Digital audio page

Index

General
information

Analogue to
digital conversion technology

Digital to
analogue conversion technology

Digital
audio formats

CD

HDCD

DVD audio

DCC

Minidisc

DAT

Digital audio
broadcasting

Networked
audio

Papers and FAQs

Speech and
music coding

Digital
audio interfaces

AES/EBU and
S/PDIF

Contents
rights management

General information

Digital audio is the most
commonly used method to represent sound inside a computer, many audio
processing device and modern audio storage devices (like CD, MD, DVD).

Digital audio technology
is a method of representing audio signal using binary numbers. An analog audio
signal is converted to digital by the use of an analog-to-digital (A/D)
converter chip by taking samples of the signal at a fixed time interval
(sampling frequency). Binary number are assigned to
these samples. This process is called sampling. This means that the audio data
is stored as a sequence of samples taken from the audio signal using constant
time intervals. A sample represents volume of the signal at the moment when it
was measured.

This digital stream of
data is then recorded onto storage media (magnetic tape, optical disk, hard
disk or computer memory) or transmission path (telecommunication network,
Internet, digital satellite, digital TV transmission). Upon playback, a
digital-to-analog (D/A) converter chip reads the binary data and reconstructs
the original analog signal. This process virtually eliminates generation loss
as every digital-to-digital copy is theoretically an exact duplicate of the
original.

In uncompressed digital
audio each sample require one or more bytes of
storage. Number of bytes required depends on number of channels (mono, stereo)
and sample format (8 or 16 bits, mu-Law, etc.). The length of this interval
determines the sampling rate. Normally used sampling rates are between 8 kHz
(telephone quality) and 48 kHz (DAT tapes).

In digital audio, bit
depth (like 16 bits, 24 bits etc.) describes the potential accuracy of a
particular piece of hardware or software that processes audio data. In general,
the more bits that are available, the more accurate the resulting output from
the data being processed. Bit depth is frequently encountered in specifications
for analog-to-digital converters (ADCs) and digital-to-analog converters (DACs), when reading about software plug-in, and when
recording audio using a professional medium such as a digital audio workstation
or a Digital Audio Tape machine. Bit depth is the number of bits you have in
which to describe something. Each additional bit in a binary number doubles the
number of possibilities. When you have a 16-bit system, there are 65,536
possible levels of audio signal. When you have a 24-bit process or piece of
24-bit hardware, there are 16,777,216 available levels of audio.

16 bits is a typical
resolution (bit depth) of many digital audio systems nowadays. The audio on CD
disk is stored at 16 bit resolution. Modern PC soundcards operate at 16 bits
resolution. In some professional applications, there is need for more bits. In
some digital professional audio systems you can encounter 20-bit or 24-but
resolutions. Some devices can process soemtimes the
data more accurately (for example at 32-bit resolution), but the information is
generally transferred and conmmunicated at maximum of
24 bit resolution. 24-bits is practically enough for any real life audio
application (pretty much on the performance limits of bers
A/D converter) and it usually the maximum resolution supported by digital audio
interfaces (AES/EBU in professional audio world, S/PDIF in consumer products).

There is universal
definition of resolution: Resolution is the ratio between the largest representable signal and the basic ambiguity in
representation, whether that ambiguity is due to random, unpredictable noise or
quantization. This amobiguity exists simply because
at low enough signal levels, you do not know whether the signal change is due
to the signal or to the noise in the system (and, by "you" we mean
"your ear" or "your test instrument" equivalently).

It should be further noted
that properly implemented digital system do not sample the music in discrete
"steps" but rather have the same unpredictable random ambiguity as conventional
analog systems.

In some instances you
might have heard of 1-bit audio systems. You might wonder how 1-bit can present
audio signals, because 1 bit can present only onle
signal level. The magic is in that those 1-bit samples are generated using a
special sampling process which produces a series of 1-bit samples that
represent the audio signal accurately after played back to playback device. The
simplest 1-bit system you can think of is delta modulation (DM), where each bit
presents should the output signal voltage increased or dicreased
a bit. This was used in some telephone systems. Sigma-delta modulation (SDM)
was developed in 1960s to overcome the limitations of delta modulation.
Sigma-delta systems quantize the delta (difference) between the current signal
and the sigma (sum) of the previous difference. SDM is also known as pulse
density modulation (PDM). The maximum quantizer range
of SDM is determined by the maximum signal amplitude and is not dependent on
signal spectrum.

For the 1-bit samples to
represent the audio signal accurately, this kind of 1-bit samples needs to be
taken at very high rate (very much higher than samples with higher resolution
are generally taken in audio applications). To convert a maximum amplitude
16-bit word, a 1-bit modulator (delta modulator) would have to perform 216
toggles per conversion period; with a sampling frequency of 44.1 kHz, this
would demand a toggle rate to approximately 2.9 GHz, an
impossibility with today's technology. For comparision
with SDM with an audio band 22,1 kHz and 64 times
oversampling, the internal sampling frequency rises to 2.8224 MHz, thus
quantization noise is spread from dc to 1.4112 MHz. However, sigma-delta
modulation adds noise-shaping benefits.

Nowadays SDM
(sigma-delta-modulation) is used in the DA converters of many CD players. The
16-bit 44.1 kHz data is requantized and oversampled
with SDM encoder and converted to analog line level signal with SDM decoder.
1-bit converter is in some cases easier and cheaper to implement and the most
important benefit is its better tolerance for small variances in component
values compared to the high demands of the conventional weighted-resistor and
R-2R ladder high-bit DACs.

The physical device that
converts analogue audio to digital audio is called ADC (Analog to Digital
Converter) and the device which converts digital audio to analogue audio is DAC
(Digital to Analog Converter).

Sampling parameters affect
quality of sound which can be reproduced from the recorded signal. The most
fundamental parameter is sampling rate which limits the highest frequency than
can be stored. It is well known (Nyquist's Sampling
Theorem) that the highest frequency that can be stored in sampled signal is at
most 1/2 of the sampling frequency. Sample encoding limits dynamic range of
recorded signal (difference between the faintest and the loudest signal that
can be recorded). In theory the maximum dynamic range of signal is number_of_bits * 6 dB . This means
that 8 bits sampling resolution gives dynamic range of 48 dB and 16 bit
resolution gives 96 dB.

1-bit
A/D and D/A Converters

A
Graphical Explanation involving Sampling

Audio
receivers tune in to binary broadcasts - Digital audio is more than
the music played on PCs, PDAs, or solid-state
Walkmans strapped to joggers' hips. Sonic-supplied Internet appliances can
transport the tunes to every room in the house.

Basics
of Digital Recording - digtal audio
introduction

Consequences of Nyquist Theorem for Acoustic Signals Stored in Digital
Format - published in proceedings from Acoustic Week in Canada 1991 -
CAA Conference, Edmonton, Alberta, Canada, October 7 - 10, 1991, from Digital Recordings

Digital
audio gets an audition part one: lossless compression - Theory's fine,
but reality's better. In this two-part series a PC, stopwatch,
oscilloscope, and spectrum analyzer are used to get answer your
audio-compression questions.

Digital
audio gets an audition. Part two: lossy
compression - Lossless compression can shrink file sizes down a fair
amount, but for serious weight loss, you need to permanently discard some
of the data. Find out how the lossy codecs work and whether you can hear the differences
between them.

Digital audio
lectures

Digital
Audio: Theory and Reality

Destination
distortion - High-resolution audio strides toward an unclear future.
Advanced audio formats' success won't necessarily come from their
high-resolution features. Advantages of the latest audio formats are
apparent for silicon, software, and technology providers; system
manufacturers; and media producers. But benefits for consumers are less
obvious. Couple that with the formats' diversity and incompatibility, and
they all may end up as niches.

FAQs (Frequently asked questions with answers) on
Digital Signal Processing

Introduction to
Digital Recording Techniques - paper was published in Canadian
Acoustical Association Journal, 1989, from Digital Recordings

Level
Practices in Digital Audio

Principles
of digital audio

Security
scheme doesn't hold water(marking) - Audio watermarking inserts media identification
data within the bit stream, using techniques that its advocates claim are
inaudible, but they might not be that reliable and unaudible
as claimed

Signal
Processing information page

S/PDIF
IEC958 (named IEC60958 at 1998)

The Video
Engineer's Guide to Digital Audio - full book on-line

White
Paper: Digital Audio Distribution Systems - paper examines the
suitability of several computer networking technologies to the task of
carrying real-time digital audio

Analogue to digital conversion technology

An analog-to-digital
converter (also known as an ADC or an A/D converter) is an electronic circuit
that measures a real-world signal (analogue audio) and converts it to a digital
representation of the signal (digital audio). A/D-converter compares the analog
input voltage to a known reference voltage and then produces a digital
representation of this analog input. The output of an ADC is a digital binary
code. By its nature, an ADC introduces a quantization error. This is simply the
information that is lost, because for a continuous analog signal there are an
infinite number of voltages but only a finite number of ADC digital codes. By
increasing the resolution of the ADC, the number of discrete steps is
increased, which reduces quantization errors.

A typical
analog-to-digital process consists of two stages: discrete-time sampling
followed by amplitude quantization. The sampling takes samples of the incoming
signal voltage at teh predetermined rate (sample
rate). Amplitude quantization convert those voltage
samples to numeric value with limited resolution (determined by the number of
bits used). Each of these two steps has requirements that must be met.
Discrete-time sampling demands that the signal must be band limited to the
appropriate Nyquist band before sampling to prevent
aliasing. The signal must also be properly dithered before quantization to
prevent the introduction of signal-correlated artifacts (usually described as
"grain and grunge"). If you do not add dithering, there is some nasty
stuff that happens to an undithered signal as it
approaches the bottom of the bit bucket, and the rapidly mouonting
severe distortion due to that. One of the big problems with quantisation
distortion is that it isn't so many dBs below signal
level, but is pretty much at a constant level regardless of the level of the programme material.

To avoid those problems,
dither is used practically always in some form. The application of the correct
amount of dither will completely eliminate all signal-correlated artifacts.
Properly-dithered digital contains no nonlinear distortion of the signal being
quantized. With any normal form of dither (technically, with any wordlength reduction to quantized PCM) you end up with an
entirely decorrelated noise floor, which mimicks the behavior of ordinary noise. Dithered noise
floor is not true Gaussian noise, indeed there are continuing discussions among
pro-audio guys as to which form of dither is best for
total inaudibility.

Dither can can applies in many ways and sometimes
you might not need any spearate ditner
noise source at all. For example when music is being played full-out there is
generally enough non-harmonic, noise-like content to do an effective job of
dithering on its own without need for a specific dither source. In a 24-bit
recording system the intrinsic analogue noise floor will almost certainly be
greater than 1 bit in amplitude, so a specific dither signal would be
unnecessary. Of course it would be needed during 16-bit truncation.

In a dithered system the the definition of resolution is is
not always very lear. With
an un-dithered quantised system, resolution is simple
- the size of the step. There can be no argument about that. When you introduce
dither, things change. You no longer have that step, and resolution turns into
a rather moot subject which is more to do with how small a signal you can
discern in the presence of noise. This is no longer simple. By choosing
appropriate bandwidths you can discern as small a signal as you like. So
resolution no longer has any easily defined meaning. It is replaced by signal
to noise ratio.

This was the conventional
architecture. Nowadays different A/D conversion architectures tend to blur the implementational division between the two steps.

A/D-conversion in audio
systems is a sampling process. All the sampling processes are limited by Nyquist limit. The Nyquist limit
is defined as half of the sampling frequency. The Nyquist
limit sets the highest frequency that the system can sample without frequency
aliasing. In a sampled data system, when the input signal of interest is
sampled at a rate slower than the Nyquist limit (fIN > 0.5fSAMPLE), the signal is effectively
"folded back" into the Nyquist band, thus
appearing to be at a lower frequency than it actually is. This unwanted signal
is indistinguishable from other signals in the desired frequency band
(fSAMPLE/2). Usually the signals are prefiltered
before they enter the A/D-converter to avoid too high frequency signal
components which can cause this kind of unwanted signals.

Typical good quality
modern digital audio systems use at least 16 bit resolution (typically 16-24
bits) and sample rate of 44.1 kHz (CD sample rate), 48 kHz (DAT sample rate) or
higher (96 kHz, 192 kHz etc.).

Sometimes you might wonder
what is the dynamic range of a digital audio system.
Hard to use accurate definition is that the total amount of uniquely representable information in the system is, indeed,
proportional to the dynamic range in amplitude multiplied by the bandwidth.

But, if you accept the
definition of dynamic range as " the ratio
between the largest possible undistorted signal and the smallest unambiguous
change in signal," then it's easy.

Actually, for
run-of-the-mill PCM data, dynamic range only depends on the number of bits. To
do the bare calculation, the dynamic range in dBs is
20 log (2^n) where n is the number of bits. But as I say, that is far from the
whole story. The generally accepted rule of thumb is 6 dB per bit. So a 10-bit
system has 60 dB, a 16-bit system 96 dB, etc. Please note that this value is
for a single sample. Through common averaging techniques, it is possible, just
like in non-digital systems, to encoded and detect
signals that are well below the noice floor. The 96
dB figure is the worst-case, non-dithered instantaneous dynamic range of a
16-bit system.

When talking about
A/D-conversion, things get more complicated. 0dB is easy - it is the biggest
sine wave you can fit into the digital space. You are driving the ADC
end-to-end. The other end is more problematic. You find it by measuring the
noise level. Unfortunately this depends on the measuring bandwidth.

Every time you halve the
measuring bandwidth, you reduce the noise by 3dB. Of course you can simply
include the entire bandwidth, but in the case of audio that isn't really fair,
because we don't hear that way. Particularly at low levels, we hear
predominantly the middle range frequencies, and the extreme lows and highs
disappear. At threshold levels even "A" weighting gives unfairly pessimistic
levels of noise.

Digital to analogue conversion technology

A digital-to-analog
converter (also known as a DAC or a D/A converter) is an electronic circuit
that converts a digital representation of a quantity into a discrete analog
value. The input to the DAC is typically a digital binary code, and this code,
along with a known reference voltage, results in a voltage or current at the
DAC output. The word "discrete" is very important to understand,
because a DAC cannot provide a continuous time output signal; rather, it
provides analog "steps." The steps can be lowpass-filtered
to obtain a continuous signal.

In D/A conversion process
the output of D/A converter is fed through a filter which will remove the
image-frequency information (signal higher than 1/2 of sampling frequency) from
the output signal. This image-frequency information can distort the output
signal. Two methods exist for removing unwanted image signals from the DAC
output to prevent alising in a following ADC. First
approach is to use a high-performance lowpass filter
(data -> DAC -> high-order lowpass filter). For
low pass filtering usually a sixth-order lowpass
filter is enough. The second methos is to use
digital-interpolation filters and a simple analogue filter (data ->
oversampling digital-interpolation filter -> DAC -> low-order lowpass filter).

1-bit
converters

A
Suggested Explanation For (Some Of The) Audible Differences Between High
Sample Rate and Conventional Sample Rate Audio Material - AES paper in
pdf format

Effects
in High Sample Rate Audio Material - document in pdf format

On
Jitter, the S/PDIF Standard, and Audio DACs
- a discussion of how the implementation of the S/PDIF standard used in
consumer digital audio can effect the fidelity of the recovered analog
signal

Bit Wars
Listening Tests - Can You Hear the Difference between 16- and 24-bit recording ?

Everything
you always wanted to know about jitter

Resolution,
Bits, SNR and Linearity - document in pdf format

Timing
Errors and Jitter - document in pdf format

Understanding the
origin and function of audio A/D converters

Why
Jitter matters in high resolution digital audio systems - white paper
from Axon in pdf format

Digital audio formats

Format comparisions

A Comparison
Between Audio Formats - DAT, LD, AC-3, radio, TV, LP, phone

CD

Compact
Disc is a stereo digital audio system based upon a 12cm single sided disc made
from a polycarbonate material with an internal reflective layer of aluminium or occasionally gold. The information is read by
sensing the presence or absense of reflected light
from a tightly focused Laser beam pointing at the pits and bumps in the
reflective surface within the disc. The digital audio signal is sampled to 16
bits per channel at a rate of 44.1KHz. The major
developers were Philips electronics of Holland
and Sony of Japan. In other words the audio format in audio CD is two-channel,
signed 16 bit two's complement integer samples, 44.1 kSamples/second, linear PCM. This is defined in an industry-standard
publication known as the "Red Book", the first in a series of
"rainbow" books that define other formats (DVD, karoke,
CD-R, etc. etc.).

According
to Red Book specifications, a standard CD is 120 mm (4.75 inches) in diameter
and 1.2 mm (0.05 inches) thick and is composed of a polycarbonate plastic
substrate (underlayer - this is the main body of the
disc), one or more thin reflective metal (usually aluminum) layers, and a
lacquer coating. CDs are divided into a lead-in area, which contains the table
of contents (TOC), a program area, which contains the audio data, and a
lead-out area, which contains no data. An audio CD can hold up to 74 minutes of
recorded sound, and up to 99 separate tracks. Data on a CD-DA (audio CD format)
is organized into sectors (the smallest possible separately addressable block)
of information. The audio information is stored in frames of 1/75 second
length. 44,100 16-bit samples per second are stored, and there are two channels
(left and right). This gives a sector size of 2,352 bytes per frame, which is
the total size of a physical block on a CD. CD data is organized into frames
(consisting of 24 bytes of user data, plus synchronization, error correction,
and control and display bits) which are intricately interleaved so that damage
to the disc will not destroy any single frame, but only small parts of many
frames. In most cases error correction (reed-salomon)
or error-concealment techniques can still solve those errors.

Some
technical data on CD system:

Playing time:          74 minutes, 33 seconds maximum (can be pushed to 78-80 minutes)

Number of channels:    2 channels

Quantization:          16-bit linear

Sampling frequency:    44.1 kHz

Data bit rate:         2.0338 Mb/sec

The actual audio signal data
from this data is

44,100 samples/second * 16
bits/sample * 2 channels = 1,411,200 bits per second.

Here is some data on CD disc
itself:

Disc diameter:
120 mm

Modulation system:     Eight-to-fourteen
Modulation (EFM)

Channel bit rate:      4.3218 Mb/sec
(data rate stored to disk)

Error correction code: Cross
Interleave Reed-Solomon Code (25% redundancy)

Laser wavelength:      780 nm
(infrared)

Lser diode power:      Typically 3-5 mW

Reading signal power:  Typical CD laser optics put out about
.3-1 mW at the objective lens

Laser focus point:     1 to 3 mm from lens
front surface

Disc material:
Any acceptable medium with a refraction index of 1.55

Pit length:
0.833 µm - 3.56 µm

Pit depth:
~0.11
µm

Pit width:
~0.5 µm

Rotation:
Counter-clockwise when viewed from readout surface

Track speed:
1.2-1.4 m/sec

Disc rotation speed:   from about 500 rpm, down to about
200 rpm

The
disc has a diameter of 120 mm with a 15 mm centre hole. The track extends
between the 46 mm and 117 mm radii. The track pitch is constant at 1.6 um,
making 22188 tracks across the radius of the disc, for a total length of 5.7
km.

The
compact disc player contains two main subsystems: the audio data processing
system and the servo/control system. The servo, control, and display system
orchestrate the mechanical operation of the player and include such items as
the spindle motor, auto-tracking, lens focus, and the user interface. The audio
data processing section covers all other player processes.

Many
audio CD players can play besides normal CDs also audio recordings done to CD-R
disks. But not all. Sometimes people ask why. The
answer is the following: All CD-R's have lower reflectivity than
"real" CD's, and many older drives made before CD-R's were widely
available don't have enough laser output (especially as they get older and the
laser wears out) or enough range in their laser AGC system to increase the
laser output enough to read CD-R's. A few older drives that had plenty of spare
laser output available have no trouble reading CD-R's, while most modern drives
deliberately have an AGC system for the laser output with enough range to cope
with a variety of media types. If your player can't play bacd
CD-R's there is not much you can do about it, except to try different brand and
colour CD-R's - the various different formulations have different amounts of
reflectivity - so some are better than others in marginal drives. Typically the
the Blue/Silver disks to work best (blue dye, silver
substrate, made by mitsubishi) and he Green/Gold ones
(Green dye, gold substrate) to cause more trouble, but different drives
sometimes prefer different colour CD-R's. In some cases it has been reported
that the speed that you write the disk at can influence how easy it is for
older drives to read the disk. The popular wisdom on this is that writing the
disk slower - even down as low as single speed - makes a disk that is more
easily read by older drives. If you have a marginal CD player you want still
use with CD-Rs, get a bunch of different brands and
keep burning with different settings until you find one that works. There are
not many regular CD players that have problems with CD-Rs.
Best CD-R media can help reduce those rare problems.

A fundamental
Introduction to the Compact Disk Player

An
Exploded CD player as Teaching Aid

CD player
features - a summary of experiences and opinions concerning CD specs
and the utility of specific player features

CD-player mods
- CD player modification ideas for propable
better sound quality

CDplayer's Project - information to make and
upgrade a CDPlayer at home

CD
Subcode Formats

Compact
Disc Formats - explanation of different types of CD disks

Compact Disc: The
Inside Story - articles on the technical details of the Compact Disc
audio recording system originally published as a series in the magazine
MAC Audio News

Differences between CD-R/CD-RW
discs and standard CD

Digital Domain - site is designed to
help audio engineers and musicians make better Compact Discs and CD ROMs

EE498
Project Reports - exploded CD player, 1-bit D/A, CD testing, karioke device

How CDs Work

Introduction to
CD player - how CD player works

CD Player
Laser Diodes - This document give you introduction to lasers used in
CD player.

Red
Book - The Red Book is the 1980 document that provides the
specifications for the standard compact disc (CD) developed by Sony and
Philips. According to legend, the document was in a binder with red
covers, originating the tradition for subsequent adaptations of CD
specifications to be referred to as variously colored books.

Subcode on the CD - check also Q-Subcode Decoding and Display project

The Compact Disc -
good introduction to CD technology by Philips

The
Compact Disk - how does it work

Transferring LPs
to CDR: Some Advice

Understanding the CD -
How CDs Work

CD copy protection

The
legal and illegal copying of CDs has increased during the last few years due to
the wide availability of low cost CD-recorders and media. This has resulted in
a fall in the demand for legitimate pressed CD albums in some countries, which
has made music companies to react to this situation. The new discs now making
their way into record stores in the United States
and Europe contain countermeasures that
prevent playback on computers and, in some unintended cases, normal CD players
as well. CD copy protection is designed to work it should play in your home CD
player but not in your CD ROM drive. The reason being that CD
ROM drives enable you to easily copy the digital audio data and burn copies of
it.

The
controversial new anti-copying technology introduces minute errors to the CDs
or changes the location of data on the discs to prevent them from being played
back on computers. In theory, most consumer CD players (but not all) can
correct the errors and decipher the structure, unlike the more finicky computer
CD drives (which uusally try to read more details
than normal audio CD players). All of the big major record labels are
experimenting with ways to block consumers from "ripping" or
transforming their CD songs into MP3 files and distributing them widely online
though on-line file sharing services.

This
"CD copy protection" has been a well discussed topic because some
record companies have started to use it on some music CDs. Copyright protected cd's do not allow you to replicate them in a cd burner nor do they allow you to rip the audio tracks
"digitally" (although can still be done through analog). Copyright
protections systems currently in use have downsides: not all CD players
(especially computer CD-ROMs and some car plyers)
will play those CDs.

The
current copy protection techniques generally either corrupt the CD table of
contents or actual audio data. The protection techniques that modify table of
contents, modif it in in
such a way that readers that know about data CDROMs
get a tummy ache and see only some trivial data files, and not the audio
tracks. This means that the protected discs can't be played on CD players that
use CDROM drives for transports, of which there are quite a few. One way to
make a CD act like is is to add a CD-Extra file track
to the end of the CD, but making that CD-Extra track somehow broken (computer
tried to read this track, but fails with error which causes the reading of
whole CD to fail in many systems). Another method does corrupt the data. The
plan is to put some deliberately-corrupted blocks, corrupted enough that the
ECC will fail, into the audio data. It's transparent to analog output because
the corrupted samples are in places where the interpolation done by the player
when it can't do the digital ECC will end up with the same answer (or very
close to it) as was in the original samples prior to corruption. Most PC
CD-ROMs can detect the errors caused by data corrupptio,
but most CD-ROM drive controllers lack the ability to pass this information of
where error back up to the PC or do the error-concealment algorithm in
firmware, prior to passing the data to the PC. A lot of the better CD-ROM
drives do the error concealment on "ripped" data, so they aren't
bothered at all by these "protected" discs. Cheaper CD-ROM drives may
not do the error concealment.

Anyway
this copy protection might make the copying of audio CD harder, but it does not
stop you from doing that if you really want to do this. If you can play back
the audio CD through your HIFI system, then there is always a way to copy it
some way (through digital or analogue means). When you play back a CD on your
normal CD player, you wll always get the correct
audio signal from the CD player analogue output and CD player digital audio
output (no matter if that CD is protected or not). No CD copy protection can
stop that (or it will stop normal playback with normal CD players as well).

There
are industry views that this kind copy protection techniques
are actually a violation of the "red book" standards, to which
everyone who presses music CDs has contractually agreed to comply. According
some megazine article Philips has been putting these
companies on notice that they may not advertise or label such products as
"Compact Discs", and cannot use the CD logo or claim CD
compatibility, as these discs deliberately violate the Red Book standard and
are thus a technical violation of the license agreements for the use of the
logos, etc. Sony is also a co-licenser of the CD system and they are very much
in favor of copy-protection. So that leaves us in a sort of twilight state of
not knowing what will happen next.

At
the very least, consumers have the right to insist that any such products be
accurately identified (warning stickers, etc.) so that we can make an informed
decision about buying them (or not!). The only power we consumers have is to
bring back the defective CD that does not play properlu
and ask for a refund or non-copyprotected
replacement.

It
seems that music companies are now trying to latch the door on the barn after
the horse escaped. What happens with many of the copy-protected discs is that
many users will have problems in playing them properly (lots of angry customers
and returned CDs to shops) and that someone breaks the copy protection and
distributes the tracks through a file-sharing service ?
Once a track enters the domain of a file-sharing service, there's no stopping
its distribution. People copy those music files at higher priority, simply
because they were copy-protected (also those people who would have bought a CD
but it did not work for them). This means that applying copy protection which
causes problems to use punishes the users which are willing to buy the original
CD. If the original CDs do not work reliably anymore, those users might be
tempted to go to the route of copying the music instead (in this way they get
their own CD copy which works for sure on many different players including
their home CD, DVD and PC).

Protected
audio CD's can be backed up using a variety of methods but usually an easy 1:1
backup is not possible. This is exactly what they want, making it harder to
make a backup so that people will buy the original. It is hard to say if audio
CD Protections have really helped in any way (the history of game CD
protections has not been a good one). In some cases CD protection even can work
against you as it limits the use of the CD. You can't play protected CDs on
CD-ROM players and even some normal CD players can not handle the protection.
So every CD-protection scheme faces a crucial obstacle: making CDs unrippable onto CD-ROMs also makes them unplayable on some
CD players -- a feature guaranteed to anger customers.

The
CD copy protection reduces the usability value of CDs. The user's
don't get anymore what they used to get when they bought it. They can't play it
on certain devices and maybe can't make legal copies to other media devices
they want to use (laptop hard disk, MD player etc.). This reduced usability
combined with the same price can easily reduce the sales of the CDs (reduced
usability with same price generally reduces sales in free market). For example
many users who buy a new protected CDs are quite
pissed off when they find out that their high quality car CD player refuses to
play that CD. And they can't either listen it at work
with their computer CD-ROM.

Combined
this with the fact that those copy protections do not stop pirate CD makers on
making pirate copies of copy protected CDs, those copy protections do not stop
CD piratism. They can even lead some people to prefer
pirated CDs, because non-copy protected pirate CDs work better for them than
the real ones. This applies to both bought pirate CDs and ones burned to CD-R
from MP3 files downloaded from net. So those user who have problems with CD
protection can easily start to think "Why should I pay high price of CD to
a large media company which I don't like and which just supplied expensive
flawed products which I can't use ?" and look for altenative
solutions.

Other
concerns are that CD copy protections can break the chain necessary to preserve
creative works. It will make them readable for a limited period of time and not
be able to be moved ahead as media deteriorates or technologies change.

From
CD end user point of view, those CD copy protection schemes are a very bad idea
which should not be used.

CD
Audio Copy Protection - Copy protection systems are now available to
reduce the threat of illegal copying. What they are and how they work ?

CD creator burns
copy-protection efforts - As major record labels unveil a new breed
of CDs designed to prevent Napster-style piracy, Dutch consumer
electronics maker Philips, the co-creator of the CD, is refusing to play
along.

CD
Protections - The last few years game developers more and more are
protecting their games with commercial CD Protections. Now the music
industry has hopped the bandwagon and now is selling protected audio
CD's. These CD's can be backed up using a variety of methods but usually
an easy 1:1 backup is not possible. This page contains a list of all
known CD Protections which are currently on the market, how they can be
detected and how to by-pass many of them.

Corrupt
audio discs, aka "Copy-Protected CDs" - Summary of Sony's
new set of non-"Red Book" audio CD releases

Corrupt audio discs, aka
"Copy-Protected CDs" - Here is a list of known the bad CDs
available in UK.

Is
Universal Music's Copy Protection a Joke? - At end of 2001 Universal
Music introduced copy protection for its line of audio CDs using Midbar Tech's Cactus Data Shield (CDS).

Kopierschutz mit Filzstift knacken -
Copy protection with felt-tip pen crack is discussed here. The text is in
german, try automatic
translation to English.

Labels
loosening up on CD copy locks - Fearful of consumer backlash, major
record labels in the United States have slowed controversial plans for
making CDs more difficult to copy.

Lawmaker: Is
CD copy-protection illegal? - Record companies' efforts to protect
CDs against digital copying are beginning to draw scrutiny from lawmakers
concerned that the plans might violate the law in USA.

Marker pens,
sticky tape crack music CD protection - Music disc copyright
protection schemes such a Cactus Data Shield 100/200 and KeyAudio can be circumvented using tools as basic as
marker pens and electrical tape.

Philips moves to
put 'poison' label on protected audio CDs - Netherlands
giant Philips Electronics has lobbed a grenade into the audio copy
protection arena by insisting that that CDs including anti-copying
technology should bear what is effectively a plague warning. They should
in Philips' view clearly inform users that they are copy-protected, and
they shouldn't use the "Compact Disc" logo because they are
not, in Philips' considered view, proper compact discs at all.

Traitors In Our Midst
- Music lovers have begun to fear that record companies purposely corrupt
the data on audio CDs in an effort to restrict their use as a source for
copies or MP3 files. Unfortunately, those fears are coming true with
Cactus Data Shield digital-music-use restriction technology.

Universal
copy-protected CD shuns players - On the eve of Universal's kickoff
of its long-awaited Pressplay online music
subscription service, the record label is distributing its first
copy-protected CDs in the United
States, adding a few new twists to the
controversial idea.

HDCD

HDCD®
(High Definition Compatible Digital®) is a patented encode/decodeprocess for delivering on Compact Discs and DVD-audio
better than normal CD audio quality. HDCD encoded CDs are encodedso
that they fit 20 bits of information onto the CD while remaining completely
compatible with the existing CD format.

HDCD
Home Page

DVD audio

DVD
audio is a multichannel(PCM) audio format which stores
many hours of high quality (better than CD) audio to a DVD disk which can be
played on "DVD-Audio" capable DVD players. This audio format makes it
possible to store more than 74 minutes of sound in two channels using the
highest sampling frequency 192 kHz with 24-bit quantization, or six channels at
a 92-kHz sampling frequency with 24-bit quantization. That's better than twice
the performance of the CD audio specifications. Thus, dynamic range is 144 dB
and frequency response is up to 96,000 Hz. The maximum transfer rate is 9.6 Mbits/second. The DVD-Audio format was originally intended
to be finalized in spring 1996, but company infighting stretched that time
line.

Even
though though version 1.0 of DVD-Audio specifications
(the PCM format descibed above) was published on the
end of 1998, there is still some fuel on the format fight. Sony and Philips are
also behind a competing system, Super Audio CD(SACD),
using a one bit sampling 2- and multichannel format called Direct Stream
Digital(DSD). A basics SACD disk will not play on normal CD player (you will
need to have a SACD player to play it). It is possible to make so called hybrid
CD/SACD, which will play back on most CD players (but not all). This hybrid disck has two data layers: one for normal CD data (that
normal CD player reads) and another for SACD data (for SACD players). The SACD
data in the disk is 1-bit bit-stream sampled data that is sampled at 2822.4 kHz
sample rate. SACD support 1 - 5.1 audio channels. The maximum sound playback
length of SACD is 109 minutes when stereo sound is stored to it. SACD has it's own copy protection mechanism that is built into the
standard from the beginning and is supported by all SACD players.

A
DVD-Audio discs can also be played on almost any DVD-Video player if it in the
Video-section includes a track that follows the DVD-Video audio specs (some
DVD-Audio disks have this as an option for compatibility). An optional CD layer
can give DVD-Audio software compatibility with some players for CD and
DVD-Video.

The
record industry is only interested in SACD/DVD-A because of the built-in
copy-protection features. Philips and Sony are interested in SACD because of
the royalties they'll receive for many years.

Digital Versatile Disc (DVD)
- new laserdisk standard for audio and video

DVD-Audio spec
unveiled

Guideline of
Transmission and Control for DVD-Video/Audio through IEEE1394 Bus

DCC

Digital Compact Cassette (DCC) is dying digital tape
format from Philips. This system was able to storage quite high quality
(comparable to MD) MPEG compressed digital audio to a compact cassette format
cassette (basically same mechanics, just better tape in it). DCC players were
also capable of playing back normal analogue compact cassette (C cassette)
tapes. Nowadays you can't buy new DCC devices. This technology lost the battle
against MD and CD-R/CD-RW technolologies.

DCC FAQ

DCC-L Mailing List
Homepage - information about DCC, mailing list subscription
information, and archives from the list, includes DCC FAQ

Minidisc

MiniDisc (MD) is a small audio disc format. The disc used in the
system is 7 cm (2.75") x 6.75 cm (2 21/32") x 0.5 cm (3/16"),
the disc inside is 64 mm in diameter. One MiniDisc give around 160MB storage for 74 minutes in audio mode and 140MB
in Data Mode. MiniDiscs are available as ready made
records to buy and as re-recordable discs for making your own recording. The
audio data in mini disc is compressed using ATRAC algorithm.

MiniDisc is suitable format for small portable audio listening
devices and also for making quite high quality audio recordings.

ATRAC: Adaptive Transform
Acoustic Coding for MiniDisc - information
on audio coding in use in MiniDisc system

MiniDisc
Community Pages - MiniDisc information
pages which are mirrored all over the world, also mirrored in Finland

MiniDisc DIGITAL Recording FAQ

Minidisc FAQ: MDLP
(``Long-Play'') Mode Topics - Minidisc Long Play is a new encoding
method for audio on Minidisc that offers two modes: one gives 160 minutes
stereo ("LP2"), the second gives 320 minutes stereo
("LP4").

Minidisc Frequently
Asked Questions

MiniDisc
Manager - software allows computer control of Sony MiniDisc recorders that have a compatible slink
control port

The MiniDisc
Appreciation page - information about mini disk and how to use it in
different applications

US MiniDisc
page

DAT

Since
Sony introduced DAT (digital-audio tape) in 1987, it has been the preferred
format option of many journalists, musicians, and live-concert and
lecture-audience members, who want to capture a sonic snapshot of the events
they attend.

Record-label-piracy
concerns and consequent copy-protection and royalty complications, though, have
curtailed DAT's acceptance in the
consumer-electronics world.

Nowadays
recording studios are moving to hard-drive-based setups, so the DAT's future is on the wane. DAT gear seems to be rapidly
disappearing from manufacturer warehouses and store shelves.

DAT-heads - home page for
the DAT-heads mailing list

DAT Tapes page
- DAT links

DAT World

How to
maintain your DAT

Live
taping to DAT

Sony TCD D-7 and TCD D-8 resource page

Sound
savings: Portable audio recorder takes on tape, part 1 - DAT gear is
rapidly disappearing from manufacturer warehouses and store shelves. What
will replace it? Core Sound, along with a symphony of worldwide
development partners, delivers a harmonic proposal.

Tascam decs FAQ

Other digital audio formats

PC-Based
Multitrack Recording Frequently Asked Questions
(FAQ)

Unofficial Tascam
DA-88 web page

Digital audio broadcasting

DMX, Digital
Music Express - digital music channels from cable

UK Digital Radio Forum

Networked audio

Networking
for Audio

Routing
Audio Via Ethernet White Papers

Papers and FAQs

AAC
audio analysis - AAC has substantially fewer artifacts and artifact
magnitudes than its MP3 codec predecessor.

Audio
Compression - very good introduction to audio compression part of Simon Fraser University CMPT
365 Course notes

Digital Audio -
Material for the lectures

Digital
Audio Transmission - Why Jitter is Important

Everythign you always wanted to know about jitter but
were afraid to ask - jitter is often misunderstood among recording
engineers and audiophiles

The
Secrets of Dither - or How to Keep Your Digital Audio Sounding Pure
from First Recording to Final Master

Speech and music coding

Information on coding

AC-3
specification - in pdf format

Audio
Compression Introduction

Comp.speech Frequently Asked Questions web page

GSM 06.10 digital
speech compression

Human
auditory masking and perception based coding - introduction to this
topic

Philips MPEG audio compression
technology - documents, technology, software, components, links

Speech
Coding - introduction to speech codecs

Software

Hyperreal
Tool Shed - audio tool programs for computers

Source
code for audio codecs in C - ADPCM, G.72x,
LPC, GSM, CELP

Digital audio interfaces

AES/EBU and S/PDIF

At
the consumer level there are basically one digital
S/PDIF format which is transported using two different interfaces. One commonly
used interface option is electrical coaxial interface, using an RCA-type jack
and plug (or sometimes a BNC bayonet connector) and a shielded, 75 ohm
impedance cable. Other option or consumer S/PDIF interface is Toslink optical, which transmits the digital data in the
form of light down a plastic fiber-optic cable (1 mm diameter fiber center).
Nowadays the official name for S/PDIF is standard IEC958 "Digital audio
interface". Professional audio equipments use AES/EBU format defined by
standard IEC-958, which uses a 3 pin, balanced XLR-type connector and 3
conductor, 110 ohm impedance cable (allowed range 88-132 ohms) or multimode
fiber optics (AT&T or ST-Glass).

Both
AES/EBU and S/PDIF interface support a wide varity of
sample rates. Here is table of sample rates and needed bandwidth.

Sampling Rate  Bandwidth

32.0 kHz       4.096 MHz

38.0 kHz       4.864 MHz

44.1 kHz       5.645 MHz

48.0 kHz       6.144 MHz

88.2 kHz      11.289 MHz

96.0 kHz      12.228 MHz

192.0 kHz      24.576 MHz

The
consumer devices with S/PDIF interface have generally supported sample rates
only up to 48 kHz. But the introduction of higher resolution
audio formats have added the sample rates up to 192 kHz.

The
maximum sample resolution supported is 24 bits. Many consumer S/PDIF
implementations are though limited to only 16 bits of resoltion.

The
cables uses in the digital audio connections vary between digital interface selected. The cable that connects to the coaxial
interface is designed to be 75 ohm ciaxial cable. It
is the same cable type as used in video industry to transfer video signal. For
example the video industry standard RG-59 coaxial cable is very ideal to carry
coaxial S/PDIF signal for long distances (the maximum supported distances). For
shorter runs practically any 75 ohm coaxial cable will do nicely. For very short
runs (up to 2 meters) the needs of coaxial cable are not so critical, and a
normal "RCA audio cable" usually works well. For anything more than
few meters run, use with 75 ohms coaxial cable and things work for sure.

Many
new, high-quality Mini Disc, pro-audio, DAT (Digital Audio Tape), CD, DVD, and
laser disc players, as well as digital amplifiers, DSS satellite receivers, and
computer sound cards, are manufactured with digital optical output connectors.
These connectors attach to optical cables, which are constructed with a PVC
jacket and a plastic core. The cables transfer information accurately over
short distances via digital light signals. Digital optical cable with
plastic-core construction is less expensive than fiber optic cable with a glass
core, but it still provides the benefits of optical transmission over short
distances. The two types of connectors associated with digital optical
transmission are TOSLINK®, a Toshiba® trademark, and the 3.5-mm Mini
Plug connector.

Sometimes
there is questions on hifi forums raised should I use
coaxial or (Toslink) optical interconnection
? They're both digital, so as long as those conversion processes are
reliable, nothing will be lost. When properly implemented their performance is
equal. Sometimes people recommend coaxial connection over optical because of
less processing on the way. But the processing between optical and electrical
signal presentations does not change the data contents. This means that there
can't be any difference in frequency response, amplitude distortion or noise.
The properties of the signal are encoded in the digital data. That data is unchaged by the conversion process. If errors happen (baddly operating devices), the system will notice those
very quicly (S/PDIF has parity bits to check data
integrity) and mute the audio output.

One
might also see arguments about optical connections introducing jitter. There
could be a difference in jitter, but comprehensive measurements fail to show
any significant differences. Jitter is way overplayed as an audio issue. As
long as you are in the digital domain, jitter is utterly irrelevant. It is only
at the moment of conversion to/from the analog domain that jitter is relevant.
If you have a situation where the jitter is demonstrably worse with an optical
connection, the fault lies not with the optical connection; it's in the D/A
converter itself, and is the result of poor clock design in the D/A converter.
In some cases, there are some bad implementations of optical. Combine that with
a BAD implementation of a D/A converter and you have a
system that does not work well. But, take solice in
the knowledge that as bad as some people can put together a bad implementation
of an optical output, there are others who can put together an
equally incompetent implementations of coaxial output as well. Some such
incometently equipment can be quite sensitive to the
interconnecting cable. The quality of the sending and receiving units in your
components will have a far greater impact than the cable type you choose.

The
only benefit I can think of from an optical (Toslink)
connection over a coax is the galvanic isolation between the two equipments,
which can avoid ground loops. This could be relevant in some audio systems
where ground loops are a problem. The downside is that Toslink
optical connection has shorter distance limits than coax and optical cable
usually costs more than coaxial cable. Optical cable is more fragile, more
expensive, and the connectors are typically flimsier.

The
general advice is use whatever (optical or electrical) you have easily
available and be happy. In the digital domain, you can even convert between
optical and electrical all you want without any loss (potentially, if your
hardware does everything right).

Optical
could be technically better for short runs (less than 10 meters) because it
resists interference and ground loops. Optical cables are usually a lot more
fragile than coax cables and usually cost more, so optical cable is not best
for all. Coax for long runs because Optical is rated only out to 10 meters or
so. There should be no difference between optical and coaxial transmission of
digital data. Within their recommended lengths, they sound the same. If there
are problems, the most likely symptoms are interrupted sound, clicks, pops, or
no sound at all.

The
main problem with Toslink is not sound quality. There
is no "quality" difference between Toslink
(optical) and coaxial (metalic). Ones are ones and
zeroes are zeroes in the digital realm. The main issues with Toslink are durability, maximum length, and installability (Toslink cables
aren't exactly supple). On the other hand Toslink
optical connection is a great solution for ground loops and EMI pickup or
creation.

In
coaxial interconnection sometimes should I use a special "digital
cable" or will a normal "RCA audio cable" work
? The right thing to use in any length is 75 ohm coax with appropriate connectors, such is used for RF and composite video. A
"normal RCA cable" does not usually have the right impedance, because
those cables can have their impeance anywhere in
45-75 ohms range and the signal attenuation usually much higher than proper
coaxial cable. In short lengths like 1 meter, just about anything that is
conductive and look like a cable will work about as well as anything else. A
standard 1 meter audio interconnect can do the job just well.

On
professiona AES/EBU interface the standard cabling
solution is to use 110 ohm shielded twisted pair wire terminated to three pin
XLR connectors as the standard defined. Most digital audio cables utilize foam
polyethylene to minimize the cable's size. The advent of digital microphones
requires AES/EBU cable designs with added flexibility, Although AES/EBU
specifications require shielding on each channel of data, datagrade
UTP .Category 5. can easily meetthe
common mode balance requirements (-30 dB) without being shielded. You might ask
can I use normal analogue audio cable (shielded twisted pair) for AES/EBU connection ? The answer is yes, but with limitation. The
actual length is determined by the error correction and jitter tolerance of the
receiving device. The impedance of most analog cables ranges from 40 ohms to 70
ohms. This large mismatch from the nominal 110 ohms results in signal reflection
and jitter causing bit errors at the receiver. Also, the high capacitance of
analog cables greatly increases the rise time of the digital square wave.

Another option to carry AES/EBU signal through 75 ohm
coaxial cable wiring. Generally this is done using baluns. The
transmission of digital audio over 75 ohm coax requires the use of baluns unless the device contains unbalanced coax AES
inputs or outputs. In case your equipment has only balances inputs, you can use
baluns to convert the unbalanced coax signal and a
110 ohm balanced transmission from one format to another. Much greater
transmission distances are obtainable over coax as compared to twisted pair.
The same coax used for digital video is ideal for digital audio. The coax used
should have a pure copper center conductor (no copper covered steel or
aluminum) and have good braid coverage (90% or more).

AES-3
distribution: Coax or twisted pair?

AES/EBU and S/PDIF
digital audio interfaces - zipped ascii
text file and GIF pictures

AES and S/PDIF
Recommended Transformers

Digital Format
Converters MSB, Midiman & Fostex Review - converters for S/PDIF conversion
between Toslink optical and coaxial electrical

Digital Studio
Cable Guide

Expert
opinion: which SPDIF, opical or digital?

FAQ for AES/EBU and
S/PDIF interfaces - posts from the USENET newsgroup rec.audio.pro collected to one zip file

Overview of Digital
Audio Interface Data Structures - This document is a short summary of
the most important parts of data format information from IEC958 and new
AES3-199x and TV84 documents.

S/PDIF
Interface - a description of data format used in CD player digital
output and some exaple circuits

SPDIF
overview

The Jitters
and Its Cure

MADI

MADI
(Multichannel Audio Digital Interface) is a professional multichannel version
of AES/EBUstandard for transmitting up to 56 channels
of digital audio data over a single coaxial cable terminated with BNC
connectors. MADI uses a second cable for word clock, with a fixed data rate of
100Mbps used on large, open-reel digital multitracks.
Optical MADI implementations are available.

MADI
(Multichannel Audio Digital Interface), also known as AES-10 standard, allows
interconnection of two devices to transmit up to 56 channels of 24-bit digital
audio with a single coaxial cable or via optical link. It is a standard
interface to digital multitrack machines and mixing
consoles like the Studer D941 On-Air Mixing console, Studer 950 Digital mixing system, Neve
Capricorn or Sony PCM3348HR Digital Tape Recorder. Some manufacturer specific
MADI extensions allow up to 64 channels in each direction and can operate at 96
kHz (offering up to 32 channels) or even up to 192 kHz (with 16 channels on one
single connection each way).

Other

Audio Distribution
and Control using the IEEE 1394 Serial Bus - this paper describes a
system for transmitting hundreds of channels of high quality digital
audio, control & monitoring protocols, and digital video over an IEEE
1394 serial bus

I2S enhanced -
preliminary specification

New Olympic
Stadium Audio System Runs Via Ethernet

SCSI Musical Data
Interchange (SMDI)

TDA1543
DUAL 16-BIT DAC Data Sheet - includes description of I2S interface

The
D7 7-pin connector

The Kurzweil
KDS 8+2 Channel Digital Audio Interface

Audio compression

The issue of using
compression is the matter of reduction of the amount of data. A direct measure
of the total amount of data or information in audio is its bitrate.
By using the compression the audio signal can be transported with lower bit
rate or stored to a smaller file size.

A typical audio file where
we start from is CD quality audio. Audio Compact Disks store the audio data in
files on the disk. The audio data is in a PCM (Pulse Code Modulation) format.
Each minute of recording time consumes about 9 MB (Megabytes) of file storage
space. A three minute song would occupy about 27 MB of file storage space, and
a 5 minute song would occupy about 45 MB of file storage space. A 650 Megabyte
Compact disc can contain up to 74 minutes of PCM audio.

The sheer size of PCM
audio data files made them unpopular as a storage medium for audio on a
computer or the fole format when the audio is
transported through slow modem connection. Digital compression came along and
changed all that. Digital compression offers possibilities for a huge reduction
in the amount of file space required to store audio data files. This reduction
in file size also reduced the time required to transmit them electronically
dramatically.

Audio compression can be
one of two categories, lossless or lossy:

Lossless: An intermediate representation of the
audio data is created where, in most cases, redundant information is
removed. The data rate, in the case of CD audio, would be less than
1,411,200, but because redundant information was removed, and we can know
how to put it back exactly, there is no loss of data when we decode it at
the other end.

Lossy: Data which is not deemed to
be audibly significant is removed. Then, redundant data is removed. The
resulting data rate is also less than 1,411,200 bits/sec, but, on
decoding, while the redundant data is restored, the audibly insiginificant data is not. Thus, the decoded data,
while it is the same data rate, is different than the original.

Lossless digital
compression is commonly used to reduce the size of computer files for
electronic transmission. In order for the files to be useable on a computer the
files that are extracted from a compressed data file must be identical to the
original file (before it was compressed). Lossless compression is great because
it makes perfect copies but it doesn't yield very high compression ratios. That
means it doesn't save huge amounts of disk storage space. ZIP, ARC, TAR, and
SIT are some of the acronyms or formats of Lossless Compression commonly used
on computers. Lossless compression sounds wonderful,
however it does not yield an extremely high compression ratio. Typically,
lossless routines give a ratio no higher than 2:1.

Lossy compression algorithms offers much higher compression ratios
than lossless algorithms but in order to achieve this they need to discard some
of the original data. Lossy Audio Compression can
yield a variety of different rates of compression based on the ability of your
hardware and software to encode and decode music audio data. Lossy compression is only suitable for use on audio or
graphical data. The audio or graphics are reproduced but at a lower overall
quality than they had before they were compressed. In some cases the difference
is difficult to perceive. The compression ratio can usually be adjusted so the
quality level can vary widely. Audio that is compressed at a 20:1 or 10:1 ratio
will certainly sound inferior to audio that was compressed at a 2:1 ratio.
There are various compression schemes to achieve different results; for
example, many codecs remove portions of the audio
signal that human hearing is less sensitive to. To some peoples ears the
resulting audio has distinguishing artifacts or characteristics that change the
listening experience. The final "quality" level is largely a matter
of personal perception. Each different encoder/decoder (CODEC) has different
strengths and weaknesses. Typically, encoding takes a long period of time and
large amounts of processing power. decoding can
usually be made real time or faster. MPEG, MP3, AAC, RA, WMF, JPEG, QT, and DivX are some of the acronyms or formats of Lossy compression commonly used for audio and video. The
most commonly used and most talked about audio compression in typical computer
and consumer audio enviroment is MP3.

Brief descriptions of some
different audio compression formats:

ADPCM: Adaptive Delta Pulse Code Modulation (ADPCM)
is popular since is returns a high quality signal with very little
processing power required for fast decoding. ADPCM comes in a few
different versions. The Microsoft algorithm found in Windows ADPCM codec
returns a compression scale of 4:1 (one fourth the original file size).

MPEG: The Moving Pictures Experts Group (MPEG) put a
considerable amount of research into ways of compressing video and audio
to make movies fit onto compact discs. It hs
released several versions of MPEG audio coding standards. Software
routines for MPEG audio decoding are fast to decode with relatively low
quality loss, but generally speaking it takes a LONG time to encode. MPEG
audio compression loses quality quickly at compression levels higher than
a 6:1 compression ratio. MPEG Audio compression continues to degrade until
it becomes absolutely dismal at 24:1 compression. An extension to MPEG-1 and
MPEG-2 audio called 'Layer 3' (commonly known by its file extension, MP3)
has been developed allowing greater compression levels and more BIT DEPTH
options. Since MPEG audio uses a perceptual model, the exact perceived
amount of lost quality varies for each individual listener. MPEG-4 has
been ratified as a standard, and new codecs will
become available in the not-too-distant future.

The compression ratio for
MPEG compression can be adjusted across a wide range of quality levels, however
sound quality drops as the compression is increased. Trying to maintain the
highest quality (without extremely noticeable signal loss) yields the following
compression ratios for different MPEG coders:

Layer 1 = 1:4 (384 kbps for stereo)

Layer 2 = 1:6 or 1:8 (256 kbps or 192 kbps for a stereo signal)

Layer 3 = 1:10 or 1:12 (128 kbps or 112 kbps for a stereo signal)

With the increase in
inexpensive mass storage and the increases in connection speeds we are rapidly
approaching a point in time when compression of audio may not be as important
of an issue as it was in the past.

General audio compression links

Compression on a
Windows PC

MP3

The
MP3 format is a compression system for music. MP3 (MPEG-1, Layer 3): a compression
standard that creates relatively small digital audio files with high-fidelity
sound. MP3 is simply a file format that compresses a song into a smaller size
so it is easier to move around on the Internet and store. The MP3 format for
digital music has had, and will continue to have, a huge impact on how people
collect, listen to and distribute music. The MP3 movement is one of the most
amazing phenomena that the music industry has ever seen: Web's ability to
advertise and distribute MP3 files is huge. Very many people are now
downloading, listening to and saving MP3 files onto CDs!

The
MP3 format helps reduce the number of bytes in a song without hurting the
quality of the song's sound. The goal of the MP3 format is to compress a
CD-quality song by a factor of 10 to 14 without noticably
affecting the CD-quality sound. With MP3, a 32-megabyte (MB) three minute song
on a CD compresses down to about 3 MB. This allows you to download music at
useable speeds from the network and store hundreds of songs on your computer's
hard disk without taking up that much space.

MP3
is lossy compression algorithm. When the music is
compressed to MP3 format, some part of the original sound is lost. MP3 encoders
create smaller files by getting rid of unnecessary audio information. When you
convert .wav files to MP3, an encoder filters out data representing sounds
outside the average person's hearing range. Because some of the original audio
data is lost, this technique is called lossy
compression. MP3 format uses characteristics of the human ear to design the
compression algorithm: there are certain sounds that the human ear hears much
better than others and certain sounds that the human ear cannot hear
(especially when played with other sounds). Using facts like these, certain
parts of a song can be eliminated without significantly hurting the quality of
the song for the listener. When you are done creating an MP3 file, what you
have is a "near CD quality" song. The MP3 version of the song does
not sound exactly the same as the original CD song because some of it has been
removed, but it's very close.

Lossy compression has its downside: The more data the encoder
throws out to shrink the file size (a setting you can control), the worse the
finished product sounds when you play it back. The result is that not all MP3
files sound alike. Also the encoding software can have effect on the sound
quality, because the human hearing model used in then can be slightly
different, so different encoders can throw out sligtly
different parts of the sound, thus generating slightly different results. The
differences in quality are usually largest at lowest bit rates (where most
compression is used).

MP3
industry claims that music encoded at 128 kbps will produce "CD
quality" MP3 files -- and at that quality, a one-minute audio file fills
about 1MB of disk space. But regular MP3 users disagree, saying that they hear
significant distortion unless the file has been coded at 160 kbps, which
produces files that take up slightly more space but sound much better. There
are some people that say that they need much higher conding
rate. In some special application lower than 128 kbps rates can be used. MP3s
coded at 96 kbps end up as far smaller files but have considerable noise.

Once
you have an encoded MP3 file, you need a decoder or player -- which can be
either software or hardware -- to convert encoded MP3 data into something you
can listen to. You can download most software MP3 players for free from the
Web. Maybe the best known MP3 playback program is Winamp.

Many
people think that the MP3 standard is free and open, and that the ISO reference
source code is also. But it is not entirely free. MPEG development is almost
open. When an MPEG format is approved by ISO, it contains several parts. It
defines the bitstream syntax that must be used to be
compliant, operations done by the decoder, and it provides an informative only
source code for encoding and decoding, which is freely available. But this
source code is purely indicative, as MPEG never defines any encoding rule. The Fraunhofer Institute has been the main developer of MPEG
audio Layer-3, and the MP3 standard that has been approved is mainly based on
their work, which Fraunhofer has protected by several
patents (a portfolio of 18 patents related to MP3). This patent portfolio owned
by Fraunhofer and Thomson Multimedia is very
extensive, and cover various aspects of MP3 encoding. Practically you can not
use any MP3 implementation without using some parts of those patents. other companies could have some patents also covering parts
of MP3. MP3 standard is established since 1992, and no one except Fraunhofer/Thomson Multimedia asked for fees.

How MP3 Files Work

How
MP3 works

How MP3 Works: Inside the
Codec - So what's the trick? How does the MP3 format accomplish its
radical feats of compression and decompression, while still managing to
maintain an acceptable level of fidelity to the original source material?
The process may seem like magic, but it isn't.

Patents and MP3 - Fraunhofer has protected MP3 technolgies
by several patents.

Philips MPEG audio compression
technology - introductory documents to MPEG audio technology,
software, components, links

MP3: The
Definitive Guide Sample Chapter 2: How MP3 Works: Inside the Codec

MPEG Audio FAQ

mp3/mp3PRO Patent and Software
Licensing Information

The
Internet-audio (r)evolution - MP3 has launched digital audio into the
mainstream, revolutionizing the way people collect and play music

Computer audio

General

Computer
speech compression techniques - comp.speech.FAQ,
speech data, software

Digital audio
experiments - pages devoted to DSP and digital audio

PC soundcard hardware
pages - information about PC soundcard hardware, software and many
related topics

File formats

About Sound Formats - a
brief description of the various sound formats

Audio
file formats FAQ

Soundcards

Interfacing Microphones to
Computer Sound Cards

Tomi
Engdahl's PC soundcard pages

Software

RealAudio - real time audio on
Internet

Sound Processing Kit
- object-oriented class library for audio signal processing implemented
in C++

Contents rights management

Tecording industry is worried about copying
of their products (CDs). The recording industry has tried putting down the
first critical pieces for a system it hopes will keep songs on the Net from
being pirated.

The
general term 'piracy" refers to the illegal duplication and distribution
of sound recordings and takes three specific forms: counterfeit, pirate and
bootleg.
Counterfeit recordings are the unauthorized recording of the prerecorded
sounds, as well as the unauthorized duplication of original artwork, label,
trademark and packaging of prerecorded music. Pirate recordings are the
unauthorized duplication of only the sounds of one or more legitimate
recordings. Bootleg recordings are the unauthorized recording of a musical
broadcast on radio or television or of a live concert.

The most talked about
technology has been Secure Digital Music Initiative (SDMI). The first phase of
the SDMI system requires that portable digital music player manufacturers
implement several security components, foremost among them a digital rights
management system (DRM). This will allow record labels to securely distribute
and track files as they are transmitted over the Net and on to portable
players.

General information

Digital Rights Management
- information portal

The Federal Anti-Piracy and
Bootleg FAQ Bootleg recordings are the unauthorized recording of a
musical broadcast on radio or television or of a live concert.

SDMI

SDMI:
Divide or Conquer? - basic information on SDMI system

SDMI Introduction by RIAA

Why
Secure Digital Music Initiative is falling apart

CD copy protection

Ever
since Napster came to prominence, the music CD publishers have been looking for
a way to stop people sharing MP3s extracted from their CDs, and now they think
they've found it -- by copy-protecting the CD releases. They hope that by
making CDs unplayable on computers that this will reduce the number of MP3s
getting onto the internet.

CD
copy protection is mainly done to make it harder to make copied of CD or
converting them to MP3 files on PC. There have been meny
technologies tied, but none of them is completely safe or problem free (and
there will not be any such for CD). Generally CD copy protection systems more
or less try put some type of errors to CD disk content (tale or contensts or audio itself) so that it does not either play
on PC or sound gets corrupted on "CD ripping" process. No CD
protection which still allows playing the music is bulletproof, even though protection
makers say "We can stop all kinds of copying, even on domestic CD
recorders." Sensible industry experts are skeptical on CD protection
systems promises of them being any long term solution,
ebecause hackers have defeated earlier security
(promised to be secure) technologies very quickly. The fact for all CD
protections is that if you can play the CD with CD-player, analogue copies can
be made to any analogue devices (recorded to tape) or current digital devices
with analogue inputs (for example PC soundcard) with some loss of quality.

Cactus
Data Shield protection for CD is designed to prevent MP3-ripping by corrupting
the table of contents so it won't play on a PC. Cactus inserts modifications to
original CDs in a way that confuses CD-ROM devices during the copying process.
If the CD is copied, however, the copier machine (a PC or disc-to-disc copier)
sees some fake control data as music. So when the copied disc is played, there
are bursts of distortion as the player tries in vain to decode the garbage.

SafeAudio is an Audio CD protection developed by Macrovision. SAFEAUDIO add samples that sound like bursts
of static, and scramble the ECC data around to make it look like an
uncorrectable error. Audio CD players will interpolate the samples during playback,
but CD-ROM drives doing digital audio extraction generally won't. The result is
a disc that plays back correctly on a CD player, but won't "rip" or
copy correctly on a CD-ROM drive. SAFEAUDIO v3 changes the music data at the
bit level, flipping a fraction of a disc's billions of 1s and to 0s ("very
subtle" degree of data corruption). y MusicGuard introduces selective alternative alterations to
the EFM data stream (can cause problems for CD-to-CD copying programs).

CDs
with the current version of Sony DADC Key2Audio cannot be recognised
by standard CD/DVD-ROM, CD-R and CD-RW drives, thus they do not play on PC.

Suncomm MediaClòQ v1.0 protected
CDs are designed such that they can not be read by any CD-ROM player or CD
burner. The CD has a heavy visible band at about the place where the audio
ends. Those CDs generally say "This CD is designed to play in standard
Audio CD players only and is not intended for use in DVD players."

Anti-piracy
system could damage loudspeakers - Some CD protection system can be
potentially dangerous to your HiFi system.

CD
Protections - This page has lots of information on game CD-ROM and
audio CD protection systems. This page contains a list of all known CD
Protections which are currently on the market, how they can be detected
and how to by-pass many of them.

Compromise
for CD copying is in the works - The record industry is experimenting
with a new strategy for protecting CDs from being copied in CD burners or
on computers. Unlike previous anti-copying measures, this plan will place
two versions of an album on a single disc: one in standard CD form,
modified so that it can't be transferred to a computer hard drive, and
another in Microsoft's Windows Media Audio digital format, rigged so that
files can be copied to a PC, but with some restrictions on how they can
be used.

Copy-protected CDs -
Buying a new CD? Watch out for inferior imitations

Could
CD protection end music swaps?

Lawmaker: Is CD
copy-protection illegal? - Record companies' efforts to protect CDs
against digital copying are beginning to draw scrutiny from lawmakers
concerned that the plans might violate the law. A 1992 law allows music
listeners to make some personal digital copies of their music. In return,
recording companies collect royalties on the blank media used for this
purpose.

Natalie Imbruglia wins CD protection race - Aussie pop
star (and former Neighbours actress) Natalie Imbruglia has won the race to produce the first
copyright-protected CD on general release to the UK
public.

SafeAudio
explained and should we fear it ? - How does it work ? - SafeAudio is an
Audio CD protection developed by Macrovision.

Copyright 2003 ELH
Communications Ltd. all rights reserved.