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1. Introduction
This paper introduces digital audio signal compression, a technique essential to the
implementation of many digital audio applications. Digital audio signal compression is the
removal of redundant or otherwise irrelevant information from a digital audio signal, a
process that is useful for conserving both transmission bandwidth and storage space. We
begin by defining some useful terminology. We then present a typical “encoder” (as
compression algorithms are often called) and explain how it functions. Finally consider
some standards that employ digital audio signal compression, and discuss the future of the
field.
2. Terminology
This paper focuses on audio compression techniques, which differ from those used in
speech compression. Speech compression uses a model of the human vocal tract to express
particular signal in a compressed format. This technique is not usually applied in the field of
audio compression due to the vast array of sounds that can be generated – models that
represent audio generation would be too complex to implement. So instead of modeling the
source of sounds, modern audio compression models the receiver, i.e., the human ear.
Figure 1 shows a generic encoder or “compressor that takes blocks of sampled audio
signal as its input. These blocks typically consist of between 500 and 1500 samples per
channel, depending on the encoder specification. For example, the MPEG-1 layer III (MP3)
specification takes 576 samples per channel per input block. The output is a compressed
representation of the input block (a “frame”) that can be transmitted or stored for
subsequent decoding.
No matter what you do, your ears are always working. They are constantly detecting,
deciphering and analyzing sounds and communicating them to the brain. In a comparatively
tiny area of our body the ear is performing many highly technical and intricate functions.
There are three distinct portions to the ear: the outer ear containing the fleshy skin and the
canal that leads to the inner ear, the middle ear containing the three smallest bones in the
human body the malleus, incus and stapes (commonly called the hammer, anvil and stirrup)
and the inner ear, made up of a cluster of three semicircular canals and the snail shaped
cochlea. Let’s take a look at them one at a time…
Scientists cannot fully explain just how the signals are transmitted to the brain. They do
know that the signals sent by all the hair cells are about the same in duration and strength.
This has led them to believe that it is not the content of the signals but rather the signals
themselves that convey some sort of message to the brain.
Our ears, so often taken for granted, thus are a marvel of intricacy and design that leaves
anything that man can produce in the shade as a cheap imitation. Your hearing can never be
replaced. Don’t take it for granted.
5. Psychoacoustics
How do we reduce the size of the input data? The basic idea is to eliminate information
that is inaudible to the ear. This type of compression is often referred to as perceptual
encoding. To help determine what can and cannot be heard, compression algorithms rely on
the field of psychoacoustics, i.e., the study of human sound perception. Waves vibrating at
different frequencies manifest themselves differently, all the way from the astronomically
slow pulsations of the universe itself to the inconceivably fast vibration of matter (and
beyond). Somewhere in between these extremes are wavelengths that are perceptible to
human beings as light and sound. Just beyond the realms of light and sound are sub- and
ultrasonic vibration, the infrared and ultraviolet light spectra, and zillions of other
frequencies imperceptible to humans (such as radio and microwave). Our sense organs are
tuned only to very narrow bandwidths of vibration in the overall picture. In fact, even our
own musical instruments create many vibrational frequencies that are imperceptible to our
ears. Frequencies are typically described in units called Hertz (Hz), which translates simply
as "cycles per second." In general, humans cannot hear frequencies below 20Hz (20 cycles
per second), nor above 20kHz (20,000 cycles per second), as shown in Figure 2.
While hearing capacities vary from one individual to the next, it's generally true that
humans perceive midrange frequencies more strongly than high and low frequencies,[2] and
that sensitivity to higher frequencies diminishes with age and prolonged exposure to loud
volumes. In fact, by the time we're adults, most of us can't hear much of anything above
16kHz (although women tend to preserve the ability to hear higher frequencies later into life
than do men). The most sensitive range of hearing for most people hovers between 2kHz to
4kHz, a level probably evolutionarily related to the normal range of the human voice, which
runs roughly from 500Hz to 2kHz.
Specifically, audio compression algorithms exploit the conditions under which signal
characteristics obscure or mask each other. This phenomenon occurs in three different
ways: threshold cut-off, frequency masking and temporal masking. The remainder of this
section explains the nature of these concepts; subsequent sections explain how they are
typically applied to audio signal compression.
Threshold Cut-off
The human ear detects sounds as a local variation in air pressure measured as the Sound
Pressure Level (SPL). If variations in the SPL are below a certain threshold in amplitude,
the ear cannot detect them. This threshold, shown in Figure 3, is a function of the sound’s
frequency. Notice in Figure 3 that because the lowest-frequency component is below the
threshold, it will not be heard.
Frequency Masking
Even if a signal component exceeds the hearing threshold, it may still be masked by
louder components that are near it in frequency. This phenomenon is known as frequency
masking or simultaneous masking. Each component in a signal can cast a “shadow” over
neighbouring components. If the neighbouring components are covered by this shadow,
they will not be heard. The effective result is that one component, the masker, shifts the
hearing threshold. Figure 4 shows a situation in which this occurs.
Temporal Masking
Just as tones cast shadows on their neighbors in the frequency domain, a sudden
increase in volume can mask quieter sounds that are temporally close. This phenomenon is
known as temporal masking. Interestingly, sounds that occur both after and before the
volume increase can be masked! Figure 5 illustrates a typical temporal masking scenario:
events below the indicated threshold will not be heard. The idea behind temporal masking is
that humans also have trouble hearing distinct sounds that are close to one another in time.
For example, if a loud sound and a quiet sound are played simultaneously, you won't be
able to hear the quiet sound. If, however, there is sufficient delay between the two sounds,
you will hear the second, quieter sound. The key to the success of temporal masking is in
determining (quantifying) the length of time between the two tones at which the second
tone becomes audible, i.e., significant enough to keep it in the bitstream rather than
throwing it away. This distance, or threshold, turns out to be around five milliseconds when
working with pure tones, though it varies up and down in accordance with different audio
passages.
6. Spectral Analysis
Of the three masking phenomena explained above, two are best described in the
frequency domain. Thus, a frequency domain representation, also called the “spectrum” of a
signal, is a useful tool for analyzing the signal’s frequency characteristics and determining
thresholds. There are several different techniques for converting a finite time sequence into
its spectral representation, and these typically fall into one of two categories: transforms
and filter banks. Transforms calculate the spectrum of their inputs in terms of a set of basis
sequences; e.g., the Fourier Transform uses basic sequences that are complex exponentials.
Filter banks apply several different band pass filters to the input. Typically the result is
several time sequences, each of which corresponds to a particular frequency band. Taking
the spectrum of a signal has two purposes:
To derive the masking thresholds in order to determine which portion of the
signal can be dropped.
To generate a representation of the signal to which the masking threshold can
be applied.
Some compression schemes use different techniques for these two tasks.
The most popular transform in signal processing is the Fast Fourier Transform (FFT).
Given a finite time sequence, the FFT produces a complex-value frequency domain
representation. Encoders often use FFTs as a first step toward determining masking thresholds.
Another popular transform is the Discrete Cosine Transform (DCT), which outputs a real-
valued frequency domain representation. Both the FFT and the DCT suffer from distortion
when transforms are taken from contiguous blocks of time data. To solve this problem, inputs
and outputs can be overlapped and windowed in such a way that, in the absence of lossy
compression techniques, entire time signals can be perfectly reconstructed. For this reason,
most transform-based encoding schemes employ an overlapped and windowed DCT known as
the Modified Discrete Cosine Transform (MDCT). Some compression algorithms that use the
MDCT are MPEG-1 layer-III, MPEG-2 AAC, and Do
Dolby AC-3. Filter banks pass a block of time samples through several band pass filters to
generate
The purpose of this section is to discuss some existing standards in digital audio
compression, in particular the MPEG-1 layer III. Features of interest for each standard include
which compression techniques are used, special details or unique characteristics, and target
applications.
7.1.1 History
In 1987, the Fraunhofer IIS started to work on perceptual audio coding in the
framework of the EUREKA project EU147, Digital Audio Broadcasting (DAB). In a joint
cooperation with the University of Erlangen (Prof. Dieter Seitzer), the Fraunhofer IIS finally
devised a very powerful algorithm that is standardized as ISO-MPEG Audio Layer-3 (IS
11172-3 and IS 13818-3).
Without data reduction, digital audio signals typically consist of 16 bit samples recorded
at a sampling rate more than twice the actual audio bandwidth (e.g. 44.l1 KHz for Compact
Discs). So
you end up with more than 1.4 Mbit to represent just one second of stereo music in CD quality.
By using MPEG audio coding, you may shrink down the original sound data from a CD by a
factor of 12, without losing sound quality. Basically, this is realized by perceptual coding
techniques addressing the perception of sound waves by the human ear.
By exploiting stereo effects and by limiting the audio bandwidth, the coding schemes
may achieve an acceptable sound quality at even lower bit rates. MPEG Layer-3 is the most
powerful member of the MPEG audio coding family. For a given sound quality level, it
requires the lowest bit rate or for a given bit rate, it achieves the highest sound quality.
MP3 uses two compression techniques to achieve its size reduction ratios over
uncompressed audio-one lossy and one lossless. First it throws away what humans can't hear
anyway (or at least it makes acceptable compromises), and then it encodes the redundancies to
achieve further compression. However, it's the first part of the process that does most of the
grunt work, requires most of the complexity.
Perceptual codecs are highly complex beasts, and all of them work a little differently.
However, the general principles of perceptual coding remain the same from one codec to the
next. In
brief, the MP3 encoding process can be subdivided into a handful of discrete tasks (not
necessarily in this order):
• Break the signal into smaller component pieces called " frames," each typically lasting a
fraction of a second. You can think of frames much as you would the frames in a movie
film.
• Analyze the signal to determine its "spectral energy distribution." In other words, on the
entire spectrum of audible frequencies, find out how the bits will need to be distributed
to best account for the audio to be encoded. Because different portions of the frequency
spectrum are most efficiently encoded via slight variants of the same algorithm, this step
breaks the signal into sub-bands, which can be processed independently for optimal
results (but note that all sub-bands use the algorithm-they just allocate the number of
bits differently, as determined by the encoder).
• The encoding bitrate is taken into account, and the maximum number of bits that can be
allocated to each frame is calculated. For instance, if you're encoding at 128 kbps, you
have an upper limit on how much data can be stored in each frame (unless you're
encoding with variable bitrates, but we'll get to that later). This step determines how
much of the available audio data will be stored, and how much will be left on the cutting
room floor.
• The frequency spread for each frame is compared to mathematical models of human
psychoacoustics, which are stored in the codec as a reference table. From this model, it
• The bitstream is run through the process of " Huffman coding," which compresses
redundant information throughout the sample. The Huffman coding does not work with
a psychoacoustic model, but achieves additional compression via more traditional
means. Thus, you can see the entire MP3 encoding process as a two-pass system: First
you run all of the psychoacoustic models, discarding data in the process, and then you
compress what's left
to shrink the storage space required by any redundancies. This second step, the Huffman
coding, does not discard any data-it just lets you store what's left in a smaller amount of
space.
• The collection of frames is assembled into a serial bitstream, with header information
preceding each data frame. The headers contain instructional "meta-data" specific to
that frame.
Along the way, many other factors enter into the equation, often as the result of options
chosen prior to beginning the encoding. In addition, algorithms for the encoding of an
individual frame often rely on the results of an encoding for the frames that precede or follow
it. The entire process usually includes some degree of simultaneity; the preceding steps are not
necessarily run in order.
* Fraunhofer IIS uses a non-ISO extension of MPEG Layer-3 for enhanced performance
(“MPEG 2.5”)
Filter Bank
The filter bank used in MPEG Layer-3 is a hybrid filter bank which consists of a poly-phase
filter bank and a Modified Discrete Cosine Transform (MDCT). This hybrid form was chosen
for reasons of compatibility to its predecessors, Layer-1 and Layer-2.
Perceptual Model
The perceptual model mainly determines the quality of a given encoder implementation. It
uses either a separate filter bank or combines the calculation of energy values (for the masking
calculations) and the main filter bank. The output of the perceptual model consists of values for
the masking threshold or the allowed noise for each coder partition. If the quantization noise
can be kept below the making threshold, then the compression results should be
indistinguishable from the original signal.
Joint Stereo
Joint stereo coding takes advantage of the fact that both channels of a stereo channel pair
contain far the same information. These stereophonic irrelevancies and redundancies are
exploited to reduce the total bit rate. Joint stereo is used in cases where only low bit rates are
available but stereo signals are desired.
gain to result in a larger quantization step sizes until the resulting bit demand for
Huffman coding is small enough.
The great bulk of the work in the MP3 system as a whole is placed on the encoding
process. Since one typically plays files more frequently than one encodes them, this makes
sense. Decoders do not need to store or work with a model of human psychoacoustic principles,
nor do they require a bit allocation procedure. All the MP3 player has to worry about is
examining the bitstream of header and data frames for spectral components and the side
information stored alongside them, and then reconstructing this information to create an audio
signal. The player is nothing but an (often) fancy interface onto your collection of MP3 files
and playlists and your sound card, encapsulating the relatively straightforward rules of
decoding the MP3 bitstream format.
While there are measurable differences in the efficiency-and audible differences in the
quality-of various MP3 decoders, the differences are largely negligible on computer hardware
manufactured in the last few years. That's not to say that decoders just sit in the background
consuming no resources. In fact, on some machines and some operating systems you'll notice a
slight (or even pronounced) sluggishness in other operations while your player is running. This
is particularly true on operating systems that don't feature a finely grained threading model,
such as MacOS and most versions of Windows. Linux and, to an even greater extent, BeOS are
largely exempt from MP3 skipping problems, given decent hardware. And of course, if you're
listening to MP3 audio streamed over the Internet, you'll get skipping problems if you don't
have enough bandwidth to handle the bitrate/sampling frequency of the stream.
Some MP3 decoders chew up more CPU time than others, but the differences between
them in terms of efficiency are not as great as the differences between their feature sets, or
between the efficiency of various encoders. Choosing an MP3 player becomes a question of
cost, extensibility, audio quality, and appearance.
Today's music technologies have turned passive listeners into active participants that
can capture, record, transform, edit, and save their music in a variety of digital formats. An
emerging technology that can significantly reduce the size of digital music files while
maintaining their original sound quality is mp3PRO.
A coding scheme for compressing audio signals, MPEG reduces the size of audio
files using three coding schemes or layers. The third layer, commonly known as MP3, uses
audio coding and psychoacoustic compression to remove the information or sounds that can't be
perceived by the human ear. The size of the original sound recording is subsequently reduced
by a factor of 12 without sacrificing sound quality.
Music compressed with MP3 is very similar to the original. However, when you
start to reduce the bit rate — thereby reducing the file size — the music begins to sound dull. In
addition, a 3-minute, satisfactory quality MP3 song takes about 15 minutes to download using a
56K modem.
9. Conclusion
By eliminating audio information that the human ear cannot detect, modern
audio coding standards are able to compress a typical 1.4 Mbps signal by a factor of
about twelve. This is done by employing several different methodologies, including
noise allocation techniques based on psychoacoustic models.
Future goals for the field of audio compression are quite broad. Several
initiatives are focused on establishing a format for digital encryption (watermarking) to
protect copyrighted audio content. Improvements in psychoacoustic models are
expected to drive bit rates lower. Finally, entirely new avenues are being explored in an
effort to compress audio based on how it is produced rather than how it is perceived.
This last approach was integral in the development of the MPEG-4 standard.
References