Audio compression (data)

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This article is about a process which reduces the data rate or storage size of digital audio
signals. For processes which reduce the dynamic range (without changing the amount of
digital data) of audio signals, see dynamic range compression.
For processes which reduce the amount of time it takes to listen to and understand a
recording, see time-compressed speech.
Audio compression is a form of data compression designed to reduce the transmission
bandwidth requirement of digital audio streams and the storage size of audio files. Audio
compression algorithms are implemented in computer software as audio codecs. Generic data
compression algorithms perform poorly with audio data, seldom reducing data size much
below 87% from the original,[citation needed] and are not designed for use in real time applications.
Consequently, specifically optimized audio lossless and lossy algorithms have been created.
Lossy algorithms provide greater compression rates and are used in mainstream consumer
audio devices.
In both lossy and lossless compression, information redundancy is reduced, using methods
such as coding, pattern recognition and linear prediction to reduce the amount of information
used to represent the uncompressed data.
The trade-off between slightly reduced audio quality and transmission or storage size is
outweighed by the latter for most practical audio applications in which users may not
perceive the loss in playback rendition quality. For example, one compact disk (CD) holds
approximately one hour of uncompressed high fidelity music, less than 2 hours of music
compressed losslessly, or 7 hours of music compressed in the MP3 format at medium bit
rates.

Contents
[hide]
• 1 Lossless audio compression
○ 1.1 Applications
○ 1.2 Formats
○ 1.3 Difficulties in lossless compression of audio data
○ 1.4 Evaluation criteria
• 2 Lossy audio compression
○ 2.1 Coding methods
 2.1.1 Transform domain methods
 2.1.2 Time domain methods
○ 2.2 Applications
○ 2.3 Usability
○ 2.4 Speech encoding
• 3 History
• 4 Glossary
 • 5 See also
• 6 References
• 7 External links

[edit] Lossless audio compression

Lossless audio compression produces a representation of digital data that can be expanded to
an exact digital duplicate of the original audio stream. This is in contrast to the irreversible
changes upon playback from lossy compression techniques such as Vorbis and MP3.
Compression ratios are similar to those for generic lossless data compression (around 50–
60% of original size), and substantially less than for lossy compression, which typically yield
5–20% of original size.[citation needed]
[edit] Applications
The primary application areas of lossless encoding are:
Archives
For archival purposes it is generally desired to maintain best possible quality.
Editing
Audio engineers use lossless compression for audio editing to avoid digital generation
loss.
High fidelity playback
Audiophiles prefer lossless compression formats to avoid compression artifacts.
Mastering of casual-use audio media
High quality master copies of recordings are used to produce lossily compressed
versions for digital audio players. As formats and encoders improve, updated lossily
compressed files may be generated from the lossless master.
As file storage and communications bandwidth have become less expensive and more
available, lossless audio compression has become more popular.[citation needed]
[edit] Formats
Shorten was an early lossless format; newer ones include Free Lossless Audio Codec
(FLAC), Apple's Apple Lossless, MPEG-4 ALS, Monkey's Audio, and TTA.
Some audio formats feature a combination of a lossy format and a lossless correction; this
allows stripping the correction to easily obtain a lossy file. Such formats include MPEG-4
SLS (Scalable to Lossless), WavPack, and OptimFROG DualStream.
Some formats are associated with a technology, such as:
• Direct Stream Transfer, used in Super Audio CD
• Meridian Lossless Packing, used in DVD-Audio, Dolby TrueHD, Blu-ray and HD
DVD
[edit] Difficulties in lossless compression of audio data
It is difficult to maintain all the data in an audio stream and achieve substantial compression.
First, the vast majority of sound recordings are highly complex, recorded from the real world.
As one of the key methods of compression is to find patterns and repetition, more chaotic
data such as audio doesn't compress well. In a similar manner, photographs compress less
efficiently with lossless methods than simpler computer-generated images do. But
interestingly, even computer generated sounds can contain very complicated waveforms that
present a challenge to many compression algorithms. This is due to the nature of audio
waveforms, which are generally difficult to simplify without a (necessarily lossy) conversion
to frequency information, as performed by the human ear.
The second reason is that values of audio samples change very quickly, so generic data
compression algorithms don't work well for audio, and strings of consecutive bytes don't
generally appear very often. However, convolution with the filter [-1 1] (that is, taking the
first difference) tends to slightly whiten (decorrelate, make flat) the spectrum, thereby
allowing traditional lossless compression at the encoder to do its job; integration at the
decoder restores the original signal. Codecs such as FLAC, Shorten and TTA use linear
prediction to estimate the spectrum of the signal. At the encoder, the estimator's inverse is
used to whiten the signal by removing spectral peaks while the estimator is used to
reconstruct the original signal at the decoder.
[edit] Evaluation criteria
Lossless audio codecs have no quality issues, so the usability can be estimated by
• Speed of compression and decompression
• Degree of compression
• Robustness and error correction
• Product support
[edit] Lossy audio compression
Lossy audio compression is used in an extremely wide range of applications. In addition to
the direct applications (mp3 players or computers), digitally compressed audio streams are
used in most video DVDs; digital television; streaming media on the internet; satellite and
cable radio; and increasingly in terrestrial radio broadcasts. Lossy compression typically
achieves far greater compression than lossless compression (data of 5 percent to 20 percent of
the original stream, rather than 50 percent to 60 percent), by discarding less-critical data.
The innovation of lossy audio compression was to use psychoacoustics to recognize that not
all data in an audio stream can be perceived by the human auditory system. Most lossy
compression reduces perceptual redundancy by first identifying sounds which are considered
perceptually irrelevant, that is, sounds that are very hard to hear. Typical examples include
high frequencies, or sounds that occur at the same time as louder sounds. Those sounds are
coded with decreased accuracy or not coded at all.
While removing or reducing these 'unhearable' sounds may account for a small percentage of
bits saved in lossy compression, the real savings comes from a complementary phenomenon:
noise shaping. Reducing the number of bits used to code a signal increases the amount of
noise in that signal. In psychoacoustics-based lossy compression, the real key is to 'hide' the
noise generated by the bit savings in areas of the audio stream that cannot be perceived. This
is done by, for instance, using very small numbers of bits to code the high frequencies of
most signals - not because the signal has little high frequency information (though this is also
often true as well), but rather because the human ear can only perceive very loud signals in
this region, so that softer sounds 'hidden' there simply aren't heard.
If reducing perceptual redundancy does not achieve sufficient compression for a particular
application, it may require further lossy compression. Depending on the audio source, this
still may not produce perceptible differences. Speech for example can be compressed far
more than music. Most lossy compression schemes allow compression parameters to be
adjusted to achieve a target rate of data, usually expressed as a bit rate. Again, the data
reduction will be guided by some model of how important the sound is as perceived by the
human ear, with the goal of efficiency and optimized quality for the target data rate. (There
are many different models used for this perceptual analysis, some better suited to different
types of audio than others.) Hence, depending on the bandwidth and storage requirements, the
use of lossy compression may result in a perceived reduction of the audio quality that ranges
from none to severe, but generally an obviously audible reduction in quality is unacceptable
to listeners.
Because data is removed during lossy compression and cannot be recovered by
decompression, some people may not prefer lossy compression for archival storage. Hence,
as noted, even those who use lossy compression (for portable audio applications, for
example) may wish to keep a losslessly compressed archive for other applications. In
addition, the technology of compression continues to advance, and achieving a state-of-the-
art lossy compression would require one to begin again with the lossless, original audio data
and compress with the new lossy codec. The nature of lossy compression (for both audio and
images) results in increasing degradation of quality if data are decompressed, then
recompressed using lossy compression.
[edit] Coding methods
[edit] Transform domain methods
In order to determine what information in an audio signal is perceptually irrelevant, most
lossy compression algorithms use transforms such as the modified discrete cosine transform
(MDCT) to convert time domain sampled waveforms into a transform domain. Once
transformed, typically into the frequency domain, component frequencies can be allocated
bits according to how audible they are. Audibility of spectral components is determined by
first calculating a masking threshold, below which it is estimated that sounds will be beyond
the limits of human perception.
The masking threshold is calculated using the absolute threshold of hearing and the principles
of simultaneous masking - the phenomenon wherein a signal is masked by another signal
separated by frequency - and, in some cases, temporal masking - where a signal is masked by
another signal separated by time. Equal-loudness contours may also be used to weight the
perceptual importance of different components. Models of the human ear-brain combination
incorporating such effects are often called psychoacoustic models.
[edit] Time domain methods
Other types of lossy compressors, such as the linear predictive coding (LPC) used with
speech, are source-based coders. These coders use a model of the sound's generator (such as
the human vocal tract with LPC) to whiten the audio signal (i.e., flatten its spectrum) prior to
quantization. LPC may also be thought of as a basic perceptual coding technique;
reconstruction of an audio signal using a linear predictor shapes the coder's quantization noise
into the spectrum of the target signal, partially masking it.
[edit] Applications
Due to the nature of lossy algorithms, audio quality suffers when a file is decompressed and
recompressed (digital generation loss). This makes lossy compression unsuitable for storing
the intermediate results in professional audio engineering applications, such as sound editing
and multitrack recording. However, they are very popular with end users (particularly MP3),
as a megabyte can store about a minute's worth of music at adequate quality.
[edit] Usability
Usability of lossy audio codecs is determined by:
• Perceived audio quality
• Compression factor
• Speed of compression and decompression
 • Inherent latency of algorithm (critical for real-time streaming applications; see below)
• Product support
Lossy formats are often used for the distribution of streaming audio, or interactive
applications (such as the coding of speech for digital transmission in cell phone networks). In
such applications, the data must be decompressed as the data flows, rather than after the
entire data stream has been transmitted. Not all audio codecs can be used for streaming
applications, and for such applications a codec designed to stream data effectively will
usually be chosen.
Latency results from the methods used to encode and decode the data. Some codecs will
analyze a longer segment of the data to optimize efficiency, and then code it in a manner that
requires a larger segment of data at one time in order to decode. (Often codecs create
segments called a "frame" to create discrete data segments for encoding and decoding.) The
inherent latency of the coding algorithm can be critical; for example, when there is two-way
transmission of data, such as with a telephone conversation, significant delays may seriously
degrade the perceived quality.
In contrast to the speed of compression, which is proportional to the number of operations
required by the algorithm, here latency refers to the number of samples which must be
analysed before a block of audio is processed. In the minimum case, latency is 0 zero samples
(e.g., if the coder/decoder simply reduces the number of bits used to quantize the signal).
Time domain algorithms such as LPC also often have low latencies, hence their popularity in
speech coding for telephony. In algorithms such as MP3, however, a large number of samples
have to be analyzed in order to implement a psychoacoustic model in the frequency domain,
and latency is on the order of 23 ms (46 ms for two-way communication).
[edit] Speech encoding
Speech encoding is an important category of audio data compression. The perceptual models
used to estimate what a human ear can hear are generally somewhat different from those used
for music. The range of frequencies needed to convey the sounds of a human voice are
normally far narrower than that needed for music, and the sound is normally less complex. As
a result, speech can be encoded at high quality using relatively low bit rates.
This is accomplished, in general, by some combination of two approaches:
• Only encoding sounds that could be made by a single human voice.
• Throwing away more of the data in the signal—keeping just enough to reconstruct an
"intelligible" voice rather than the full frequency range of human hearing.
Perhaps the earliest algorithms used in speech encoding (and audio data compression in
general) were the A-law algorithm and the µ-law algorithm. Protocol now requires for 7-Zip
programmes to stop compressing audio files, due to legal reasons.[citation needed]
[edit] History
Solidyne 922: The world's first commercial audio bit compression card for PC, 1990
A literature compendium for a large variety of audio coding systems was published in the
IEEE Journal on Selected Areas in Communications (JSAC), February 1988. While there
were some papers from before that time, this collection documented an entire variety of
finished, working audio coders, nearly all of them using perceptual (i.e. masking) techniques
and some kind of frequency analysis and back-end noiseless coding.[1] Several of these papers
remarked on the difficulty of obtaining good, clean digital audio for research purposes. Most,
if not all, of the authors in the JSAC edition were also active in the MPEG-1 Audio
committee.
The world's first commercial broadcast automation audio compression system was developed
by Oscar Bonello, an Engineering professor at the University of Buenos Aires.[2] In 1983,
using the psychoacoustic principle of the masking of critical bands first published in 1967,[3]
he started developing a practical application based on the recently developed IBM PC
computer, and the broadcast automation system was launched in 1987 under the name
Audicom. 20 years later, almost all the radio stations in the world were using similar
technology, manufactured by a number of companies