coding || .aac, .m4p, .m4a, .3gp, .mp4 || yes || usually|| any (including many non-PC devices, iTunes DRM) || yes
+| Advanced Audio Coding || .aac, .m4p, .m4a, .3gp, .mp4 || yes || rarely || any (including many non-PC devices, iTunes DRM) || yes
 |-
-| Vorbis || .ogg || yes || yes || any (open standard) || usually not
+| Vorbis || .ogg || yes || no || any (open standard) || usually not
 |-
 | Windows Media Audio || .wma, .asf || yes || optional || MS Windows || yes
 |-
-| Apple Lossless (ALAC) || .m4a || yes || no || formerly Apple Mac only, now any || yes
+| Apple Lossless (ALAC) || .m4a || yes || yes || formerly Apple Mac only, now any || yes
 |-
-| Free Lossless Audio Codec || .flac || yes || no || any || no
+| Free Lossless Audio Codec || .flac || yes || yes || any || no
 |}

← Older revision		Revision as of 03:14, 11 June 2016
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	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.		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.
		+
		+	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.		'''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.

@@ Line 48: / Line 48: @@
 | MPEG-2 Audio Layer III || .mp3 || yes || yes || any || no
 |-
-| MPEG-4 Audio Part 14 || .mp4, .m4a, .m4p || yes || yes || any (iTunes DRM-free) || no
+| MPEG-4 Audio Part 14 || .mp4, .m4a, .m4p || yes || yes || any (iTunes DRM-free) || yes
 |-
 | Advanced Audio Coding || .aac, .m4p, .m4a, .3gp, .mp4 || yes || usually|| any (including many non-PC devices, iTunes DRM) || yes

@@ Line 48: / Line 48: @@
 | MPEG-2 Audio Layer III || .mp3 || yes || yes || any || no
 |-
-| MPEG 4 Audio || .m4a || yes || yes || any (iTunes DRM-free) || no
+| MPEG-4 Audio Part 14 || .mp4, .m4a, .m4p || yes || yes || any (iTunes DRM-free) || no
 |-
 | Vorbis || .ogg || yes || yes || any (open standard) || usually not
 |-
-| Advanced Audio Coding || .aac || yes || yes || any (including many non-PC devices, iTunes DRM) || yes
+| Windows Media Audio || .wma, .asf || yes || optional || MS Windows || yes
 |-
-| Windows Media Audio || .wma || yes || optional || MS Windows || yes
+| Apple Lossless (ALAC) || .m4a || yes || no || formerly Apple Mac only, now any || yes
 |-
 | Free Lossless Audio Codec || .flac || yes || no || any || no

@@ Line 37: / Line 37: @@
 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 (.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 !! Lossy !! Platform !! DRM-support
 |-

@@ Line 36: / Line 36: @@
 === 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 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.
 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.

@@ Line 9: / Line 9: @@
 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 2^16 or 65,536 distinct values.  Commercial music production typically uses 24-bits, representing 2^24 or 16,777,216 values.
+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 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 much higher rates in order to reproduce physical processes whose operation may affect lower-pitched, audible audio frequencies.

@@ Line 5: / Line 5: @@
 == Basic Principles ==
-The most basic way to think of digital audio is like animation or video.  By playing a number of individual "frames" quickly enough, it gives the appearance of motion.  In the case of audio, the frames are called '''samples'''.  Unlike video, however, digital audio is not an illusion - the digital representation of the sound is capable of storing all the information required to reproduce identical sound waves within a given frequency range, at least theoretically.
+The most basic way to think of digital audio is like animation or video, which switches a number of individual pictures called "frames" quickly enough to create an illusion of motion.  In the case of audio, the frames are called '''samples'''; however, digital audio is not an illusion - the digital representation of the sound is capable of storing all the information 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'' 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'''.
+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'''.
 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 much higher rates in order to reproduce physical processes whose operation may affect lower-pitched, audible audio frequencies.

@@ Line 2: / Line 2: @@
 {{Recording Crumbs|1}}
-'''Digital Audio''' is the representation of sound waves digitally, such as on consumer audio CD's and downloadable .mp3's.
+'''Digital Audio''' 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.
 == Basic Principles ==